- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] event i_media_error 988 Incomplete offer/answer
by Joel Mange 29 Dec '10

29 Dec '10

Problem resolved -- thank you for pointing me in the right direction. Was following a Cisco application guide for router configuration, which specified the following voice service voip command: signaling forward unconditional. This enabled SIP GTD. Following command disabled SIP GTD and corrected issue for me: signaling forward none. Thanks again. --- On Tue, 12/28/10, Joel Mange <telconettech(a)yahoo.com> wrote: From: Joel Mange <telconettech(a)yahoo.com> Subject: event i_media_error 988 Incomplete offer/answer To: freeswitch-users(a)lists.freeswitch.org Date: Tuesday, December 28, 2010, 7:35 PM Hello. New to FS and in need of help. Environment is as follows: Linksys SPA-962 (x94140) <---> FS <---sip---> Cisco3845 <---qsig---> TDM PBX <--> TDM phone (x91237) Can complete call from TDM phone to Linksys SPA-962. Can't complete call from Linksys to TDM phone: TDM phone rings, but call fails when it goes off-hook. Getting the following sofia error: nua(0x8b60cf0): event i_media_error 988 Incomplete offer/answer. I think the above message tells me that the other side of the call sent an invalid SDP. Having trouble identifying the invalid SDP and making necessary correction. Siptrace and router running-config at: http://pastebin.freeswitch.org/14896 Any help appreciated.

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FreeSWITCH Conference Call Today
by Michael Collins 29 Dec '10

29 Dec '10

Hey gang, Here is today's agenda page: http://wiki.freeswitch.org/wiki/FS_weekly_2010_12_29 It is light since we have lots of people on vacation and otherwise taking holiday. If everyone is up to it I'd like to do what we did last week and have everyone help out with some documentation. We have several things that we can work on: New jitterbuffer API FIFO params and vars Voicemail page needs to be updated Thanks! Talk to you shortly. -MC

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event i_media_error 988 Incomplete offer/answer
by Joel Mange 29 Dec '10

29 Dec '10

Hello. New to FS and in need of help. Environment is as follows: Linksys SPA-962 (x94140) <---> FS <---sip---> Cisco3845 <---qsig---> TDM PBX <--> TDM phone (x91237) Can complete call from TDM phone to Linksys SPA-962. Can't complete call from Linksys to TDM phone: TDM phone rings, but call fails when it goes off-hook. Getting the following sofia error: nua(0x8b60cf0): event i_media_error 988 Incomplete offer/answer. I think the above message tells me that the other side of the call sent an invalid SDP. Having trouble identifying the invalid SDP and making necessary correction. Siptrace and router running-config at: http://pastebin.freeswitch.org/14896 Any help appreciated.

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bridge application dial string
by rajkumar 29 Dec '10

29 Dec '10

Hi, Normally we can specify the channel in bridge application. Is it possible to specify the dial plan extension in bridge dial string? regards rajkumar k -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/bridge-application-dial-strin… Sent from the freeswitch-users mailing list archive at Nabble.com.

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calling card app design decision
by Rafqat . 29 Dec '10

29 Dec '10

Hi, I am wanting to develop a calling card service using FreeSWITCH. It's seems FreeSWITCH is very flexible in terms of how one could go about implementing this service. I am intending to use mod_nibblebill and mod_lcr to do the core stuff. I would like to use Java for any programming (as it's the language I am most familiar with) however, I am unsure whether to take an event socket based approach to the application or to use something like mod_java. Any advice/suggestions will be most welcome. Cheers Raf

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files.freeswitch.org
by Brian West 29 Dec '10

29 Dec '10

Our CDN is having some issues in the past 4 hours so I have disabled it for now please let me know if you continue to have issues with files.freeswitch.org Thanks, Brian

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ANN : PySWITCH Release – 0.1alpha
by Godson Gera 29 Dec '10

29 Dec '10

Hi All, I am glad to announce the first alpha release of PySWITCH. http://pyswitch.sf.net The idea of PySWITCH is to offer a complete library to Python and Twisted programmers for interacting with FreeSWITCH using EventSocket interface. The target is to cover all FreeSWITCH API commands and Dialplan tools. PySWITCH handles all the low level details in executing FreeSWITCH commands, so the programmer can easily concentrate on quickly building FreeSWITCH applications. As an example, the API functions offered by PySWITCH often executes many FreeSWITCH commands under the hood and finally returns the desired result. Suppose you execute a background job, PySWITCH API will automatically wait and catch the backgroundjob event parse the result and will fire the deferred. The current release covers good amount of API commands and a few Dialplan tools. The protocol communication issues are ironed out. It has a nice event call back interface. I’ll present its usage in couple of tutorials soon. -- Thanks & Regards, Godson Gera http://godson.in

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FreeTDM hangs on Partial PRI
by Phone 28 Dec '10

28 Dec '10

We are using FreeSWITCH Version 1.0.head (git-8825b6e 2010-11-28 17-15-39 -0500) WANPIPE Release: 3.5.14 Hardware is Sangoma A101 Partial PRI D-chan=24 B-chans=16-23 We are having trouble with the Partial PRI (channel 16) getting stuck in a state where it cannot take calls. This happens intermittently. The system may run several days or even a week or two without having the problem. It seems that the channel is getting stuck in some odd state when trying to hang up. To resolve this, we unload the FreeTDM mod, restarting Wanrouter, and then reload the FreeTDM mod. This clears the channels and allows the system to take calls normally again. In testing we have also observed the following conditions: 1. Place a call in. Cannot complete. Hang up. 2. Place a call in. It completes and works correctly. 3. Place a call in. Cannot complete. Hang up. 4. Place a call in. It completes and works correctly. The above pattern can be repeated multiple times, with the same result. Here is some log info: 2010-12-26 20:51:20.868083 [DEBUG] ftmod_sangoma_boost.c:1889 RX EVENT (N): CALL_START:(80) [w1g16] CSid=0 Seq=390 Cn=[JONES J ] Cd=[7878030] Ci=[8015542086] Rdnis=[] 2010-12-26 20:51:20.868083 [DEBUG] ftmod_sangoma_boost.c:274 Channel 1:1 ~ 1:16 is already in use in state TERMINATING 2010-12-26 20:51:20.868083 [ERR] ftmod_sangoma_boost.c:1008 s1c16: rejecting incoming call in channel state TERMINATING 2010-12-26 20:51:20.868083 [DEBUG] sangoma_boost_client.c:252 TX EVENT (N): CALL_START_NACK:(82) [s1c16] Rc=44 CSid=0 Seq=5 2010-12-26 20:51:28.563720 [DEBUG] ftmod_sangoma_boost.c:1889 RX EVENT (N): CALL_START_NACK:(82) [s1c16] Rc=16 CSid=0 Seq=391 2010-12-26 20:51:28.563720 [WARNING] ftmod_sangoma_boost.c:811 [s1c1][1:16] Why bother changing state from TERMINATING to TERMINATING Hope this information is useful. Thanks.

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Cross-compilation crash on src/mod/languages/mod_perl
by mazilo 28 Dec '10

28 Dec '10

I ran into this http://pastebin.com/1JNVfwM1 problem when trying to cross-compile FS git (2010/12/27) src/mod/languages/mod_perl for an ARM platform. The error (line# 6) indicates the compilation was trying to use host perl and its library. Can FS developers please take a look and see if that can be resolved? Thanks. ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Cross-compilation-crash-on-sr… Sent from the freeswitch-users mailing list archive at Nabble.com.

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group_confirm - Question
by Aloysius Lloyd 28 Dec '10

28 Dec '10

Hi All, Dial plan below use group_confrim, work for both internal and external (PSTN) Calls without any issues <action application="bridge" data="{ignore_early_media=true,* group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/followme/press-1-to-accept.wav,group_confirm_key=1 *}user/${dialed_extension}@ ${domain_name},sofia/gateway/voipms/${user_data(${dialed_extension}@${domain_name} var followme_number)}"/> ----- But I would like to use the group_confirm feature for external PSTN (Calls only). The following Dial plan is not working and the console log below. <action application="bridge" data="{ignore_early_media=true}user/${dialed_extension}@${domain_name},{* group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/followme/press-1-to-accept.wav,group_confirm_key=1 *}sofia/gateway/voipms/${user_data(${dialed_extension}@${domain_name} var followme_number)}"/> *Console Log* 2010-12-26 15:23:21.796308 [ERR] switch_core_session.c:380 Could not locate channel type {group_confirm_file= 2010-12-26 15:23:21.796308 [ERR] switch_ivr_originate.c:2614 Cannot create outgoing channel of type [{group_confirm_file=] cause: [CHAN_NOT_IMPLEMENTED] 2010-12-26 15:23:21.796308 [ERR] switch_core_session.c:380 Could not locate channel type group_confirm_key=1}sofia 2010-12-26 15:23:21.796308 [ERR] switch_ivr_originate.c:2614 Cannot create outgoing channel of type [group_confirm_key=1}sofia] cause: [CHAN_NOT_IMPLEMENTED] -------- Any help is appreciated. Thanks Lloyd

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