- FreeSWITCH-users - FreeSWITCH Mailing Lists

ESL, "181 Call Is Being Forwarded" and voice mail detection
by Marcin S 28 Aug '15

28 Aug '15

Hi, I have a dialer app that uses ESL and freeswitch. I would like to detect when voice mail answers call. For such calls there is a "181 Cal Is Being Forwarded" message in SIP signalization. However, in ESL there is no event fired (maybe beacuse there was 183 and 180 earlier). How can I detect, that there was 181 received? Here is an example sip trace: 1 0.000000 192.168.88.237 -> 89.108.20.42 SIP/SDP Request: INVITE sip:123456789@89.108.20.42, with session description 2 0.018028 89.108.20.42 -> 192.168.88.237 SIP Status: 100 Trying 3 0.418550 89.108.20.42 -> 192.168.88.237 SIP/SDP Status: 183 Session Progress, with session description 4 1.405465 89.108.20.42 -> 192.168.88.237 SIP Status: 180 Ringing 5 29.270672 89.108.20.42 -> 192.168.88.237 SIP Status: 181 Call Is Being Forwarded 6 29.659371 89.108.20.42 -> 192.168.88.237 SIP/SDP Status: 200 OK, with session description 7 29.670833 192.168.88.237 -> 89.108.20.42 SIP Request: ACK sip:3c6d5343fb70df3a18f4dd32e7c9d226@89.108.20.42:5060;transport=udp 8 70.590778 192.168.88.237 -> 89.108.20.42 SIP Request: BYE sip:3c6d5343fb70df3a18f4dd32e7c9d226@89.108.20.42:5060;transport=udp 9 70.625224 89.108.20.42 -> 192.168.88.237 SIP Status: 200 OK

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Crash in switch_ivr_play_and_detect_speech (mod_managed + mod_pocketshinx)
by Prashant Choudhary 28 Aug '15

28 Aug '15

Hello I am getting crash once my application reaches this line. freeswitch.switch_ivr_play_and_detect_speech(IvrSession.InternalSession, @"hello.wav", "pocketsphinx", "pizza_toppings", ref result, 3000, null); I am using mod_managed in AppContext mode. It says no speech Detected and crashes immediately everytime. I am using XLite to dial into freeswitch. The sound file starts playing, but if I speak some words from the dictionary, the application would crash. I am attaching the log. Thanks in advance. :)

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outbound-proxy
by Bruce Lefko 27 Aug '15

27 Aug '15

Can this be a domain name? Alternatively, can I use mod_distributor and mod xml curl to dynamically specify the outbound proxy? Essentially, my problem is that we have multiple kamailio proxies for HA and want to be able to rely on both as well as occasionally take one out of rotation for maintenance. How can we acheive this? Thanks!

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preserve A-leg From header value and P-Asserted-Identity in B-leg
by Vik Killa 27 Aug '15

27 Aug '15

Hello, According to Q.1912 spec, the 'Generic Number' or sometimes called 'Additional calling number' may be placed in the "From" SIP header, while the Calling Party Number can be placed in the P-Asserted-Identity. We've noticed that in the case when FreeSWITCH receives an INVITE like this (From header & PAI header have different information), FreeSWITCH will disregard the From and use only the PAI number in both B-leg From header and PAI. I've found a work around to solve this by exporting the A-leg variable ${sip_from_user} into the B-leg's ${sip_from_user} Example: <action application="bridge" data="{sip_from_uri=${ sip_from_user}@172.20.207.155}sofia/internal/sip:555@172.20.207.154:5060"/> This seems like a hack at best... my question is, there a more elegant way to preserve the respective From & PAI of A-leg and just place them in B-leg? Thanks in advance.

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freeswitch 1.4 on debian 8.1, good idea ?
by Yuriy Nasida 27 Aug '15

27 Aug '15

Hello list, I know that docs recommend debian 8.1 for FS 1.6.* but... do anybody have positive experience with FS 1.4.* on debian 8.1 ? I would like to use debian 8 because of new unixODBC packet - 2.3.1 which here by default. I tried to install unixODBC 2.3.1 from source at Debian 7 because of a lot of FS segfailts (they related with ODBC) but it didn't solve my issue. I also would not like to use FS 1.6 at this moment. Please advice, Thanks

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Re: [Freeswitch-users] FreeSWITCH HA Project Discussion Date/Time
by Alex Goulis 27 Aug '15

27 Aug '15

Let's do this!!! Thank you, Alex GoulisSales EngineerOpenSIPS Solutions(401) 641-1072agoulis(a)opensips.org -------- Original message -------- From: Brian West <brian(a)freeswitch.org> Date: 08/27/2015 8:18 AM (GMT-06:00) To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: [Freeswitch-users] FreeSWITCH HA Project Discussion Date/Time How does Friday August 28th at 10AM Central sound? -- Brian West brian(a)freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.comGot Bugs? Report them here! | Reddit: /r/freeswitch T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest

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FreeSWITCH HA Project Discussion Date/Time
by Brian West 27 Aug '15

27 Aug '15

How does Friday August 28th at 10AM Central sound? -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest

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Set Variable in Dial String from Lua Script
by Ximo Peral 27 Aug '15

27 Aug '15

I am using a lua script in Freeswitch.The routes are generated from SQL queries. And there may be several trunks output.For that I use bridge failover: bridge({ignore_early_media=false}[nobal_amt=-250,execute_on_post_originate=limit hash outbound NameTrunk 500 !NORMAL_TEMPORARY_FAILURE,gateway=4]sofia/gateway/NameTrunk/xxxxxxxxxxxxx|[nobal_amt=-250,execute_on_post_originate=limit hash outbound NameTrunk2 500 !NORMAL_TEMPORARY_FAILURE,gateway=5]sofia/gateway/NameTrunk2/xxxxxxxxxxxxx) The problem are the channels variable in dial strings, these are not stored in the XML CDR or CDR CSV and do not know the variable "gateway" (gateway=4 or gateway=5). It is very important that I can save channels variables that are in dial strings. Any ideas? Thanks!

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INVITE and hangup in the middle of a call
by Oleg Stolyar 27 Aug '15

27 Aug '15

Hi guys, Several weeks ago I started getting an occasional problem where FS is sending an INVITE to the other side in the middle of a call, the other side does not respond and FS hangs up the leg. Below is the relevant log. The user experience is that they keep talking and hearing each other up to the very end. I have a recording of that call, so can confirm. The call uses WebRTC and is originated by JsSip from Chrome. Then the user is put into a conference but I doubt it's relevant in this case since the INVITE and disconnect are not happening from mod_conference I suspect it's a re-INVITE but what triggers FS to send it? I couldn't find anything in the logs that could shed light. send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933: ------------------------------------------------------------------------ INVITE sip:dj012n41@u40rf5qikah5.invalid;transport=ws;ob SIP/2.0 Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr Max-Forwards: 70 From: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@anonymous.invalid >;tag=KQecUSr12rSQp To: "user1" <sip:anonymous@anonymous.invalid>;tag=v1rlqab64i Call-ID: g8980rbrbk2t45oj5mru CSeq: 75703945 INVITE Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@ ##.##.###.###:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Session-Expires: 120;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 825 v=0 o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.## s=FreeSWITCH c=IN IP4 ##.##.##.## t=0 0 a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW m=audio 22670 RTP/SAVPF 9 126 106 a=rtpmap:9 G722/8000 a=rtpmap:126 telephone-event/8000 a=rtpmap:106 CN/8000 a=ptime:20 a=fingerprint:sha-256 E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69 a=rtcp-mux a=rtcp:22670 IN IP4 ##.##.##.## a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0 a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0 a=ice-ufrag:5dS3Fzx1Thrmdy9Z a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host generation 0 ------------------------------------------------------------------------ fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous(a)anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous(a)anonymous.invalid entering state [calling][0] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous(a)anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous(a)anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous(a)anonymous.invalid entering state [terminating][503] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous(a)anonymous.invalid [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_channel.c:3242 Send signal sofia/leia_agent/anonymous(a)anonymous.invalid [KILL] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1396 Send signal sofia/leia_agent/anonymous(a)anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:5057 Channel leaving conference, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous(a)anonymous.invalid skip receive message [UNBRIDGE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_media.c:7772 sofia/leia_agent/anonymous(a)anonymous.invalid skip receive message [REFER_EVENT] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous(a)anonymous.invalid Restore previous codec G722:9. fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_session.c:2901 sofia/leia_agent/anonymous(a)anonymous.invalid skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:535 (sofia/leia_agent/anonymous(a)anonymous.invalid) State EXECUTE going to sleep fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:472 (sofia/leia_agent/anonymous(a)anonymous.invalid) Running State Change CS_HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:735 (sofia/leia_agent/anonymous(a)anonymous.invalid) Callstate Change ACTIVE -> HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous(a)anonymous.invalid) State HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous(a)anonymous.invalid hanging up, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:60 sofia/leia_agent/anonymous(a)anonymous.invalid Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous(a)anonymous.invalid) State HANGUP going to sleep

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Re: [Freeswitch-users] bridge unexpectedly disconnects after 5-6 sec
by Sergey Zhuravlov 27 Aug '15

27 Aug '15

Hi I looked at the number that you dial 7* I take my crystal ball and assume that you are using Multifon as gatewey ;-) I was with him a similar problem. The problem is solved by setting the parameter <param name="rtcp-audio-interval-msec" value="5000"/> for profile external here it is in Multifon, it just works ---------- Пересылаемое сообщение ---------- > From: Artur Mega <findmeinwland(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Wed, 26 Aug 2015 02:28:46 +0500 > Subject: [Freeswitch-users] bridge unexpectedly disconnects after 5-6 sec > i set up freeswitch, and almost all configs are default (i made some > little changes in directory and dialplan). I can call anywhere, for example > to my cell phone, but call longs approximetely only 5 sec... In siptrace i > found nothing. No timeouts like call_timeout or leg_timeout. > > Also i thought maybe the reason in the gateway registerations... > Server has private (static) ip. > > log is here: > http://pastebin.com/mLrkGz3c > > -- > > Arthur > > > ---------- Пересылаемое сообщение ---------- > From: Anthony Minessale <anthony.minessale(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 25 Aug 2015 16:48:30 -0500 > Subject: Re: [Freeswitch-users] bridge unexpectedly disconnects after 5-6 > sec > That trace is too vague. > > Do these cli commands first then it will probably be more obvious: > > sofia global siptrace on > console loglevel debug > > > On Tue, Aug 25, 2015 at 4:28 PM, Artur Mega <findmeinwland(a)gmail.com> > wrote: > >> i set up freeswitch, and almost all configs are default (i made some >> little changes in directory and dialplan). I can call anywhere, for example >> to my cell phone, but call longs approximetely only 5 sec... In siptrace i >> found nothing. No timeouts like call_timeout or leg_timeout. >> >> Also i thought maybe the reason in the gateway registerations... >> Server has private (static) ip. >> >> log is here: >> http://pastebin.com/mLrkGz3c >> >> -- >> >> Arthur >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ > > ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ > http://twitter.com/FreeSWITCH > ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ > <http://freeswitch.org/g+>* > > ClueCon Weekly Development Call > ☎ sip:888@conference.freeswitch.org ☎ +19193869900 > > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- WBR, Sergey GTALK/JABBER:4orbit@gmail.com

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