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  • Juan Ortega, Thu 15 of May, 2008 [10:33 UTC]: Hi everybody, I'm Juan, an ITCom student, and I need to know what basic elements I need to create a VoIP network. Can anybody helpme, please?,Thank you very much
  • gineta, Wed 14 of May, 2008 [03:58 UTC]: any here not fine the configuration of firewall juniper -screem for VOIP asterisk????
  • Anoop Prabhakaran, Tue 13 of May, 2008 [12:16 UTC]: I am developing Asterisk IVR, Whenever i make a internation call to the IVR system, the DTMF is not getting detected properly, this happens only for the first time, second call onwards system works fine. why this is happening
  • joe, Mon 12 of May, 2008 [04:27 UTC]: Is there an opensource browser based softphone, or a system like Busta where everything is not manages through their website?
  • Nick Barnes, Fri 09 of May, 2008 [11:36 UTC]: Christopher - yesterday I tried an Asterisk install on a CentOS 5.1 box with stock GUI and it all worked fine. Sorry I can't help.
  • aero, Fri 09 of May, 2008 [08:20 UTC]: can someone help me out on this, i tried to play some sound files on my asterisk box and this is the error message i got. WARNING[4429]: format_wav.c:169 check_header: Unexpected freqency 22050 May 8 11:17:39 WARNING[4433]: codec_gsm.c:194 gsmtolin_fra
  • Christopher Faust, Thu 08 of May, 2008 [14:15 UTC]: I beleive that I may have to change something in the xserver configuration. Please advise
  • Christopher Faust, Thu 08 of May, 2008 [14:14 UTC]: Everything was perfect. In the bios I have increased the memory allocated Still receive input not supported on my display.
  • Christopher Faust, Thu 08 of May, 2008 [14:13 UTC]: This would not be my main box. I am doing some testing to see if I can install zaptel and asterisk 1.4 on a full centos 5.1 box with development software Its bizzare, because before I went through the asterisk and zaptel installation everything was perfe
  • Nick Barnes, Thu 08 of May, 2008 [13:44 UTC]: Christopher - I can't see any way in which an Asterisk installation would muck your GUI, but remember that it is advised not to use a GUI on an Asterisk box anyway.
Server Stats
  • Execution time: 0.64s
  • Memory usage: 2.33MB
  • Database queries: 37
  • GZIP: Disabled
  • Server load: 0.64

SIP Express Router

SER - SIP Express Router
http://developer.berlios.de/projects/ser/

Original Website: http://www.iptel.org/ser/
Website of the OpenSER fork: http://openser.org/
SIP Express Router (ser) is a high-performance, configurable, free SIP ( RFC3261 ) server . It can act as registrar, proxy or redirect server. SER features an application-server interface, presence support, SMS gateway, SIMPLE2Jabber gateway, RADIUS/syslog accounting and authorization, server status monitoring, FCP security, etc. Web-based user provisioning, serweb, available.

Its performance allows it to deal with operational burdens, such as broken network components, attacks, power-up reboots and rapidly growing user population.

SER's configuration ability meets needs of a whole range of scenarios including small-office use, enterprise PBX replacements and carrier services.
Which version of SER is documented here?

This Wiki covers both the stable and the development branch of SER. When adding new commands, modules, and options, please also add a note on *when* this was added so that users may compare with their version date.

  • SER is an Open Source SIP server, licensed under the GPL
  • SER supports SIP over TCP and UDP according to RFC 3261
  • SER supports ENUM
  • SER supports several NAT support mechanisms
  • SER may interoperate with the jabber instant messaging architecture
  • SER supports multiple user DNS domains in parallell
  • SER is extensible with modules for various additional functions
  • SER supports DNS SRV lookups

SER supports SIP connections with more features and more scalability than Asterisk. Normally, SER would be used in conjunction with Asterisk when a SIP phone needed to connect to the PSTN.

SER modules

If "experimental" this applies to the 0.8.11 release.

Ser pages

Ser web interfaces

Platforms

  • ser has been written in ANSI C. It has been extensively tested on PC/Linux and Sun/Solaris. Ports to BSD and IPAQ/Linux exist.
  • SIPatH Project - porting ser to the mipsel architecture OpenWRT - Summary - Website
  • SER OS Platforms - What Operating Systems SER works with.
  • SER Linksys NSLU2

References

Resources


Created by jht2, Last modification by linkx on Thu 28 of Feb, 2008 [11:03 UTC]

Comments

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Re: sip or voip

by satish patel on Wednesday 16 of April, 2008 [12:36:21 UTC]
I have integrate SER + Asterisk http://linuxbug.org/index_files/projects.html

Satish Patel
feel free contact :- satish.lx@gmail.com

Aliwei IAD work with Asterisk configuration example

by Robin on Tuesday 08 of April, 2008 [15:14:24 UTC]
service password-encryption
service timestamps debug

hostname Aliwei

enable password 7 z1SUMF6JUPe1pf5P

username admin password 7 y1CRyR4p8kJudB
username sapian password 7 VhVcZNYcNKqjggK
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8

voice vad-time 300
voice rtp start-port 10001
voice pattern sip

interface fastethernet 0/0
ip address 172.16.2.11 255.255.255.0

interface ethernet 0/0
ip address 192.168.140.11 255.255.255.0

interface async 0/0

line vty 0 31

sip-ua
transport udp
ip local 172.16.2.11
ip media 172.16.2.11
registrar ipv4:172.16.2.1
mode end-to-end
remote-port 5060
enable-use-rport
ring-mode ring-mode-183
enable-same-number

ip route 0.0.0.0 0.0.0.0 172.16.2.1
ip name-server 172.16.2.1

voice-port 1/0
port-type FXO
busytone 480 300 300
signal groundStart
pre-dial-delay 1
enable-rfc2833
fxo-number 3113343083
local_zone_number 183

voice-port 1/1
port-type FXO
busytone 400 300 300
pre-dial-delay 1
enable-rfc2833
local_zone_number 183

voice-port 1/2
port-type FXO
pre-dial-delay 1
enable-rfc2833

voice-port 1/3
port-type FXO
pre-dial-delay 1
enable-rfc2833

voice-port 1/4
port-type FXO
pre-dial-delay 1
enable-rfc2833

voice-port 1/5
port-type FXO
pre-dial-delay 1
enable-rfc2833

voice-port 1/6
port-type FXO
pre-dial-delay 1
enable-rfc2833

voice-port 1/7
port-type FXO
pre-dial-delay 1
enable-rfc2833

dial-peer voice 1 pots
description fxo0
destination-pattern T
port 1/0
no register e164
no vad
forward-digits all
caller-id

dial-peer voice 2 pots
destination-pattern T
port 1/1
no register e164
no vad
forward-digits all
caller-id

dial-peer voice 3 pots
destination-pattern T
port 1/2
no register e164
no vad
caller-id
call-forward 673
shutdown

dial-peer voice 4 pots
destination-pattern T
port 1/3
no register e164
no vad
caller-id
call-forward 674
shutdown

dial-peer voice 5 pots
destination-pattern T
port 1/4
no register e164
no vad
caller-id
call-forward 675
shutdown

dial-peer voice 6 pots
destination-pattern T
port 1/5
no register e164
no vad
caller-id
call-forward 676
shutdown

dial-peer voice 7 pots
destination-pattern T
port 1/6
no register e164
no vad
caller-id
call-forward 677
shutdown

dial-peer voice 8 pots
destination-pattern T
port 1/7
no register e164
no vad
caller-id
call-forward 678
shutdown

dial-peer voice 100 voip
destination-pattern T
codec g711alaw
voice-class codec 1
session target ipv4:172.16.2.1
dtmf-relay cisco-rtp
no vad
transfer-mode blind

ntp source ethernet 0/0
ntp server 172.16.2.1
ntp clock-period 0
media-wait-for-connect
tone-generate busy f1 440 f2 0 f3 0 f4 0 on-time 500 off-time 500 numCad 1 repCounter 0 on_time_2 0 off_time_2 0 on_time_3 0 off_time_3 0 tonePwr1 12 tonePwr2 0 tonePwr3 0 tonePwr4 0
tone-generate ring-back f1 440 f2 0 f3 0 f4 0 on-time 500 off-time 500 numCad 1 repCounter 0 on_time_2 0 off_time_2 0 on_time_3 0 off_time_3 0 tonePwr1 12 tonePwr2 0 tonePwr3 0 tonePwr4 0
tone-generate dial f1 380 f2 0 f3 0 f4 0 on-time 2400 off-time 2400 numCad 1 repCounter 1 on_time_2 0 off_time_2 0 on_time_3 0 off_time_3 0 tonePwr1 2 tonePwr2 0 tonePwr3 0 tonePwr4 0
trunk-to-voip
voice-bind sip

end

How to integrate Asterisk with SER

by satish patel on Thursday 10 of January, 2008 [14:40:09 UTC]
I have to integarte SER with Asterisk means SIP client Register on SER and make call to PSTN with the help of asterisk and also can do meet me confrance and many PBX feature is it possible and How

sip or voip

by habib hayek on Thursday 10 of January, 2008 [13:46:31 UTC]
hi my name is habib and looking to this informations plz:
first i am new user of sip and voip i am in canada and i want to call our office out of canada in dubai for example if i have an internet connection or a dsl connection in my office there i ahve a ata device here in canada and dubai and i have sip software on both computer what do i need to call dubai over voip or sip and i want to connect my land line in dubai to my device or modem if my computer so i can make calls to dubai via voip or sip but i will pay local call not international because i am using my dubai land line is that posssible or i am dreaming i hope i can find something thx for u all plz send me email to hayik@hotmail.com

Use ser for Prepaid internet phone

by Phan Van Duc on Wednesday 08 of November, 2006 [06:35:02 UTC]
Have any company use Ser to provide Prepaid internet phone service?
my company is looking for Prepaid internet phone solution.
Thank you, phanvanduc@gmail.com

Ser+Asterisk

by siqhamo on Sunday 03 of September, 2006 [12:39:26 UTC]
R there any detailed docs on Ser+asterisk integration .

SER help available

by Mike on Friday 24 of March, 2006 [21:03:03 UTC]
If you're trying to implement SER on your VoIP network, feel free to contact us voipinfo@idv.net or through our website http://www.idv.net
We have extensive background in implementing SER with Cisco voice gateways, Asterisk, ATAs, etc.
We can also help with custom development.


we pay for your help

by reza on Saturday 04 of February, 2006 [12:28:10 UTC]
we are looking for a linux based programmer for improving our SIP proxy Project
sip professional programmer please contact me at grkashani@yahoo.com

Explication please

by David Goldstein on Friday 11 of November, 2005 [02:28:36 UTC]
Hi,

Can anyone expand on this comment (SER supports SIP connections with more features and more scalability than Asterisk. Normally, SER would be used in conjunction with Asterisk when a SIP phone needed to connect to the PSTN) so that a first-time Asterisk novice can understand the logic of how SER and Asterisk work together to connect to the PSTN? What functions/processes is Asterisk responsible for and what functions/processes is SER responsible for? What handoffs are going on? Thank You, dave_rep@yahoo.com


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