February 2015 - FreeSWITCH-dev - FreeSWITCH Mailing Lists

Silence in conference recording
by miguelo sana 06 May '15

06 May '15

Hi, I use the recording functionality of mod_conference to record conference calls. The recording is save to a .wav file. Now for the second time in a few months I experienced something strange. The recording goes silent after 15 minutes of recording. But the recording is not stopped. There is nothing strange found in logging. And other recordings are not having this issue. Is there anyone who knows anything about this and how to fix or prevent this. Thanks. Miguelo

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Bare Minimum Freeswitch
by Jacob Hodges 09 Mar '15

09 Mar '15

Hi all, I'm quite new to freeswitch and have used it in the past to establish a SIP Trunk between Lync and another SIP provider. I'm now looking at Freeswitch as a potential replacement for a legacy PBX at my local church. I'm extremely wary of the default Freeswitch configuration as it seem to include a lot of functionality that I don't need and a lot configuration I don't understand. All I need is the following: - 5 extensions that can dial each other and the PSTN - Ability to do group pickup, paging and voicemail to email - Ability to set day/night mode - Ability to use SPA3102 as a gateway to the PSTN. Can anyone tell me the minimum number of modules / configuration that is needed to accomplish this? Thanks, Jacob.

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mod_rayo patching
by ik 08 Mar '15

08 Mar '15

Hello, There is a bug in mod_rayo that I wish to patch myself, however I have never touched any sources of FS before, not to mention rayo itself (and it's been many years since I touched C). Before I'll start to figure out the proper source code to touch, is there a good guide to read about FS development that can help in the matter ? Thanks Ido

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 27 Feb '15

27 Feb '15

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 20 Feb '15

20 Feb '15

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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valet parking new feature
by e schmidbauer 19 Feb '15

19 Feb '15

Hi, Would anyone want to test out new parking feature for polycoms? The feature is outlined in this doc: http://support.polycom.com/global/documents/support/technical/products/voic… Just pull clean FS branch and do a git apply with that diff like git apply --ignore-whitespace ~/valet_sofia.c.diff make mod_sofia-install make mod_valet_parking-install restart FS Provisiong the polycom with: attendant.resourceList.2.address="orbit+lot(a)mydomain.com" attendant.resourceList.2.label="Orbit Lot" attendant.resourceList.2.type="automata" FreeSWITCH dialplan Park in Orbit: <extension name="parking_auto_in"> <condition field="destination_number" expression="^\*7$"> <action application="answer"/> <action application="valet_park" data="lot(a)mydomain.com auto in 701 799"/> </condition> </extension> Pick up from Orbit: <extension name="parking_pickup"> <condition field="destination_number" expression="^\*97(7[0-9][1-9])$"> <action application="answer"/> <action application="valet_park" data="lot(a)mydomain.com $1"/> </condition> </extension> Please test and let me know what you think... I'd like to get this feature pushed to valet_parking instead of forking the module Thanks

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 13 Feb '15

13 Feb '15

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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Can't negotiate L16@16000h with Asterisk
by Antonio Eugenio Burriel 09 Feb '15

09 Feb '15

Yesterday I posted the subject "LIN16 at 16000h Freeswitch <-> Asterisk" in the user mailing list, but after having taken a peek at the source, I thought it was better to share my analysis here: Problem: Freeswitch can't negotiate L16@16000h codec with some peers (like Aterisk) which assigns a dynamic PT (over 95) to L16. Internally Freeswitch assigns a PT of 70 to L16; in switch_core_media_negotiate_sdp() we have the following match check: match = (!strcasecmp(rm_encoding, imp->iananame) && ((map->rm_pt < 96 && imp->ianacode < 96) || (map->rm_pt > 95 && imp->ianacode > 95)) && (remote_codec_rate == codec_rate || fmtp_remote_codec_rate == imp->actual_samples_per_second)) ? 1 : 0; Which means that when our peer's codec is dynamic, our match must be also dynamic and, if not, out match mustn't. Except for L16@44100h, which has a PT of 10, its PT should be dynamic (>95) according to RFC3551. But Freeswitch is internally assigning 70, so it will never match. Incidentally, Asterisk erroneously assigns to L16@8000h a PT of 10, which passes the check. I'm not quite sure which is the proper way to fix this. If we change PT from 70 to something over 95, we'll break interoperability with older Freeswitches and other peers, such LN16@800h with Asterisk. So maybe its cleaner to add a exception to the match check like: match = (!strcasecmp(rm_encoding, imp->iananame) && ((map->rm_pt < 96 && imp->ianacode < 96) || (map->rm_pt > 95 && (imp->ianacode > 95 || imp->ianacode == 70))) && (remote_codec_rate == codec_rate || fmtp_remote_codec_rate == imp->actual_samples_per_second)) ? 1 : 0; Any thoughts?

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Using FS with TLS and not TLS
by Yücel ALTUNAY 09 Feb '15

09 Feb '15

Hi, I want to use freeswitch with TLS on my mobile phones and without TLS on my GSM gateways. I see some examples but i coudn't do it. Is someone has example to do this. Thank you.

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GDB
by Ali R. 07 Feb '15

07 Feb '15

Dear All,I have FS configured/compiled with "devel-bootstrap.sh" for better debugging. Is it possible to extract the parsed SIP message from a core dump (or as much SIP as possible.) traversing the back trace, I was able to see host, port, destination number which seems to be fine but I have a feeling there might be some structures with deep SIP info. I'm having random core dumps and I would like to get ideas at the call patterns that causes this crash. I will for sure report all details on this particular crash to JIRA for better tracking. In the mean time, any guidance is appreciated. #0 0x00007f6d8192f165 in raise () from /lib/x86_64-linux-gnu/libc.so.6#1 0x00007f6d819323e0 in abort () from /lib/x86_64-linux-gnu/libc.so.6#2 0x00007f6d8196a1cb in ?? () from /lib/x86_64-linux-gnu/libc.so.6#3 0x00007f6d81973a16 in ?? () from /lib/x86_64-linux-gnu/libc.so.6#4 0x00007f6d819772f8 in ?? () from /lib/x86_64-linux-gnu/libc.so.6#5 0x00007f6d819788a0 in malloc () from /lib/x86_64-linux-gnu/libc.so.6#6 0x00007f6d82bf78c2 in my_dup (s=0x7f6d54313360 "user=phone") at src/switch_event.c:113#7 0x00007f6d82bf8e20 in switch_event_add_header_string (event=0x7f6d555b3ae0, stack=11636, stack@entry=SWITCH_STACK_BOTTOM, header_name=0x6 <Address 0x6 out of bounds>, header_name@entry=0x7f6d77d9afbc "sip_contact_params", data=0xffffffffffffffff <Address 0xffffffffffffffff out of bounds>, data@entry=0x7f6d54313360 "user=phone") at src/switch_event.c:1182#8 0x00007f6d82b8ca02 in switch_channel_set_variable_var_check (channel=channel@entry=0x7f6d5492e380, varname=varname@entry=0x7f6d77d9afbc "sip_contact_params", value=0x7f6d54313360 "user=phone", var_check=var_check@entry=SWITCH_TRUE) at src/switch_channel.c:1385#9 0x00007f6d77c7462d in _url_set_chanvars (session=session@entry=0x7f6d55395d88, url=0x7f6d5432c488, user_var=user_var@entry=0x7f6d77d9afab "sip_contact_user", host_var=host_var@entry=0x7f6d77d9af9a "sip_contact_host", port_var=port_var@entry=0x7f6d77d9af89 "sip_contact_port", uri_var=uri_var@entry=0x7f6d77d9af79 "sip_contact_uri", params_var=params_var@entry=0x7f6d77d9afbc "sip_contact_params") at sofia.c:119#10 0x00007f6d77c8ec40 in sofia_handle_sip_i_invite (session=session@entry=0x7f6d55395d88, nua=nua@entry=0x7f6d782aae80, profile=profile@entry=0xb60550, nh=nh@entry=0x7f6d5498d830, sofia_private=0x7f6d552b0c70, sip=sip@entry=0x7f6d540492c8, de=de@entry=0x7f6d54207b30, tags=tags@entry=0x7f6d554de4b0) at sofia.c:9434#11 0x00007f6d77c96bc1 in our_sofia_event_callback (event=nua_i_invite, status=100, phrase=0x7f6d554de4c0 "Trying", nua=0x7f6d782aae80, profile=0xb60550, nh=0x7f6d5498d830, sofia_private=0x7f6d552b0c70, sip=0x7f6d540492c8, de=de@entry=0x7f6d54207b30, tags=0x7f6d554de4b0) at sofia.c:1491#12 0x00007f6d77c9a96c in sofia_process_dispatch_event (dep=dep@entry=0x7f6d427e2108) at sofia.c:1903#13 0x00007f6d77c6ace9 in sofia_receive_message (session=0x7f6d55395d88, msg=0x7f6d427e2a90) at mod_sofia.c:1115#14 0x00007f6d82bb14ae in switch_core_session_perform_receive_message (session=session@entry=0x7f6d55395d88, message=message@entry=0x7f6d427e2a90, file=file@entry=0x7f6d82caf290 "src/switch_ivr.c", func=func@entry=0x7f6d82cafb60 "switch_ivr_parse_all_signal_data", line=line@entry=838) at src/switch_core_session.c:775#15 0x00007f6d82c35705 in switch_ivr_parse_all_signal_data (session=session@entry=0x7f6d55395d88) at src/switch_ivr.c:838#16 0x00007f6d82c35750 in switch_ivr_parse_all_messages (session=session@entry=0x7f6d55395d88) at src/switch_ivr.c:797#17 0x00007f6d82bb5fdf in switch_core_session_run (session=0x7f6d55395d88) at src/switch_core_state_machine.c:475#18 0x00007f6d82bb0f1e in switch_core_session_thread (thread=<optimized out>, obj=0x7f6d55395d88) at src/switch_core_session.c:1598#19 0x00007f6d82bad1b5 in switch_core_session_thread_pool_worker (thread=0x7f6d710c7d40, obj=<optimized out>) at src/switch_core_session.c:1690#20 0x00007f6d822e7b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0#21 0x00007f6d819d97bd in clone () from /lib/x86_64-linux-gnu/libc.so.6#22 0x0000000000000000 in ?? () Ali R.

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