by mondster, Monday 14 of July, 2008 [15:45:30 UTC]
Hi,
Is this still open for Administration?
Thanks.
Raymond
222
333Asterisk Programming and Admin work
by sunkefer, Thursday 22 of May, 2008 [16:04:50 UTC]
I have some small internal projects to complete our asterisk server with basic call routing and ivr's. And I also have a potential programming project that covers a paid phone service that is straighforward astbill or asterisktobilling + asterisk development.
If the administration side or the programming side is something you are interested in learning more of please feel free to contact me with your going rate. While we are in Buenos Aires, Arg. I am willing to work with remote people on much/ all of this.
You can email us at jobs AT sourcesouth DOT com
222
333fix: asterisk voip2pots linux rejects some calls
by gurkenschaeler, Friday 22 of February, 2008 [19:34:03 UTC]
this is a paid gig.
we have a recent trixbox installation with Asterisk 1.2.13-BRIstuffed-0.3.0-PRE-1v for a small office. one landline number that goes into the box and maybe 6 internal voip phones. now everything works, BUT callers from one certain POTS provider get rejected. i.e. they call the +431... number and get a busy signal. there's some obscure message in the logs, but nothing more.
grep 'Rejecting call' /var/log/asterisk/full|grep -v "from ''"|sed -e 's/^.*from .//g'| sed -e 's/. does not.*$//'|sort -n|uniq -c
all the caller id numbers are from an Astrian provider called Tele2/UTA (www.tele2.at or www.uta.at)
we need this fixed.
do not know whether it's something in asterisk itself or bristuff ...
please get in touch at uplink DOT team AT gmail DOT com
222
333Linux/Asterisk guy wanted!
by manhattan1, Monday 31 of December, 2007 [04:34:32 UTC]
The Linux Company Open Computing a division under the Scandinavian company M.K. Andersen Outsourcing is hiring:
1 Junior Programmer with good English comm. skills.
1 Senior Programmer with min. 3-5 years of expirence.
Applicants can contact us on e-mail: info@opencomputing.dk
Mark: Jr. Programmer or Sr. Programmer.
Location: Cebu City, the Philippines.
www.opencomputing.dk & www.kirkoutsourcing.com
222
333VPN for VoIP Blocking
by jenniferhan, Wednesday 12 of December, 2007 [03:25:58 UTC]
Somebody use VPN to solve the VoIP Blocking issue. But it seems not a good way to solve the voip blocking issue. Because VPN will take more bandwidth and will take effection on the Voice Quality
Currently I am using the VGCP, a new solution to solve the VoIP Blocking issue. Following is theirs website:
http://www.speed-voip.com/index-36.html
If any of you have interested, you may try to use it to solve your VoIP Blocking problems. Thanks.
Andy
andywong-01@hotmail.com
222
333Asteria Solutions Group Hiring Developers
by kpotter, Thursday 11 of October, 2007 [17:06:36 UTC]
We are currently hiring developers to do web based server side scripting (PHP, ASP, JSP), client side scripting in Javascript, SQL, Linux/UNIX, GUI development in C++. Experience with Telecom/PBX a plus. Send resumes to resumes@asteriasgi.com.
222
333Senior Developer / Programmer
by Callmotions, Thursday 23 of August, 2007 [20:07:55 UTC]
We are a dynamic fast growing service provider to the small and medium sized business community in NYC. The company is looking for a senior developer who still enjoys programming to lead the next stage of development for a web-based service application. The opportunity is significant for the right entrepreneurial person.
B.S./M.S. in Computer Science or equivalent work experience
Hands-on experience with Asterisk
Understanding of telephony technologies such as VoIP, IVRs, audio formats helpful
Cisco knowledge a plus.
Experience with Databases and Server side programming
Linux experience essential
Familiarity with Open Source Environments
Self motivated, highly proactive and reliable. Team Player. Ability to excel under pressure.
Experience with software development methodology, source code control bug tracking system.
What you need to apply:
Entrepreneurial Spirit
2+ years of PHP programming experience (PHP5 beneficial)
1+ year leading a team or mentoring other engineers
Linux, Unix expertise
MySQL experience
Contact: Roger Gins at sd@callmotions.com
222
333Consultant/Freelancer
by 051583, Wednesday 25 of July, 2007 [23:30:27 UTC]
We are looking for a consultant/freelancer to assign medium to large sized jobs. We have a number of different projects currently in queue. Please review our requirements below. Please send contact information, rates, etc. to 051583@gmail.com
Asterisk knowledge is a must
VPN tunnel design and troubleshooting, specifically SSL tunnels
Understanding of various voice codec(s) like g711 and 729
Understanding of SIP and IAX protocol messages
Understanding of TDM technologies
QoS implementations
Practical knowledge of networking and NAT
MySQL and SQL query construction and database management
Understanding of T.38 protocol a plus
Understanding of DUNDi a plus
Understanding of SNMP a plus
Ability to understand, design, and work with advanced dial plans
Experience with Asterisk 1.4 and 1.2
Experience with linux a must
Familiarity with openvpn, mysql, sipp, rsync, iptables, cron, sendmail, ssh, vsftp, netopia enterprise routers, cisco IOS, zenoss, or wireless point to point a plus
222
333Consultant/Freelancer
by 051583, Wednesday 25 of July, 2007 [15:35:57 UTC]
We are looking for a consultant/freelancer to assign medium to large sized jobs. We have a number of different projects currently in queue. Please review our requirements below. Please send contact information, rates, etc. to 051583@gmail.com
Asterisk knowledge is a must
VPN tunnel design and troubleshooting, specifically SSL tunnels
Understanding of various voice codec(s) like g711 and 729
Understanding of SIP and IAX protocol messages
Understanding of TDM technologies
QoS implementations
Practical knowledge of networking and NAT
MySQL and SQL query construction and database management
Understanding of T.38 protocol a plus
Understanding of DUNDi a plus
Understanding of SNMP a plus
Ability to understand, design, and work with advanced dial plans
Experience with Asterisk 1.4 and 1.2
Experience with linux a must
Familiarity with openvpn, mysql, sipp, rsync, iptables, cron, sendmail, ssh, vsftp, netopia enterprise routers, cisco IOS, zenoss, or wireless point to point a plus
222
333Asterisk Developer Position
by wallyhts, Wednesday 13 of June, 2007 [14:20:20 UTC]
I am in the need of asterisk developers. Please submit your resume to davidwallace@worldbridgepartners.com
for confidential consideration.
What I need to do is record telephone calls that are made on regular phones on a regular telephone line, and record the calls somehow using a VoIP system.
I wish to use this system using a PBX and VoIP phone adapters without using an actual VoIP external line. The line that comes into my house is a POTS line. The setup does not look like it will work to me, but I have seen this exact setup where I work so I am very confident that it does work. I wish to do it like this:
Regular Phone ----> phone line splitter -----> VoIP Phone Adapter -----> PBX
...
There are two ways of doing this which immediately spring to mind (there will be many many more).
The first is to use the LOCAL channel and then initiate the call from within that. This should create a new CDR from that point in the call - I know we've used it for this reason, but it was a while ago, so I can't remember if we needed to do anything else.
The second way would be to log the times to the database that the dial started and when (if) the call is bridged. FUNC_ODBC is your friend!
Those should push you in the right direction, but it is worth bearing in mind that there are a number of issues with CDRs - it is not recommended that you use them as your only source of billing information unless you are absolutely sure that you are recording exactly what you want.
...
Hi,
I wrote a DialScript that receibe a call from the PSTN, ask for destination (with a READ) , and then make a call using a Dial to the final destination.
I need to have a CDR that represent ONLY the second leg of the call (with your own billsec, and disposition, etc)
Anyone know how to do this??? HELP ME!!!!
Thanks.
...
I would like be sure that when using "exterhost" parameter, although it is not recommended for production environments, the DNS lookup made to determine the IP for "exterhost" will not use the DNS server posted on /etc/resolver. Because if I am running also a name server inside my local network the result for this query will never be the public IP for the Asterisk behind the NAT.
Comments
333Re: Asterisk Programming and Admin work
Is this still open for Administration?
Thanks.
Raymond
333Asterisk Programming and Admin work
If the administration side or the programming side is something you are interested in learning more of please feel free to contact me with your going rate. While we are in Buenos Aires, Arg. I am willing to work with remote people on much/ all of this.
You can email us at jobs AT sourcesouth DOT com
333fix: asterisk voip2pots linux rejects some calls
this is a paid gig.
we have a recent trixbox installation with Asterisk 1.2.13-BRIstuffed-0.3.0-PRE-1v for a small office. one landline number that goes into the box and maybe 6 internal voip phones. now everything works, BUT callers from one certain POTS provider get rejected. i.e. they call the +431... number and get a busy signal. there's some obscure message in the logs, but nothing more.
grep 'Rejecting call' /var/log/asterisk/full|grep -v "from ''"|sed -e 's/^.*from .//g'| sed -e 's/. does not.*$//'|sort -n|uniq -c
gives something like
2 0424229106
3 0512333414
1 0577001255
2 0577001256
1 0577001668
24 0720308837
all the caller id numbers are from an Astrian provider called Tele2/UTA (www.tele2.at or www.uta.at)
we need this fixed.
do not know whether it's something in asterisk itself or bristuff ...
please get in touch at uplink DOT team AT gmail DOT com
333Linux/Asterisk guy wanted!
1 Junior Programmer with good English comm. skills.
1 Senior Programmer with min. 3-5 years of expirence.
Applicants can contact us on e-mail: info@opencomputing.dk
Mark: Jr. Programmer or Sr. Programmer.
Location: Cebu City, the Philippines.
www.opencomputing.dk & www.kirkoutsourcing.com
333VPN for VoIP Blocking
Currently I am using the VGCP, a new solution to solve the VoIP Blocking issue. Following is theirs website:
http://www.speed-voip.com/index-36.html
If any of you have interested, you may try to use it to solve your VoIP Blocking problems. Thanks.
Andy
andywong-01@hotmail.com
333Asteria Solutions Group Hiring Developers
333Senior Developer / Programmer
B.S./M.S. in Computer Science or equivalent work experience
Hands-on experience with Asterisk
Understanding of telephony technologies such as VoIP, IVRs, audio formats helpful
Cisco knowledge a plus.
Experience with Databases and Server side programming
Linux experience essential
Familiarity with Open Source Environments
Self motivated, highly proactive and reliable. Team Player. Ability to excel under pressure.
Experience with software development methodology, source code control bug tracking system.
What you need to apply:
Entrepreneurial Spirit
2+ years of PHP programming experience (PHP5 beneficial)
1+ year leading a team or mentoring other engineers
Linux, Unix expertise
MySQL experience
Contact: Roger Gins at sd@callmotions.com
333Consultant/Freelancer
333Consultant/Freelancer
333Asterisk Developer Position
for confidential consideration.