Html/js/flash/air SIP Clients?

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Dear list, I am looking for an open source SIP client(or any SDK) that can work on a browser. It may be based html5, javascript, flash, adobe air. I have done some research myself and I would like to ask the community if they have any further hints for me. Real life experience would be awesome.

Thanks, Regards, Arstan Jusupov

Asterisk Users 3.1 years ago 6 Answers

Problem With Callfile And CDR

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Good afternoon list.

I am experiencing a problem with the CDR and callfiles. What is happening is this: When generating a call with a callfile, everything works perfectly, but the CDR is recorded in the table when they answer the call destination. The field disposition is being recorded correctly, but the duration field is marked with the ring time and billsec is marked with 0. This just happens to connections through callfiles. Yes, the call is working usually remains. I did several tests with durations from seconds to 20 minutes.

I tested in two servers. With an Asterisk 1.4.44 (debian 5 64bits),…

Asterisk Users 3.1 years ago 6 Answers

Asterisk Dahdi 1.6.2.23 iaxmodem

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Hello, I have anolog lines coming throug Dahdi to Asterisk Server, one of the anolog lines is used for fax line. I received fax fine without any problems using Iaxmodem with Hylafax Server. Outgoint fax is the problem, when IAXMODEM dial out using Dahdi channel, dahdi answers and start to dial the outside number however Iaxmodem thinks that dahdi is the remote fax machine and starts sending fax data eventually giving up. Here is my dialplan in extensions.conf

[fax-out]
exten => _XXXXXXX,1,Set(__SIP_CODEC=alaw)
exten => _XXXXXXX,2,Dial(dahdi/g3)
If the remote machine answers within the first ring, the outgoing fax works fine,…

Asterisk Users 3.1 years ago 12 Answers

Planned Service Outage For Community Services On August 2, 2012

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On August 2, 2012 during a window from 10:00AM to 11:00AM (Central Daylight Time, GMT-5), the core routers that provide connectivity through to all Asterisk community services will be swapped out. The actual service interruption may only last a few minutes, but could last longer. These services could be unavailable during most, if not all, of this time window. Once the move is complete, the services will be available again, with no user-visible changes. The services affected include: bamboo.asterisk.org code.asterisk.org downloads.digium.com downloads.asterisk.org git.asterisk.org issues.asterisk.org packages.asterisk.org reviewboard.asterisk.org svn.asterisk.org svnview.digium.com wiki.asterisk.org

Asterisk Users 3.1 years ago 0 Answers

CallerID

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When I use a call file to start a call I set the CallerID: field and the polycom phone shows the correct information.

When I use a call file to start a conf call I set the CallerID: field and my polycom phones show "asterisk" not the callerID I have set.

Is there something additional needed to set the CallerID in this case? I am also using the "Local" channel does that make a difference?

Jerry

Asterisk Users 3.1 years ago 0 Answers

App_swift 3 And Asterisk 1.8.13.0 Fails With Undefined Symbol: Swift_port_close

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All,

I am experiencing this same issue. it seems that you were able to resolve it offline. Could you by any chance post the solution.

telephonics1*CLI> module load app_swift.so Unable to load module app_swift.so Command 'module load app_swift.so' failed. [Aug 1 05:01:01] WARNING[28635]: loader.c:458 load_dynamic_module: Error loading module 'app_swift.so': /usr/lib/asterisk/modules/app_swift.so: undefined symbol: swift_port_close [Aug 1 05:01:01] WARNING[28635]: loader.c:848 load_resource: Module 'app_swift.so' could not be loaded. telephonics1*CLI>

I suspect an issue when linking app_swift.so. Here is how the module is linked:

gcc -shared -Xlinker -x -o app_swift.so -L/opt/swift/lib -L/usr/lib -lswift -lceplang_en -lceplex_us app_swift.o

However, ldd shows the module doesn't list libswift.so as a dependency:

# ldd app_swift.so…

Asterisk Users 3.1 years ago 1 Answer

Problem Provisioning Cisco SPA303

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Hello. I've got a Cisco SPA303 that I'm trying to provision via http. I noticed that this device looks very similar to a PAP2T, so I used that as a template for my provisioning file.

However, the result is less than stellar. Line 1 registers and works. However, lines 2 and 3 also register as line 1, effectively giving me a 1-line phone with 3 buttons.

Also, the line name is the same on all 3 phone lies.

I've looked on Cisco's website and Googled around, but I can not find a true example of a provisioning file for this device. Anything I…

Asterisk Users 3.1 years ago 2 Answers