March 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Core dump at mod_sofia.c:966
by Denis Gasparin 10 Mar '14

10 Mar '14

Hi. I'm using Freeswitch master branch with webrtc on Debian 7. After two days without any problem, today I've got a core dump with the following error: Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". Core was generated by `/u/freeswitch/bin/freeswitch -ncwait -rp'. Program terminated with signal 7, Bus error. #0 sofia_read_frame (session=0x12aea08, frame=0x7f067e9cb038, flags=2, stream_id=0) at mod_sofia.c:966 966 sofia_clear_flag_locked(tech_pvt, TFLAG_READING); I'm not able to reproduce but i have the last log lines before the core dump: recv 362 bytes from ws/[ip_address]:1983 at 16:40:09.569920: ------------------------------------------------------------------------ BYE sip:4099@freeswitch_ip_address:5060;transport=udp SIP/2.0 Via: SIP/2.0/WS ct12n6htslrn.invalid;branch=z9hG4bK6881335 Max-Forwards: 69 To: <sip:4099@freeswitch_ip_address >;tag=2aKcSpHygmD8e From: <sip:my_agent_48@freeswitch_ip_address >;tag=usavf945ak Call-ID: ct72vo1al4goc67fb7bt CSeq: 3388 BYE Supported: path, outbound, gruu User-Agent: JsSIP 0.3.0 Content-Length: 0 ------------------------------------------------------------------------ 5ee0dd16-a867-11e3-a3a0-197b4536e6b9 2014-03-10 16:40:19.396567 [DEBUG] switch_core_session.c:1050 Send signal sofia/internal/my_agent_48@freeswitch_ip_address [BREAK] The last commit is: commit 42b74ba62eab3cfe7b6e38c82c0eac30d2b6ceaf Author: Brian West <brian(a)freeswitch.org> Date: Sat Mar 8 08:51:58 2014 -0600 remove generated file from git Thank you in advance for your help. Denis Gasparin

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Confirm ClueCon 2014 date
by Seven Du 10 Mar '14

10 Mar '14

Hi, I’d like to confirm the Cluecon date so can buy a flight ticket ahead, freeswitch.org shows count down 146 days 13 hours, is that means Aug 3rd - 5th? Thanks. dujinfang=# select now() + '146 days 13 hours'; ?column? ------------------------------- 2014-08-03 23:11:58.720293+08 (1 row) -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow (http://www.sparrowmailapp.com/?sig)

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Plans for new features?
by D D 10 Mar '14

10 Mar '14

Hi, Can you share your plans for new features? Are there any new video features planned to be available soon? Thanks, David

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PlayAndGetDigits query
by Srini K 10 Mar '14

10 Mar '14

Hi, Iam using Mod_Managed to PlayAndGetDigits. It plays EnterZip.wav file and waits for 5 digit Zip code to be entered. No issues on succesful entry. On invalid entry (within the retry limits) it plays InvalidZip.wav followed by EnterZip.wav. Is it the expected behaviour ? On Invalid entry I want only InvalidZip.wav to be played NOT EnterZip.wav is there a way? My API call looks like digits = session.PlayAndGetDigits(5, 5, 3, 5000, "#*", "EnterZip.wav", "InvalidZip.wav", "\\d+", "", 3000, null) I referred the following wiki too. http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits https://wiki.freeswitch.org/wiki/Mod_lua#session:playAndGetDigits Thanks Srini

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MWI
by Joel White 10 Mar '14

10 Mar '14

I am working to integrate Kamailio and FreeSWITCH I have the two systems talking, yet the SUBSCRIBE to FreeSWITCH is not yielding any results as in the message indicator on my Polycom phone. Does anyone know how to make this work and also the correct subscribe string to put on the endpoint?

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Attended transfer using the event socket library protocol
by John Platts 10 Mar '14

10 Mar '14

I need some help on how to do an attended transfer using the event socket library protocol. Original call: Channel A <-------------> Channel B(transferring endpoint) (other endpoint of original call) Channel C <-------------> Channel D(transferring endpoint) (endpoint to transfer to) How do I go about disconnecting Channels A and Channel C, and bridging Channel B with Channel D? In the scenario that I have described, I do know the UUIDs for all four of the channels involved. I also need some help on how to conference in a third party into an existing phone call using the event socket library. In this scenario, the call is already connected and I know the UUID of both of the channels of the call. How do I go about conferencing in a third party using the event socket library protocol?

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Build issues in master.
by Michael Jerris 10 Mar '14

10 Mar '14

I think I have all the build breakage from friday in master all worked out now. if anyone continues to have any build issues please file a Jira. Thanks mike

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Bridge command getting truncated
by Manish Kumar 10 Mar '14

10 Mar '14

Hi, I am facing an issue where my argument of bridge command is getting truncated while using through perl esl. I am having a requirement where i need to bridge the call to 8 agents in a sequential manner and i am using pipe separator to fulfil this. This works well when i am using 2-3 bridging numbers, however the argument part is truncated when trying to use more than 4-5. The argument is truncated at exact length of 503 characters. Please find the below sample from the freeswitch DEBUG logs. bridge({originate_timeout=40,vcall_id=20140307212134279125021410419292,user_id=11,script_name=ibd_bridge_test,interface_id=81,ignore_early_media=true,caller_id_number=xxxxxxxxxx,api_to_call= http://xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx… }[leg_timeout=40,ignore_early_media=true,destination=xxxxxxxxxx]freetdm/wp4/a/xxxxxxxxxxx|[leg_timeout=40,ignore_early_media=true,destination=xxxxxxxxxx]freetdm/wp4/a/0xxxxxxxxxx|[leg_timeout=40,ignore_early_media=true,destination=99) I am using freeswitch version 1.2.14 with perl ESL, and the bridge command is sent to the freeswitch using below line , $conn->execute("bridge","----args as mentioned above----");

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YeaLink :(
by Blake Priddy 09 Mar '14

09 Mar '14

Anyone had an issue when calling in to a yealink ext it just goes to vm and says NO_USER_RESPONSE in the logs. -- *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656

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Dialplan record on demand - email recording on exit?
by Sean Devoy 09 Mar '14

09 Mar '14

Hi All, I am learning to setup the record on demand (via dtmf) from the wiki. Seems pretty straight forward, except my user needs the sound file when done. How can I email the sound file when the session ends? Of course, there may not be a recording as it is "on demand". Thanks, Sean

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