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Asterisk phone cisco 79xx

Created by: oej,Last modification on Tue 19 of Aug, 2008 [02:26 UTC] by vile

Configuring Cisco 79xx phones with Asterisk

This page documents how you configure a Cisco IP phone with Asterisk.


By default most Cisco VoIP phones come configured for Call Manager, which uses the 'Skinny' protocol - SCCP.
Asterisk has 2 implementations for this channel (required for the 7910/20):
  • Skinny implements a very basic set of telephone functions and ships with asterisk.
  • SCCP has implemented more of the SCCP protocol, so some class 5 features (hold, transfer, forward, etc) should work.

However, the 7905/7912 and 7940/60 can be reconfigured to use SIP (recommended for use with Asterisk):
http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a0080094584.shtml

The 7905 doesn't have Speakerphone (only a speaker for call monitoring or on-hook dialing), its SIP image has by far the best user interface of all cisco SIP phones, plus it's nicely compact, and the display even has a higher resolution than that of the 7960. The 7912 is the same as the 7905, but with a built-in ethernet switch.

The 7912/05 phones use a different configuration file format and syntax; for more information about configuring these phones please refer to Cisco 7905/7912 IP Phones.

With the right Cable, 79xx series phones can use standard POE injectors. They also work out of the box with Aironet power injectors. (N.B., the wrong cable may damage your phone!)





LATEST FIRMWARE VERSION

Version 8.8 is now released (a few minor bugfixes)
V8.8 Release Notes: http://www.cisco.com/en/US/products/sw/voicesw/ps2156/prod_release_note09186a00808c2dbb.html
V8.8 Download available at: http://www.xs4all.nl/~graver1/cisco/SIP-7960/P0S3-08-8-00.zip

Version: v8.7 is now released as per this security advisory.
http://www.cisco.com/warp/public/707/cisco-sr-20070821-sip.shtml

V8.6 Release Notes: http://www.cisco.com/en/US/products/hw/phones/ps379/prod_release_note09186a008079f14c.html
V8.6 Download available at: http://www.xs4all.nl/~graver1/cisco/P0S3-08-6-00.zip

CISCO SIP FIRMWARE VERSION AVAILABLE TO PUBLIC
ftp://ftp.cisco.com/pub/voice/ip-phone/sip-7960/

ARCHIVED IMAGES
http://www.xs4all.nl/~graver1/cisco/SIP-7960/





Cisco 7940/7960 SIP Phone Software Images
Cisco's SIP phone software images including versions 3.0, 4.4, 5.3, 6.x and 7.x work well with Asterisk. Features have been implemented and caveats (earlier problems) corrected with each release. The v5.x and v6.x images have incorporated a software security feature that makes it impossible to revert back to earlier images, although reverting to earlier versions is possible within the same major release (i.e. 7.5 firmware can be rolled back to 7.4 firmware with no problems). Version 6.0 SIP image has been very stable for many Asterisk users. The new version 7.x images are largely untested, but feature mainly bug fixes, not new features.


Cisco SIP IP Telephone 7940/7960 Software (NOTE: This page is only available to registered Cisco.com users with a Cisco Service Agreement)

http://www.cisco.com/pcgi-bin/tablebuild.pl/sip-ip-phone7960

It appears to be impossible to upgrade directly from v3.x to v8.x (probably any version prior to v6.x), I had to upgrade to v6.3 first. It is very hard to find a copy of this the v6.3 firmware (not even on Cisco site). Search for P0S3-06-3-00.zip or get it at http://www.xs4all.nl/~graver1/cisco/SIP-7960/P0S3-06-3-00.zip

v8.5 Release Notes:
http://www.cisco.com/en/US/products/hw/phones/ps379/prod_release_note09186a008079a1da.html

v8.4 Release Notes:
http://www.cisco.com/en/US/products/sw/voicesw/ps2156/prod_release_note09186a008070c6dc.html
http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/ip-7900ser/phnrn84s.pdf&app=Tablebuild&status=showC2A


Version: v8.3
The file from cisco is designed for the cisco call manager software and is a ".cop" file.
The file from cisco is designed for the cisco call manager software and is a ".cop" file. This file is just a GZIP compressed TAR file. Just ungzip and untar the file to extract the the new files for the phone. It installs just like the version 7 software with a loader and an application file. The standard ZIP file should be released soon.
  • No longer lists server IP on display for caller's number.
  • At least on the 7960, this release seems to break the Asterisk qualify sip.conf setting. The phone shows as unreachable just after it registers. If you set "qualify=no" it works.
  • I have also seen some other phone display strangeness with this release on the 7960
v8.3 Release Notes:
http://www.cisco.com/en/US/products/sw/voicesw/ps2156/prod_release_note09186a008068f6c5.html

Version: v8.2
  • Using the existing config files (that worked with v7.x) it seems to work, execpt the CID now displays the IP address also.
  • new config params:
    • line1_contact: number, changes the username part that is sent in the sip Contact: header
    • transfer_onhook_enabled: 1 or 0. If enabled, a call will be transferred when the handset is hung up (rather than having to press the Trnsfr button again. The original behaviour still works.)
    • call_manager1_addr and call_manager1_sip_port, not sure what difference these make
    • dscpForAudio: id, tag audio packets for qos with this dscp id, replace tos_media field
    • connection_monitor_duration: sec
    • encrypt_key: 32hexdigits, careful with this one, some config encryption support or similar
    • xml_card_dir and xml_card_file, actually these appeared earlier, doesnt seem to try load the card.xml over tftp for this, anyone know?
    • Current bug: There appears to be a bug in v8.2 that causes the phone to be unable to register with Broadvoice's SIP proxy. Downgrading the firmware appears to resolve the issue.
v8.2 Release Notes:
http://www.cisco.com/en/US/products/hw/phones/ps379/prod_release_note09186a00806216ab.html


Version: v7.5
  • There seems to be a bug in the 7.5 SIP software (at least on the 7960) that cause the phones to drop registration with Asterisk and not re-register. A phone reboot forces them to reregister for a time. 7.4 with the same configuration does not seem to have the same registration problem. It's unclear exactly where the issue exists (phone config, asterisk, time settings, etc). - doretel
  • Firmware 7.5 breaks RFC compliance, by not ACKing a 487. http://bugs.digium.com/view.php?id=5336. Also, we have experienced 7940/60 phone hangs/reboots related to "XML Parse Error" displayed on phone screens. Reverting to v.7.4 eliminated these problems. -vechers (The "XML Parse Error" problem and the previously mentioned fix confirmed by rrizzi7210)

v7.5 Release Notes:
http://www.cisco.com/en/US/products/sw/voicesw/ps2156/prod_release_note09186a00804b85e8.html

v7.4 Release Notes:
http://www.cisco.com/en/US/products/sw/voicesw/ps2156/prod_release_note09186a008040567b.html

v7.3 Release Notes:
http://www.cisco.com/en/US/products/sw/voicesw/ps2156/prod_release_note09186a0080335cdc.html

v7.2 Release Notes:
http://www.cisco.com/en/US/products/sw/voicesw/ps2156/prod_release_note09186a008029b053.html

v7.1 Release Notes:
http://www.cisco.com/en/US/products/sw/voicesw/ps2156/prod_release_note09186a008022c65e.html

v7.0 Release Notes:
http://www.cisco.com/en/US/products/sw/voicesw/ps2156/prod_release_note09186a00802295b7.html

v7.0 and up:
  • A new design approach is used to store the existing software base in the Cisco 7940 and Cisco 7960 telephones. The new approach utilizes a "run from flash" design to use the Flash and RAM memory available more efficiently.
  • Bugfix: Media takes 0.4 sec to be set up (finally...)
  • Bugfix: HTTP requests fail, sysbufs get hung
  • Numerous other bug fixes, too many to mention (see release notes below)
Click here to see a step by step guide to upgrading to sip version 7.x Cisco 7940-7960 upgrade to version 7.x

Cisco SIP IP Phone Administrator Guide, Versions 6.x and 7.x
http://cisco.com/en/US/products/sw/voicesw/ps2156/products_administration_guide_book09186a00801d1978.html

The v6.0 software has implemented:
  • Alert-Info (play internal ring tones based on Alert-Info within the SIP header)
  • Auto Answer (2-way paging conversation without picking up handset)
  • DHCP Option 66
  • Directory Enhancements (user can add/change/delete entries in Personal Directory)
  • DSP (new digital signal processor)
  • DSP Alarms, Debugging Aids, and Logging (help diagnose problems)
  • Enhanced Tone and Ring Support (support for more complex tones and ringing patterns)
  • Hot Line / Speeddials (each line button can be programmed to act as a speeddial button)
  • Local Call Forwarding (redirects incoming calls to another extension/URL)
  • Message Waiting Stutter Tone
  • Multiple Call Appearance (receptionist style, all lines have the same extension)
  • Outbound Proxy Redesign (improves use of outbound proxy based on multiple DNS records)
  • SIP Call Statistics (call statistics sent in BYE / 200 OK messages)
  • Resolved Caveats (several previously documented problems have been resolved)

Note: Cisco software images are only available from Cisco's web site and are protected by copyright laws. Access to their web site requires an account be established. The easiest way to do that is to purchase a Maintenance Agreement from Cisco for approximately $8 per year (US).

Cisco 7941/7961 SIP Phone Software Images

Cisco's SIP phone range now includes the 7941/7961, as new firmware v8.0(1) came out for these that supports SIP in March 2006. Previously some v7.0 releases of SCCP had been released. These phones look physically identical to the 7940 phone with the exception of the much better quality display which can display grey shades, however the software for the 7941/7961 appears to have major differences to the 7940/7960 software.

Release notes for v8.0 SIP are at http://www.cisco.com/en/US/products/hw/phones/ps379/prod_release_note09186a0080621246.html

Note that the release notes clearly state at the top of the document that "This SIP firmware was designed and tested to interoperate with Cisco call control, most notably Cisco Unified CallManager version 5.0. Although this SIP deployment is IETF RFC 3261 compliant, it is not supported by Cisco TAC or Engineering for use with non-Cisco call control systems.". So YMMV if you call the TAC for support.

The v8.0 software for the 7941/7961 has these significant differences over the v8.0 software for the 7940/7960:
  • The built-in web server previously only in the non-SIP images is now in the v8.0 SIP builds
  • The phone does not appear to support the old style config files, all configs for this phone need to be in XML
  • The phone does not appear to support telnet, but instead supports SSH-2.0, although this appears to be undocumented on CCO. At this stage unsure what the default user/password is
  • The config file name is therefore now SEP<mac-address>.cnf.xml
  • Upgrading the firmware load involves the phone uploading 5 signed binary files: jar41sip.8-0-1-18.sbn, cnu41.3-1-1-15.sbn, apps41.1-1-1-15.sbn, dsp41.1-1-1-15.sbn, and cvm41sip.8-0-1-18.sbn (this was for release load version 8-0-2SR1-0-1 which is the latest as of early April 2006). These phones can be upgraded and the protocol support changed via the normal means of adding <loadInformation>term41.default</loadInformation> to the .cfg.xml file

At this stage, documentation on the format of the xml config file for SIP has proven to be extremely hard to find. Cisco are likely assuming that all adopters of this phone will be running it in a CallManager environment whereby the configs are automatically generated by the CCM. If you do find some or have access to a CallManager 5.0 which can generate these files, please document the basic format so others can configure these phones for SIP in a non CCM environment.

More information on this new config file format and this new series of phones can be found at the 79x1 xml config page Please edit it and improve it as much as you can as information of the nature in that document has not to date been easy to find.


Software Upgrade Requirements

All software images are upgraded through a TFTP server at your location. If you don't have one, several free Unix and Windows packages are available via the Internet. The specific instructions as to exactly how to accomplish the upgrade should be reviewed from Cisco's web site as the exact steps (and possible backout steps) may change from version to version.

The TFTP server directory must include the following files as a minimium (most are upper/lower case sensitive):
  • OS79XX.TXT (The content of this file is solely the software image filename stripped of the .bin, i.e. P0S3-06-1-00)(Contains the universal application loader image in 7.x)
Note:(The 7.1 image upload needs to be spelled P003-07-1-00 in the OS79XX.TXT and P0S3-07-1-00 in the SIPDefault.cnf)
Note:(7941G does not fetch OS79XX.TXT when upgrading from SCCP 8.x to SIP 8.x)
  • P003-xx-y-zz.bin (Nonsecure universal application loader for upgrades from pre-5.x images.)
  • P003-xx-y-zz.sbn (Secure universal application loader for upgrades from images 5.x or later.)
  • P0a3-xx-y-zz.loads (File that contains the universal application loader and application image, where "a" represents the protocol of the application image loads file 0-SCCP, S-SIP, M-MGCP.)
  • P0a3-xx-y-zz.sb2 (Application firmware image, where "a" represents the application firmware image.)
  • SIPDefault.cnf (Contains generic parameters for all Cisco phones at your location)
  • SIP00036BAAD139.cnf (Where the last 12 hex digits is the MAC address of your Cisco phone) Sample php script to create the cnf file.

In addition, the following optional files may also be present in the TFTP directory:
  • dialplan.xml (contains entries like "9,1...." that cause the phone to automatically dial after a match)
  • RINGLIST.DAT (a list of ringing tones to be downloaded, like ringer1.pcm)
  • ringer1.pcm (a ringing tone to be downloaded to the phone)

See also: John Todd's examples

*** Simplify Updates (Auto-Loader Support) ***

This will save you alot of wasted time trying to update newer firmware! For easiest, direct firmware updates from say factory installed SCCP images direct to latest SIP firmware (e.g. SCCP v3.1 to SIP v7.4) — add the following files to your TFTP directory to assist the SCCP based generic Auto-Loader added as of v5.x to Cisco's SIP/SCCP images:
  • XMLDefault.cnf.xml
  • xmlDefault.CNF.XML

Due to inconsistent coding by Cisco, different firmware may look for different case-sensitive versions of the same file, thus the need for at least the two variations above to cover new phones and a good portion of older one's. Additionally, here is a usable example of the XML content that should be inserted into the files (be sure and update with the firmware version you wish to load, and match the SIPDefault.cnf and SIPxxxxxxx.cnf file's "image_version=" entries to match!):

Just add the loadInformation lines relevant to the phone that you have.

<Default>
  <callManagerGroup>
     <members>
        <member priority="0">
           <callManager>
              <ports>
                 <ethernetPhonePort>2000</ethernetPhonePort>
                 <mgcpPorts>
                    <listen>2427</listen>
                    <keepAlive>2428</keepAlive>
                 </mgcpPorts>
              </ports>
              <processNodeName></processNodeName>
           </callManager>
        </member>
     </members>
  </callManagerGroup>
 <loadInformation30008  model="Cisco 7902">CP7902080001SCCP051117A</loadInformation30008>
 <loadInformation20000  model="Cisco 7905">CP7905080001SCCP051117A</loadInformation20000>
 <loadInformation6  model="Cisco 7910">P00405000700</loadInformation6> 
 <loadInformation307  model="Cisco 7911">SCCP11.8-0-1-0S</loadInformation307>
 <loadInformation30007  model="Cisco 7912">CP7912080001SCCP051117A</loadInformation30007>
 <loadInformation30002  model="Cisco 7920">cmterm_7920.4.0-02-01</loadInformation30002>
 <loadInformation9  model="Cisco 7935">P00503011200</loadInformation9>
 <loadInformation30019  model="Cisco 7936">cmterm_7936.3-3-9-0</loadInformation30019>
 <loadInformation8  model="Cisco 7940">P00308000100</loadInformation8>
 <loadInformation115  model="Cisco 7941">SCCP41.8-0-1-0S</loadInformation115>
 <loadInformation309  model="Cisco 7941G-GE">SCCP41.8-0-1-0S</loadInformation309>
 <loadInformation7  model="Cisco 7960">P00308000100</loadInformation7>
 <loadInformation30018  model="Cisco 7961">SCCP41.8-0-1-0S</loadInformation30018>
 <loadInformation308  model="Cisco 7961G-GE">SCCP41.8-0-1-0S</loadInformation308>
 <loadInformation30006  model="Cisco 7970">SCCP70.8-0-1-0S</loadInformation30006>
 <loadInformation119  model="Cisco 7971">SCCP70.8-0-1-0S</loadInformation119>
 <loadInformation302  model="Cisco 7985">cmterm_7985.4-0-2-0</loadInformation302>
 <loadInformation30028  model="ISDN BRI Phone"></loadInformation30028>
 <loadInformation30016  model="Cisco IP Communicator"></loadInformation30016>
 <loadInformation358  model="Cisco Unified Personal Communicator"></loadInformation358>
 <loadInformation12  model="Cisco ATA 186">ATA030203SCCP051201A</loadInformation12>
 <loadInformation61  model="H.323 Phone"></loadInformation61>
 <authenticationURL></authenticationURL>
 <directoryURL></directoryURL>
 <idleURL></idleURL>
 <informationURL></informationURL>
 <messagesURL></messagesURL>
 <servicesURL></servicesURL>
</Default>

Misc

See reboot.pl (down as of late - Internet Archive link: http://web.archive.org/web/20060427051951/http://mklein.bendtel.net/mkreboot.pl). A perl script to handle remote rebooting of the 79xx class phones (useful for multiple-phone upgrades). Requires Net::Telnet.

NOTE: For Reference, reboot.pl has been moved to http://www.nmedia.net/~mklein/reboot.pl. Still Requires Net::Telnet.

So you blew up your 7940/7960 trying to load new firmware



Logo Displayed on 79XX Screen

A non-Cisco logo can be displayed on the 79XX screen. Cisco's documentation suggests the logo be a Windows Bitmap form (*.BMP) with 256 colors and 90 x 56 pixels in size. Only two colors are displayed, black or white. The image must be saved in greyscale format. In GIMP it is Image->Mode->Greyscale. If the size of the logo is larger then this specification, the phone will rescale to fit (within reason). Microsoft Paint and many other applications can be used to create the logo image.

Once the image is created, place the *.BMP file on any web site available to you (suggest /asterisk/mylogo.bmp, where the /asterisk directory is not freely advertised).
Modify the SIPDefault.cnt file entry to point to the web site:
logo_url: "http://www.mywebserver.com/asterisk/logo.bmp"
and reboot the Cisco phone.
Note: the smaller the logo file, the quicker it will load. Typcial logo files should be around 10k bytes.

Note: 7940/7960 phones actually display two colors at 2-bit depth, or four solid colors. For best results creating images/logos that don't dither and degrade, use #000000 (black) and #FFFFFF (white or clear on the LCD display), then #404040 (dark grey) and #808080 (light gray) as your alternate solid colors (or as the antialiasing colors for excellent results).

Company Telephone Directory

The 79XX phones have four panel keys labeled as Messages, Services, Directories, and Settings. The Directory key can be programmed to view your company's telephone directory by displaying Names and Telephone Numbers that are stored on any web site available to you.
Modify the SIPDefault.cnf file entry to point to the web site:
directory_url: "http://www.mywebserver.com/asterisk/directory.xml"
The phone must be rebooted in order to read the above file.
The web site file /asterisk/directory.xml should include xml entries like:
 
 <CiscoIPPhoneDirectory>       
   <Title>IP Telephony Directory</Title>
   <Prompt>People reachable via VoIP</Prompt>                    
   <DirectoryEntry>                                              
     <Name>Rich</Name>         
     <Telephone>3000</Telephone>
   </DirectoryEntry>                                      
   <DirectoryEntry>             
     <Name>Todd</Name>       
     <Telephone>3001</Telephone>                                            
   </DirectoryEntry>      
 </CiscoIPPhoneDirectory>


Note: Each time a user presses the Directory key and accesses the External Directory option from the menu, the phone will access the contents of this html file and display whatever text entries included in it. Therefore, changes to the html file do not require any futher rebooting of the Cisco phone. Cisco has published Cisco IP Phone Services Application Development Notes (CMXML_App_Guide.pdf ) that further explains options and file contents.

Asterisk Cisco 79XX XML Services

Services Button

The 79XX phones have four panel keys labeled as Messages, Services, Directories, and Settings. The Services key can be programmed to execute CGI scripts that are stored on any web site available to you. The CGI scripts can perform any action that you are capable of programming. None are provided by Cisco.
Modify the SIPDefault.cnf file entry to point to the web site:
services_url: "http://www.mywebserver.com/asterisk/myscriptpage.html"
The phone must be rebooted in order to read the above file.
The web site file /asterisk/myscriptpage.html should include entries for each service that you plan to make available to your phone users. The exact content and syntax is also documented in the Cisco IP Phone Services Application Development Notes (CMXML_App_Guide.pdf ) noted above.

Messages Button

When the Messages button is pressed, you can cause the phone to directly dial an extension in your asterisk dialplan. Just configure the phone as:
       messages_uri:  "<extension>"
where <extension> is what you wish the phone to dial when the Messages button is pressed. You can then catch the call in either a standard VoiceMailMain() invocation a la
       exten => _42,1,VoiceMailMain()
or, be cute and bypass entry of mailbox number and password a la
       exten => _42,1,VoiceMailMain(s<mbox num>)
To make the Messages button work for any extension (assuming your extensions are numbered appropriately), use:
       exten => _42,1,VoiceMailMain(s${CALLERIDNUM})




Asterisk Cisco 79XX XML Services

Ringtones

http://www.cisco.com/en/US/products/sw/voicesw/ps2156/products_administration_guide_chapter09186a0080087511.html#1042487
The Cisco SIP IP phone ships with two ring types: Chirp1 and Chirp2. By
default, your ring type options will be those two choices. However,
using the RINGLIST.DAT file, you can customize the ring types that are
available to the Cisco SIP IP phone users.

Step 1
Create a pulse code modulation (PCM) file of the desired ring
types and store the PCM files in the root directory of your TFTP server.
PCM files must contain no header information and comply with the
following format guidelines:

8000 Hz sampling rate
8 bits per sample
ulaw compression
240 - 16080 samples long ( 0.03 sec - 2.01 sec )

For example, to use sox to generate the tones, use

sox -t wav in.wav -t raw -r 8000 -U -b -c 1 out.raw resample -ql


Step 2
Using a ASCII editor, open the RINGLIST.DAT file and for each
of the ring types you are adding, specify the name as you want it to
display on the Ring Type menu, press Tab, and then specify the filename
of the ring type. For example, the format of a pointer in your
RINGLIST.DAT file should appear similar to the following:

Ring Type 1 ringer1.pcm

Step 3
After defining pointers for each of the ring types you are
adding, save your modifications and close the RINGLIST.DAT file.

Caveat:

If you have configured a secondary tftp-server(ie. dyn_tftp_addr : 192.168.1.10) in SIPDefault.cnf, or SIP<MACADDR>.cnf which cannot be reached then the phone will not attempt to download the RINGLIST.DAT file.


Controlling ring tones from Asterisk

By setting the Asterisk variable ALERT_INFO before you call Dial, Asterisk will add ringer tone info to the SIP invite that is sent to the phone. Note that this only seems to work for the internal ringtones and not for any custom ringtones.

Asterisk version <1
 exten => 3010,1,SetVar(__ALERT_INFO=<Bellcore-dr1>) 

Assterisk 1.0 and 1.2
 exten => 3010,1,SetVar(_ALERT_INFO=<Bellcore-dr1>)

Asterisk 1.4
 exten => 3010,1,SIPAddHeader(Alert-Info: <Bellcore-dr1>)


Available ring tones by default

 Bellcore-BusyVerify 
 Bellcore-Stutter 
 Bellcore-MsgWaiting 
 Bellcore-dr1  
 Bellcore-dr2 
 Bellcore-dr3 
 Bellcore-dr4 
 Bellcore-dr5

While examining the firmware for the 7940/7960 I found the following strings. I have not verified if these are in fact ringtones because I don't have a phone to test them with at the moment but they might be worth a try. {I have tried them, and they don't seem to do anything on the 7940 with
Bellcore-Inside
Bellcore-Outside
Bellcore-Busy
FBellcore-Alerting
FBellcore-BusyVerify
Bellcore-Stutter
Bellcore-MsgWaiting
Bellcore-Reorder
Bellcore-CallWaiting
Bellcore-Cw2
Bellcore-Cw3

Bellcore-Cw4
Bellcore-Hold
FBellcore-Confirmation
FBellcore-Permanent
Bellcore-Reminder
FBellcore-None
Bellcore-dr1
Bellcore-dr2
Bellcore-dr3
Bellcore-dr4
Bellcore-dr5

FCisco-ZipZip
Cisco-Zip
FCisco-BeepBonk




DialPlan Notes (DIALPLAN.XML)

The DIALPLAN.XML file controls the phone's matching of digits. By default "*" matches anything and times out after 5 seconds. Users must push 'Dial' or '#' to connect if they don't want to wait 5 seconds. For a variety of reasons, not least of which being that most phone users are not accustomed to pressing 'Dial' on their offices, it may be desirable to configure a dial plan for your organization.

If you wish to use the hash (by default it will immediately dial the number entered) include it explicitly as part of a pattern in DIALPLAN.XML
   <DIALTEMPLATE>
       <TEMPLATE MATCH="#..." Timeout="5" User="Phone" />
       <TEMPLATE MATCH="*" Timeout="5" User="Phone" />
   </DIALTEMPLATE>

In another example, we immediately match 9+10digits or 9+1+10digits, and match 5+2digits as internal extensions
  <DIALTEMPLATE>
     <TEMPLATE MATCH="5.." TIMEOUT="0"/>
     <TEMPLATE MATCH="9,1.........." TIMEOUT="0" Tone="Bellcore-Alerting"/>
     <TEMPLATE MATCH="9,.........." TIMEOUT="0"/>
  </DIALTEMPLATE>

In the above example, a secondary dial tone is invoked by the comma character. If the Tone attribute is left blank, the default will be used. Or you can specify one of the following:

  • Bellcore-Alerting
  • Bellcore-Busy
  • Bellcore-BusyVerify
  • Bellcore-CallWaiting
  • Bellcore-Confirmation
  • Bellcore-dr1
  • Bellcore-dr2
  • Bellcore-dr3
  • Bellcore-dr4
  • Bellcore-dr5
  • Bellcore-dr6
  • Bellcore-Hold
  • Bellcore-Inside
  • Bellcore-None
  • Bellcore-Outside (default)
  • Bellcore-Permanent
  • Bellcore-Reminder
  • Bellcore-Reorder
  • Bellcore-Stutter

  • CallWaiting-2
  • CallWaiting-3
  • CallWaiting-4

  • Cisco-BeepBonk
  • Cisco-Zip
  • Cisco-ZipZip

Notes: This file is case sensitive in some firmware versions; all elements and attributes should be uppercase (except Tone) or the entries may be ignored. According to Cisco, the phone will always match the LONGEST expression.



Call Waiting Feature

The 79XX series phones have a good way of handling SIP registrations provided the Call Waiting feature isn't turned off. Most other SIP phones require an individual SIP username and password for each line appearance. Instead, the 79XX will automatically roll-over to the next available line. So, for example, on a 7960 you can have all six lines programmed to the same SIP username/password and the phone will automatically handle the call waiting function. Note: If you use any other method of ringing multiple lines on the phone (i.e. dialing SIP/123&SIP/456) your phone will show a confusingly high number of missed calls.


For example:


SIPXXXXX.cnf:

 # Line 1 Settings
 line1_name: "510"                     ; Line 1 Extension\User ID
 line1_displayname: "x510"           ; Line 1 Display Name
 line1_shortname: "x510"      ; Comment next to the button
 line1_authname: "510"         ; Line 1 Registration Authentication
 line1_password: "test"         ; Line 1 Registration Password

 # Line 2 Settings
 line2_name: "510"                          ; Line 2 Extension\User ID
 line2_displayname: "x510"                   ; Line 2 Display Name
 line2_shortname: "x510"            ; Comment next to the button
 line2_authname: "510"         ; Line 2 Registration Authentication
 line2_password: "test"         ; Line 2 Registration Password

In your sip.conf:

 [510]
 type=friend
 username=510
 secret=test
 host=dynamic
 dtmfmode=rfc2833
 context=whatever
 canreinvite=no
 nat=no
 mailbox=510@default
 callerid=<510>

In your extensions.conf:

 exten => 510,1,Dial(SIP/510,20,mTt)
 exten => 510,2,Voicemail(u510@default)
 exten => 510,3,Hangup
 exten => 510,102,Voicemail(b510@default)
 exten => 510,103,Hangup



Asterisk Configuration File Examples



sip.conf

 [3014]
 type=friend ; This device takes and makes calls
 host=dynamic ; This host is not on the same IP addr every time
 username=3014 ; Username programmed into Cisco phone
 secret=mypassword ; Password for device
 context=from-sip        ; Inbound calls from this phone go to this context
 nat=yes ; nat=yes if this phone is behind a NAT box or firewall
 callgroup=2 ; the group to which this phone belongs for *8 phone ringing pickup
 pickupgroup=2 ; the pickup group allowed from this phone when *8 is dialed
 mailbox=3014             ; Activate the message waiting light if this voicemailbox has messages in it

extensions.conf

 exten => 3014,1,Dial(SIP/3014,15,t) ; see "show application dial" for options and formats
 exten => 3014,2,Voicemail2(u3014) ; go to Voicemail2 if phone is "U"nanswered
 exten => 3014,102,Voicemail2(b3014)  ; go to Voicemail2 if phone is "B"usy
 exten => 3014,103,Hangup ; and then hangup.

voicemail.conf

 format=gsm 
 servermail=mail.myserver.com

 attach=no
 maxmessage=120 
 maxsilence=10
 pbxskip=yes
 fromstring=NPI VM
 emailbody=\nVM for x ${VM_MAILBOX} from ${VM_CALLERID} dur: ${VM_DUR} \n
 [default]
 ; Note: following sends a text message to a cell phone telling me someone left a voicemail
 3014 => 3014,FirstName LastName,4015719329@vtext.com 

Troubleshooting Phone Registration

From the asterisk command line, execute "sip show peers" and "sip show users" to display the current status of the Cisco phone. If no entries appear in the list for this phone, then review the "username=3014" and "secret=mypassword" in sip.conf to ensure they match the entries programmed into the Cisco phone.
 *CLI> sip show peers
 Name/username    Host                 Mask             Port     Status    

 3014/3014        67.11.89.61     (D)  255.255.255.255  5060     Unmonitored
 *CLI> sip show users
 Username         Secret           Authen           Def.Context      A/C  
 3014             mypassword          md5,plaintext    from-sip         No   

Troubleshooting Cisco Phone

The Cisco 79XX phones support telnet. (Enable telnet_level: 2 in the config) To diagnose problems with the Company Directory function noted above (as an example), telnet to the phone's IP address using the login password provided in the SIP00036BAAD139.cnf file noted above. For example, to diagnose a possible http problem, do the following:

 SIP Phone> debug http
 Enabling bug logging on this terminal - use 'tty mon 0' to disable
 debugs: http timestamp
 SIP Phone> [11:39:39] Connect2WWWIPPort called IpAddr[0], port[80], hostName[www.mydomain.com]
 [11:39:39] Connect2WWWIPPort Sending Request to IpAddr[207.212.93.75], port[80]
 [11:39:39] HTTP RECV (ACK CMD)
 [11:39:39] HTTP RECV (OPEN CMD)
 [11:39:39]
 HTTP Send [178] Bytes of Data
 Data Packet is:
 ===============
 GET /asterisk/directory.html?name=SIP00036BC38B88 HTTP/1.1
 User-Agent: Allegro-Software-WebClient/3.10b1
 Host: www.mydomain.com
 Connection: Close


Troubleshooting Cisco Phone Configuration issues.


  • Any configuration parse error warnings are shown in the phone menu (Settings - Status - Status messages). Or from the telnet CLI, you can type "show status"

  • Check the file that it reports a problem with. Check the permissions on the TFTP server that it can read the file.

  • Carefully check the config file for obvious errors (missing speechmarks, spurious spaces etc.)

If you still cannot locate the problem, you can do the following to erase the phone config and debug it when it loads it back in again:
(Unfortunately this will delete any speed dial keys configured on the handset, and also auto answer setting as these cannot be put back again via TFTP.)

  • Telnet to the phone:



telnet 192.168.1.100

Connected to 192.168.1.100.
Escape character is ']'.


Password :*****

Cisco Systems, Inc. Copyright 2000-2005
Cisco IP phone MAC: 000a:8a2c:864a
Loadid: SW: P0S3-08-6-00 ARM: PAS3ARM1 Boot: PC030301 DSP: 4.0(2.0)[A0]
SIP Phone> show status

Current Phone Status
--------------------
W350 unprovisioned proxy_backup
W351 unprovisioned proxy_emergency
W310 1 Error(s) Parsing: SIPDefault.cnf


Phone has reported an error parsing the SIPDefault.cnf, but unfortunately does not say exactly what it was!
The warnings about proxy_backup and proxy_emergency are normal in my case because I do not use that feature.

Now (Make sure your TFTP files are in place and that your DHCP server is set to tell your phone the next-server of it.)

SIP Phone> debug xml-events
SIP Phone> debug xml-vars
SIP Phone> erase protflash

Now among the (copious) output you will see the phone parsing the config:


[00:28:46:4136406] ---LIST---
[00:28:46:4136407] href=basic
[00:28:46:4136407] card=status
[00:28:46:4136407] icon=WAIT
[00:28:46:4136407] status=Requesting Configuration
[00:28:46:4136408] ------
[00:28:46:4136408] XML Event: href=basic, event=(null), target=(null), action=(null), card=status
[00:28:46:4136410] TFTP: Request file:SIPDefault.cnf from: <x.x.x.x>
[00:28:48:4136630] TFTP: File received successfully!

[00:28:48:4136633] Parse error: var: wibble_foo not found in table
[00:28:48:4136636] Parse error: 1 Errors found

[00:28:48:4136636] %W350 unprovisioned proxy_backup
[00:28:48:4136637] %W351 unprovisioned proxy_emergency
[00:28:48:4136637] %W362 No Valid Line Names Provisioned
[00:28:48:4136649] %W310 1 Error(s) Parsing: SIPDefault.cnf
[00:28:48:4136650] TFTP: Request file:SIP0009E8B4AE3E.cnf from: <x.x.x.x>
[00:28:48:4136668] TFTP: File received successfully!
[00:28:48:4136672] %W350 unprovisioned proxy_backup
[00:28:48:4136673] %W351 unprovisioned proxy_emergency
[00:28:48:4136685] upgrade_check(P0S3-08-6-00)

In this case it did not like my "wibble_foo: thing" in SIPDefault.conf !


Asterisk + SIP + Cisco 7975


This is working configuration of my server and runing gr8 with all feature....enjoy it...

Home page:- http://www.linuxbug.org

contact detail:-

Satish Patel
Mobile:- +91-9818875535
Email:- satish.lx@gmail.com

Installation method

Copy this configuration file in your tftp root directory and configure your phone for tftp and reboot your ip phone. beafore doing this change your setting accoring your setup.

SEP<MAC_ADDRESS>.cnf


<device xsi:type="axl:XIPPhone" ctiid="203849429" uuid="{96f8508b-10ef-f98c-d20d-0471777ec725}">
<fullConfig>true</fullConfig>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>user</sshUserId>
<sshPassword>pass</sshPassword>
<devicePool uuid="{a755aa55-089c-2b47-9603-c7d51b9ca4b5}">
<name>Dallas 5.0 Beta</name>
<dateTimeSetting uuid="{9ec4850a-7748-11d3-bdf0-00108302ead1}">
<name>CMLocal</name>
<dateTemplate>D/M/Y</dateTemplate>
<timeZone>GMT Standard/Daylight Time</timeZone>
</dateTimeSetting>
<callManagerGroup>
<name>5.0 Beta</name>
<tftpDefault>true</tftpDefault>
<members>
<member priority="0">
<callManager>
<name>71.5.250.225</name>
<description>CallManager 5.0 Beta Pub - 5.0.1.032</description>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
</ports>
<processNodeName>71.5.250.225</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>

<sipProfile>
<sipProxies>
<backupProxy></backupProxy>
<backupProxyPort></backupProxyPort>
<emergencyProxy></emergencyProxy>
<emergencyProxyPort></emergencyProxyPort>
<outboundProxy></outboundProxy>
<outboundProxyPort></outboundProxyPort>
<registerWithProxy>true</registerWithProxy>
</sipProxies>

<sipCallFeatures>
<cnfJoinEnabled>true</cnfJoinEnabled>
<callForwardURI>x-cisco-serviceuri-cfwdall</callForwardURI>
<callPickupURI>x-cisco-serviceuri-pickup</callPickupURI>
<callPickupListURI>x-cisco-serviceuri-opickup</callPickupListURI>
<callPickupGroupURI>x-cisco-serviceuri-gpickup</callPickupGroupURI>
<meetMeServiceURI>x-cisco-serviceuri-meetme</meetMeServiceURI>
<abbreviatedDialURI>x-cisco-serviceuri-abbrdial</abbreviatedDialURI>
<rfc2543Hold>false</rfc2543Hold>
<callHoldRingback>2</callHoldRingback>
<localCfwdEnable>true</localCfwdEnable>
<semiAttendedTransfer>true</semiAttendedTransfer>
<anonymousCallBlock>2</anonymousCallBlock>
<callerIdBlocking>2</callerIdBlocking>
<dndControl>0</dndControl>
<remoteCcEnable>true</remoteCcEnable>
</sipCallFeatures>

<sipStack>
<sipInviteRetx>6</sipInviteRetx>
<sipRetx>10</sipRetx>
<timerInviteExpires>180</timerInviteExpires>
<timerRegisterExpires>3600</timerRegisterExpires>
<timerRegisterDelta>5</timerRegisterDelta>
<timerKeepAliveExpires>120</timerKeepAliveExpires>
<timerSubscribeExpires>120</timerSubscribeExpires>
<timerSubscribeDelta>5</timerSubscribeDelta>
<timerT1>500</timerT1>
<timerT2>4000</timerT2>
<maxRedirects>70</maxRedirects>
<remotePartyID>true</remotePartyID>
<userInfo>None</userInfo>
</sipStack>

<autoAnswerTimer>1</autoAnswerTimer>
<autoAnswerAltBehavior>false</autoAnswerAltBehavior>
<autoAnswerOverride>true</autoAnswerOverride>
<transferOnhookEnabled>false</transferOnhookEnabled>
<enableVad>false</enableVad>
g711
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<phoneLabel>satish</phoneLabel>
<stutterMsgWaiting>2</stutterMsgWaiting>
<callStats>false</callStats>
<offhookToFirstDigitTimer>15000</offhookToFirstDigitTimer>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>true</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>

<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>5493</featureLabel>
<proxy>71.5.250.225</proxy>
<port>5060</port>
<name>5493</name>
<displayName>5493</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>5493</authName>
<authPassword>5493</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messagesNumber></messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>5493</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>

<line button="2">
<featureID></featureID>
<featureLabel></featureLabel>
<speedDialNumber></speedDialNumber>
</line>
</sipLines>

<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>
<softKeyFile>SK50719900-3bee-4594-bc3f-6400e1a33bf0.xml</softKeyFile>
</sipProfile>

<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>

<loadInformation>SIP75.8-3-3SR2S</loadInformation>

<vendorConfig>
<disableSpeaker>false</disableSpeaker>
<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
<pcPort>0</pcPort>
<settingsAccess>1</settingsAccess>
<garp>0</garp>
<voiceVlanAccess>0</voiceVlanAccess>
<videoCapability>0</videoCapability>
<autoSelectLineEnable>0</autoSelectLineEnable>
<webAccess>1</webAccess>
<daysDisplayNotActive>1,7</daysDisplayNotActive>
<displayOnTime>08:00</displayOnTime>
<displayOnDuration>10:30</displayOnDuration>
<displayIdleTimeout>01:00</displayIdleTimeout>
<spanToPCPort>1</spanToPCPort>
</vendorConfig>

<versionStamp>1136931633-57191cee-5ffc-4342-b286-4246b4991890</versionStamp>

<userLocale>
<name>English_United_States</name>
<uid>1</uid>
<langCode>en_US</langCode>
<version>1.0.0.0-1</version>
<winCharSet>iso-8859-1</winCharSet>
</userLocale>

<networkLocale>United_States</networkLocale>
<networkLocaleInfo>
<name>United_States</name>
<uid>64</uid>
<version>1.0.0.0-1</version>
</networkLocaleInfo>

<deviceSecurityMode>1</deviceSecurityMode>
<idleTimeout>0</idleTimeout>
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<servicesURL></servicesURL>
<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>
<transportLayerProtocol>4</transportLayerProtocol>
<capfAuthMode>0</capfAuthMode>

<capfList>
<capf>
<phonePort>3804</phonePort>
</capf>
</capfList>

<certHash></certHash>
<encrConfig>false</encrConfig>
</device>


[+]







If you still cannot find it, try adding "debug config" which will generate even more config debugging.

(Note: You can also reset the 7960 by pressing *+6+settings at the same time.)
(Note: additional technical data was displayed but it was clipped from this documentation.)

Check out:
http://www.cisco.com/en/US/products/sw/voicesw/ps2156/products_administration_guide_chapter09186a00801d1988.html
    
 This site has all the information you will need to help you work with the phone in the telnet prompt!

See also:

Where to buy:




Asterisk | Asterisk Configuration | Channel Configuration | Configuration for Specific Phones

Comments

Comments Filter
222

333Perl script for Cisco 7960 VoIP phone to dial number remotely over Telnet.

by norim, Monday 07 of April, 2008 [13:34:08 UTC]
the script that will dial a number and hung up after number of seconds specified.
EXAMPLE1: perl scriptname.pl 172.16.31.10 303 5 lim
this would dial number 303 on cisco phone @172.16.31.10 and hung up after 5 seconds.
EXAMPLE2: perl scriptname.pl 172.16.31.10 303
this would dial number 303 on cisco phone @172.16.31.10 and exit.


# check for no arguments; if none act like 'cat'
if ($#ARGV == -1) {
	print <"1st IP\n2nd number to dial\n3ed seconds to talk\n\n4th type lim if u want to limit conversation\n">; exit;

} elsif ($ARGV[0] eq "-h") {
	print <"help\n">;
}


$ip=$ARGV[0];
$number=$ARGV[1];
$slp=$ARGV[2];
$stay=$ARGV[3];

use Net::Telnet;
sub login {
$telnet = new Net::Telnet ( Timeout=>10, Errmode=>'die');
$telnet->open("$ip");
$telnet->waitfor('/Password :$/i');
$telnet->print('cisco');
$telnet->waitfor('/admin> $/i');
}

sub dial_num {
if (status() eq "IDLE") {
print $telnet->cmd('test open');
print $telnet->cmd('test key spkr');
print $telnet->cmd("test key $number");
print $telnet->cmd('test key soft3');
}
	else {
print "BUSY, phones state=$state\n";
}	
	}


sub status {
@lines = $telnet->cmd('show lsm');
$lsm = @lines[4];
$state =  substr $lsm, 19, 4;
return ($state);
}

sub hungup {
if (status() eq "IDLE") {
exit;
}
else {print $telnet->cmd('test key soft2');}
}



login();
if (status() ne "IDLE") {
exit;
}
else {
dial_num();
if ($ARGV[3] eq lim) {
sleep($slp);
hungup();}
else {
exit
}
}


222

333Cisco IP Phone Visual Voicemail

by fcnorris4, Tuesday 13 of November, 2007 [20:26:09 UTC]
I have written a Visual Voicemail script for the Cisco IP Phone connected to Asterisk. It has been tested with Asterisk 1.4 and a Cisco 7970 IP Phone. It is written in Perl. You can find it on my minimal website: http://norrisnet.homeip.net
Feel free to contact me at cory.norris@earthlink.net with comments, concerns and suggestions.


What this script does is gives you a way to access, play, and delete your voicemails all from the Cisco IP Phone's screen/interface.

The specifics are this:
When you add the URL listed in the example services.xml file to your services menu for your Cisco Phone, it invokes the script. The script queries the phone to obtain the phone's extension. It uses that extension as the mailbox ID for the Asterisk/trixbox voicemail system. It then checks the Asterisk voicemail directory structure under your voicemail id and displays a list of available sub folders (Inbox and other unique ones). This list of folders is displayed on the phone's screen. You then may select a sub folder. When done, the script displays a list of voicemails within that directory providing the full caller id and timestamp. If you highlight a message (in any order) and select "Play" from the phone's menu, the message will play over the phone's speaker in the same fashion that, say, a ringtone would. If you highlight a message and press the "Delete" button, the message will be deleted.

Some specifics are that the menus are "CiscoIPPhoneGraphicFileMenu" types which are not supported on all Cisco IP phones. I plan to release a newer version that uses a menu type that is compatible with a given Cisco IP Phone model when oyu provide the model number.

Additionally, for the file to play, I use SoX to create a .raw sound file from Asterisk's .wav. I create it in the /tftpboot directory with the timestamp as the name. I am still working on a reliable mechanism to clean-up these .raw files after the message is played.

I also have this written for configurations where the web server providing the script access, the tftp server providing the phone's config and the Asterisk server are all running on the same machine.
222

3337941g and ad hoc conferences

by spyder40, Monday 30 of April, 2007 [17:35:37 UTC]
We are using 7941g's (SIP41.8-2-2SR1S) and having some trouble with ad hoc conferences. The conferencing works fine but we can't drop off individual legs. Looking at Cisco we should have a "remove" softkey when the conference is going on but don't, only the "EndCall" key. Highlighting any number in the conference then hitting it ends the entire conference as expected Any ideas why the "remove" key doesn't show up? Or where the softkeys are stored/enabled/disabled?
222

3337941g and ad hoc conferences

by spyder40, Monday 30 of April, 2007 [17:30:34 UTC]
We are using 7941g's (SIP41.8-2-2SR1S) and having some trouble with ad hoc conferences. The conferencing works fine but we can't drop off individual legs. Looking at Cisco we should have a "remove" softkey when the conference is going on but don't, only the "EndCall" key. Highlighting any number in the conference then hitting it ends the entire conference as expected Any ideas why the "remove" key doesn't show up? Or where the softkeys are stored/enabled/disabled?
222

333New features on 797X an SCCP 8.2.2

by Chaos2000, Wednesday 11 of April, 2007 [13:48:25 UTC]
Hello,

on the 8.2.2 there are some new features (e.g. <displayOnWhenIncomingCall>), but how to configure it in xml?
Does somebody have an example for the new xml-file.

The <displayOnWhenIncomingCall> can be putted under <vendorConfig> with 0/1 as values
222

333Cisco 7961

by AstroGuru, Thursday 05 of April, 2007 [13:44:15 UTC]
Hi Guyz,

I am almost dead working on 7961. Anyone here got working one? in tcpdump I cannot see the errors. It's communicating but never gets registered. Anyone here knows what's happening.
I have used
cmterm-7941_7961-sip.8-2-2SR1 for firmware upgrade.

What is wrong? Any idea?
Thanks for reading.

my conf file is here SEP-MAC-.conf.xml


<device>

<deviceProtocol>SIP</deviceProtocol>

  <sshUserId>root</sshUserId>

  <sshPassword>cisco</sshPassword>

  <devicePool>
     <dateTimeSetting>
        <dateTemplate>D-M-YA</dateTemplate>
        <timeZone>+270</timeZone>
        <ntps>
             <ntp>
                 <name>172.16.33.30</name>
                 <ntpMode>Unicast</ntpMode>
             </ntp>
        </ntps>
     </dateTimeSetting>
     <callManagerGroup>
        <members>
           <member priority="0">
              <callManager>
                 <ports>
                    <ethernetPhonePort>2000</ethernetPhonePort>
                    <sipPort>5060</sipPort>
                    <securedSipPort>5061</securedSipPort>
                 </ports>
                 <processNodeName>172.16.33.30</processNodeName>
              </callManager>
           </member>
        </members>
     </callManagerGroup>
  </devicePool>
  <sipProfile>
     <sipProxies>
        <backupProxy></backupProxy>
        <backupProxyPort></backupProxyPort>
        <emergencyProxy></emergencyProxy>
        <emergencyProxyPort></emergencyProxyPort>
        <outboundProxy></outboundProxy>
        <outboundProxyPort></outboundProxyPort>
        <registerWithProxy>true</registerWithProxy>
     </sipProxies>
     <sipCallFeatures>
        <cnfJoinEnabled>true</cnfJoinEnabled>
        <callForwardURI>x--serviceuri-cfwdall</callForwardURI>
        <callPickupURI>x-cisco-serviceuri-pickup</callPickupURI>
        <callPickupListURI>x-cisco-serviceuri-opickup</callPickupListURI>
        <callPickupGroupURI>x-cisco-serviceuri-gpickup</callPickupGroupURI>
        <meetMeServiceURI>x-cisco-serviceuri-meetme</meetMeServiceURI>
        <abbreviatedDialURI>x-cisco-serviceuri-abbrdial</abbreviatedDialURI>
        <rfc2543Hold>false</rfc2543Hold>
        <callHoldRingback>2</callHoldRingback>
        <localCfwdEnable>true</localCfwdEnable>
        <semiAttendedTransfer>true</semiAttendedTransfer>
        <anonymousCallBlock>2</anonymousCallBlock>
        <callerIdBlocking>2</callerIdBlocking>
        <dndControl>0</dndControl>
        <remoteCcEnable>true</remoteCcEnable>
     </sipCallFeatures>
     <sipStack>
        <sipInviteRetx>6</sipInviteRetx>
        <sipRetx>10</sipRetx>
        <timerInviteExpires>180</timerInviteExpires>
        <timerRegisterExpires>3600</timerRegisterExpires>
        <timerRegisterDelta>5</timerRegisterDelta>
        <timerKeepAliveExpires>120</timerKeepAliveExpires>
        <timerSubscribeExpires>120</timerSubscribeExpires>
        <timerSubscribeDelta>5</timerSubscribeDelta>
        <timerT1>500</timerT1>
        <timerT2>4000</timerT2>
        <maxRedirects>70</maxRedirects>
        <remotePartyID>false</remotePartyID>
        <userInfo>None</userInfo>
     </sipStack>
     <autoAnswerTimer>1</autoAnswerTimer>
     <autoAnswerAltBehavior>false</autoAnswerAltBehavior>
     <autoAnswerOverride>true</autoAnswerOverride>
     <transferOnhookEnabled>false</transferOnhookEnabled>
     <enableVad>false</enableVad>
     g711ulaw
     <dtmfAvtPayload>101</dtmfAvtPayload>
     <dtmfDbLevel>3</dtmfDbLevel>
     <dtmfOutofBand>avt</dtmfOutofBand>
     <alwaysUsePrimeLine>false</alwaysUsePrimeLine>
     <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
     <kpml>3</kpml>
     <natEnabled>0</natEnabled>
     <natAddress></natAddress>
     <phoneLabel>Charmed</phoneLabel>
     <stutterMsgWaiting>2</stutterMsgWaiting>
     <callStats>false</callStats>
     <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBu
rsts>
     <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
     <startMediaPort>16384</startMediaPort>
     <stopMediaPort>32766</stopMediaPort>
     <sipLines>
        <line button="1">
           <featureID>9</featureID>
           <featureLabel>465</featureLabel>
           <proxy>172.16.33.30</proxy>
           <port>5060</port>
           <name>465</name>
           <displayName>465</displayName>
           <autoAnswer>
              <autoAnswerEnabled>1</autoAnswerEnabled>
           </autoAnswer>
           <callWaiting>3</callWaiting>
           <authName>465</authName>
           <authPassword>123</authPassword>
           <sharedLine>false</sharedLine>
           <messageWaitingLampPolicy>3</messageWaitingLampPolicy>
           <messagesNumber>465</messagesNumber>
           <ringSettingIdle>4</ringSettingIdle>
           <ringSettingActive>5</ringSettingActive>
           <contact>465</contact>
           <forwardCallInfoDisplay>
              <callerName>true</callerName>
              <callerNumber>false</callerNumber>
              <redirectedNumber>false</redirectedNumber>
              <dialedNumber>true</dialedNumber>
           </forwardCallInfoDisplay>
        </line>
        <line button="2">
           <featureID>21</featureID>
           <featureLabel>465</featureLabel>
           <speedDialNumber>465</speedDialNumber>
        </line>
     </sipLines>
     <voipControlPort>5060</voipControlPort>
     <dscpForAudio>184</dscpForAudio>
     <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
     <dialTemplate>DRdialplan.xml</dialTemplate>
  </sipProfile>
  <commonProfile>
     <phonePassword>cisco</phonePassword>
     <backgroundImageAccess>true</backgroundImageAccess>
     <callLogBlfEnabled>2</callLogBlfEnabled>
  </commonProfile>
  <loadInformation>SIP41.8-0-2SR1S</loadInformation>
  <vendorConfig>
     <disableSpeaker>false</disableSpeaker>
     <disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
     <pcPort>0</pcPort>
     <settingsAccess>1</settingsAccess>
     <garp>0</garp>
     <voiceVlanAccess>0</voiceVlanAccess>
     <videoCapability>0</videoCapability>
     <autoSelectLineEnable>0</autoSelectLineEnable>
     <webAccess>1</webAccess>
     <spanToPCPort>1</spanToPCPort>
     <loggingDisplay>1</loggingDisplay>
     <loadServer></loadServer>
  </vendorConfig>
  <versionStamp>1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37</versionStamp>
  <networkLocale>us</networkLocale>
  <networkLocaleInfo>
     <name>us</name>
     <version>5.0(2)</version>
  </networkLocaleInfo>
  <deviceSecurityMode>0</deviceSecurityMode>
  <authenticationURL>http://www/authenticate.php</authenticationURL>
  <directoryURL>http://www/directory.xml</directoryURL>
  <idleURL></idleURL>
  <informationURL>http://www/GetTelecasterHelpText.jsp</informationURL>
  <messagesURL></messagesURL>
  <proxyServerURL></proxyServerURL>
  <servicesURL>http://www/services.xml</servicesURL>
  <dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
  <dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
  <dscpForCm2Dvce>96</dscpForCm2Dvce>
  <transportLayerProtocol>4</transportLayerProtocol>
  <capfAuthMode>0</capfAuthMode>
  <capfList>
     <capf>
        <phonePort>3804</phonePort>
     </capf>
  </capfList>
  <certHash></certHash>
  <encrConfig>false</encrConfig>
</device>

222

333Re: MWI with SIP 8.3

by yansolo90, Wednesday 17 of January, 2007 [15:33:25 UTC]
What does he means with "the extension/SIP ID configured in asterisk" ? The sip username ?
222

3337941 with SIP 8.4 firmware

by betatester, Thursday 30 of November, 2006 [19:12:25 UTC]
Here is a working xml SEPXXX.cnf.xml example. Everything works except MWI (bad notify error). Also, the phone reported "Erro Updating Locale" if you look at the log but everything seems fine. Please note: to enable web access, you have to set "0" on <webAccess>0</webAccess> in the xml. The following example assumed that you already have the phone upgraded with SIP firmware. The "loadInformation" was lefted blank by purpose.

<device>

 <deviceProtocol>SIP</deviceProtocol>

 <sshUserId>cisco</sshUserId>
 <sshPassword>cisco</sshPassword>

 <devicePool>
<dateTimeSetting>
<dateTemplate>M/D/Ya</dateTemplate>
<timeZone>Eastern Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>yourpbxip</name>
<ntpMode>Unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>

    <callManagerGroup>
       <members>
          <member priority="0">
             <callManager>
                <ports>
                   <ethernetPhonePort>2000</ethernetPhonePort>
                   <sipPort>5060</sipPort>
                   <securedSipPort>5061</securedSipPort>
                </ports>
                <processNodeName>yourpbxip</processNodeName>
             </callManager>
          </member>
       </members>
    </callManagerGroup>
 </devicePool>

 <commonProfile>
    <phonePassword></phonePassword>
    <backgroundImageAccess>true</backgroundImageAccess>
    <callLogBlfEnabled>2</callLogBlfEnabled>
 </commonProfile>

 <loadInformation></loadInformation>

 <vendorConfig>
    <disableSpeaker>false</disableSpeaker>
    <disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
    <pcPort>0</pcPort>
    <settingsAccess>1</settingsAccess>
    <garp>1</garp>
    <voiceVlanAccess>0</voiceVlanAccess>
    <videoCapability>0</videoCapability>
    <autoSelectLineEnable>0</autoSelectLineEnable>

    <webAccess>0</webAccess>
    <spanToPCPort>1</spanToPCPort>
    <loggingDisplay>1</loggingDisplay>
    <loadServer></loadServer>
 </vendorConfig>
  <userLocale>
<name>English_United_States</name>
<uid>1</uid>
<langCode>en</langCode>
<version>4.1(3)</version>
<winCharSet>iso-8859-1</winCharSet>
  </userLocale>
 <networkLocale>United_States</networkLocale> 

<networkLocaleInfo>
<name>United_States</name>
<uid>64</uid>
<version>4.1(3)</version>
</networkLocaleInfo>

 <deviceSecurityMode>1</deviceSecurityMode>

 <authenticationURL>http://yourpbxip/cisco/services/authentication.php</authenticationURL>
 <directoryURL>http://yourpbxip/cisco/services/PhoneDirectory.php</directoryURL>
 <idleURL></idleURL>
 <informationURL>http://yourpbxip/cisco/services/help.php</informationURL>
 
 <messagesURL></messagesURL>
 <proxyServerURL></proxyServerURL>
 <servicesURL>http://yourpbxip/cisco/services/index_cisco.php</servicesURL>
 <dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
 <dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
 <dscpForCm2Dvce>96</dscpForCm2Dvce>

 <transportLayerProtocol>4</transportLayerProtocol>

 <capfAuthMode>0</capfAuthMode>
 <capfList>
    <capf>
       <phonePort>3804</phonePort>
    </capf>
 </capfList>

 <certHash></certHash>
 <encrConfig>false</encrConfig>
 
  <sipProfile>
    <sipProxies>
       <backupProxy></backupProxy>
       <backupProxyPort></backupProxyPort>
       <emergencyProxy></emergencyProxy>
       <emergencyProxyPort></emergencyProxyPort>
       <outboundProxy></outboundProxy>
       <outboundProxyPort></outboundProxyPort>
       <registerWithProxy>true</registerWithProxy>
    </sipProxies>

    <sipCallFeatures>
       <cnfJoinEnabled>true</cnfJoinEnabled>
       <callForwardURI>x--serviceuri-cfwdall</callForwardURI>
       <callPickupURI>x-cisco-serviceuri-pickup</callPickupURI>
       <callPickupListURI>x-cisco-serviceuri-opickup</callPickupListURI>
       <callPickupGroupURI>x-cisco-serviceuri-gpickup</callPickupGroupURI>
       <meetMeServiceURI>x-cisco-serviceuri-meetme</meetMeServiceURI>
       <abbreviatedDialURI>x-cisco-serviceuri-abbrdial</abbreviatedDialURI>
       <rfc2543Hold>false</rfc2543Hold>