January 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Hiredis + limit_execute bridge ends up in hangup(-ERR)
by Alexandru Covalschi 31 Jan '18

31 Jan '18

Hello list, Recently I've changed limit_execute engine from hash to hiredis and it leaded to some weird floating bug. Sometimes I get hangup(-ERR) with no further explanation, it doesn't occur each time, sometimes similar bridge can be fine, sometimes it results in immediate hangup(-ERR) with no channel B created (invite is not even coming out). When I switch the limit_execute engine back to hash it works fine. That occurs both on Centos 7 and Debian 8 machines, on Centos I have latest code from git branch v.1.6, on Debian - latest packages from FS repo. Here's the log of the problematic place https://pastebin.com/AJG8S5Yx <https://pastebin.com/AJG8S5Yx> If you need the full log please let me know. Redis benchmark showed that redis is working fine, I have only two machines per one redis-server, so it's not a socket or memory limit (but anyway I didn't check that yet, will do later today) Thanks in advance ________________________________ Alexandru Covalschi VoIP Engineer and System Administrator tel: +373 673 98 493

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How to modify P-asserted-id on UPDATE?
by Chris B. Ware 31 Jan '18

31 Jan '18

Hi, I use to set send-display-update in order to show external caller number on phones, even after an attended transfer or a call pickup. This number is saved on phone's call history and can not be used to call it again because we use zero as prefix to call outside. That's why I'd like to add a zero to caller number used un UPDATE P-asserted-id value. How can I do it? For example: - A (mobile fone 312345) calls B (extension 201) - C (extension 203) picks up that call - C talks with A and on C display I see 312345 due to sip UPDATE P-asserted-id I'd like to send 0312345 on that P-asserted. Is it possible? Thank you. chrisbware

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Direct transfer Callcenter Agent
by Andres Gomez 31 Jan '18

31 Jan '18

Hi Friends. With mod_callcenter can I transfer direct to agent queue? Regards Andres Gomez.

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start_dtmf_generate is not working with opus code
by Andrew Paul 29 Jan '18

29 Jan '18

Hi all, I am facing a issue with start_dtmf_generate when i use opus codec and SIP INFO dtmf. Caller ( opus/dtmf ) ------------------------ Freeswitch ----------------------------- Calleee ( ulaw/inband) Caller is using jssip webrtc and dtmf is in INFO. For inband dtmf generation i am using start_dtmf_generate in caller channel. The setup is working when i use ulaw or ilbc or alaw codec. When i use webrtc with OPUS codec i am seeing the inband dtmf generation is not working. Webrtc with ulaw codec and dtmf INFO is working fine. Version : FreeSWITCH Version 1.6.17~64bit ( 64bit) I am attaching the opus.conf.xml for the reference. Any idea, why start_dtmf_generate is not working with opus codec ?. Do i need to set any setting for opus ?. Regards

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DeutschlandLAN SIP Trunk
by Paul Muaddib 29 Jan '18

29 Jan '18

Hi, does anyone here have experience with the SIP trunk service from the German Telekom: "DeutschlandLAN SIP-Trunk"? There is little information from Deutsche Telekom on the Internet besides the "Technical Technical Specification of the SIP - 1TR114" As I have read so far, Deutsche Telekom has not necessarily been sticking to the standard. What I know so far Transport protocol is probably TCP instead of UDP Codecs: G. 711a and G. 722 Fax either via G. 711a or T. 38 My Setup Internet <-----> Router/Firewall(NAT) <----> Freeswitch According to the instructions, they require you to open several ports in the firewall. Is that really necessary? Does the Keep-Alive function work without opening any port? At the Federal Office for Information Security (BSI) it is additionally recommended to use a SIP proxy such as kamailio At the moment we are still using an ISDN gateway. I don't like to run into problems when we switch over, because I can't go back then. Best regards, Paul

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Code 88
by Rick Jarvis 29 Jan '18

29 Jan '18

I had a box which was running FS 1.4, so I upgraded it to 1.6. Now I’m getting incompatible destination when dialing out through my provider, despite having changed nothing else. I’ve tried changing the outbound codec prefs to just PCMU,PCMA in case that was it, but still no joy. Here’s a SIP trace, can we tell anything from this, or are there any other likely explanations given the upgrade? INVITE sip:0<desintationnumber>@sip.mysipprovider SIP/2.0 Via: SIP/2.0/UDP 1.2.3.4:5080;rport;branch=z9hG4bKe57KcBtmmr9cB Max-Forwards: 69 From: "0<sourcenumber>" <sip:<accountname>@sip.mysipprovider>;tag=DvXrSpNBaFc1e To: <sip:0<desintationnumber>@sip.mysipprovider> Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITE Contact: <sip:gw+gatewayname@1.2.3.4:5080;transport=udp;gw=gatewayname> User-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Proxy-Authorization: Digest username="<accountname>", realm="sip.mysipprovider", nonce="81346aeb-453b-466e-ba50-5d027c1a55e3", cnonce="pVaZvn+kEjaVvQQBGMFfAQ", algorithm=MD5, uri="sip:0<desintationnumber>@sip.mysipprovider", response="4702ba8b72c43939d5fa88528eafaf2b", qop=auth, nc=00000001 Content-Type: application/sdp Content-Disposition: session Content-Length: 246 X-FS-Support: update_display,send_info Remote-Party-ID: "0<sourcenumber>" <sip:0<sourcenumber>@sip.mysipprovider>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1517205769 1517205770 IN IP4 1.2.3.4 s=FreeSWITCH c=IN IP4 1.2.3.4 t=0 0 m=audio 30824 RTP/AVP 0 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 ------------------------------------------------------------------------ 2018-01-29 14:36:33.926697 [DEBUG] sofia.c:7084 Channel sofia/external/0<desintationnumber> entering state [calling][0] recv 338 bytes from udp/[<providerip>]:5060 at 14:36:34.354355: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bKe57KcBtmmr9cB From: "0<sourcenumber>" <sip:<accountname>@sip.mysipprovider>;tag=DvXrSpNBaFc1e To: <sip:0<desintationnumber>@sip.mysipprovider> Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITE User-Agent: mysipprovider Content-Length: 0 ------------------------------------------------------------------------ 2018-01-29 14:36:33.966698 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47730 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.066706 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47731 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.186741 [DEBUG] switch_core_io.c:448 Setting BUG Codec PCMU:0 2018-01-29 14:36:34.266726 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47732 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.366714 [INFO] switch_rtp.c:7268 Auto Changing audio port from 10.10.131.204:11782 to <myphonesip>:11782 2018-01-29 14:36:34.466703 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47733 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.666721 [DEBUG] switch_rtp.c:1463 [ zrtp engine]: WARNING! HELLO have been resent 5 times without a response. Raising ZRTP_EVENT_NO_ZRTP_QUICK event. ID=5 2018-01-29 14:36:34.666721 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47734 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:34.866711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47735 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.066711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47736 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.266717 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47737 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.466732 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47738 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.666724 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47739 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:35.866711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47740 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.066719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47741 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.266719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47742 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.466719 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47743 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.666711 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47744 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:36.866721 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47745 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:37.066709 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47746 size=148. Stream 5:CLEAR:START 2018-01-29 14:36:37.266699 [DEBUG] switch_rtp.c:1463 [ zrtp utils]: Send <HELLO> ssrc=3933957113 seq=47747 size=148. Stream 5:CLEAR:START recv 705 bytes from udp/[<providerip>]:5060 at 14:36:37.863874: ------------------------------------------------------------------------ SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bKe57KcBtmmr9cB Max-Forwards: 69 From: "0<sourcenumber>" <sip:<accountname>@sip.mysipprovider>;tag=DvXrSpNBaFc1e To: <sip:0<desintationnumber>@sip.mysipprovider>;tag=v2D22KNm5y4KS Call-ID: a553eda8-7fa4-1236-bd95-040118c15f01 CSeq: 118282234 INVITE User-Agent: mysipprovider Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 P-Charging-Vector: icid-value=0954708e-d24b-489c-a16a-0244d89313f8 ------------------------------------------------------------------------

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Re: [Freeswitch-users] Not trying second route on NORMAL_TEMPORARY_FAILURE
by Avi Marcus 29 Jan '18

29 Jan '18

Hi -- I'm still having this issue, any ideas? I'm only seeing it with one route and one carrier. Any useful debug info for this? -Avi Marcus 1-718-989-9485 (USA) 02-372-1570 (Israel) 020-3298-2875 (UK) On Thu, Jan 25, 2018 at 9:46 PM, Avi Marcus <avi(a)avimarcus.net> wrote: > I have a bridge string with failover of "route A| route B" but I found a > place where it's not working. > > I have continue_on_fail=true in the A leg. > > originate_disposition is telling me SUCCESS > > even though bridge_hangup_cause is recording NORMAL_TEMPORARY_FAILURE > and sip_hangup_disposition is recv_refuse > > So.. what's wrong? How can I make sure the bridge failover works? As far > as I know, normally it properly fails over -- but I just saw this several > times today. > > Thanks, > -Avi Marcus >

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spandsp_start_tone_detect problem
by Raimundo Pérez Nieves 29 Jan '18

29 Jan '18

Hi guys, I execute my outbound event socket and FS debug shows me the Tone Detected, but there is no event captured. I already test $con->event(“plain”,”all”); and nothing….. $con->sendRecv("myevents $uuidLegB”); $con->execute("spandsp_start_tone_detect","34”); while($con->connected()) { my $e = $con->recvEvent(); if ($e) { my $name = $e->getHeader("event-name"); print "EVENT [$name]\n"; } } Thanks

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start_dtmf_generate not working with opus codec
by Andrew Paul 29 Jan '18

29 Jan '18

Hi all, I am facing a issue with start_dtmf_generate when i use opus codec and SIP INFO dtmf. Caller ( opus/dtmf ) ------------------------ Freeswitch ----------------------------- Calleee ( ulaw/inband) Caller is using jssip webrtc and dtmf is in INFO. For inband dtmf generation i am using start_dtmf_generate in caller channel. The setup is working when i use ulaw or ilbc or alaw codec. When i use webrtc with OPUS codec i am seeing the inband dtmf generation is not working. Webrtc with ulaw codec and dtmf INFO is working fine. Version : FreeSWITCH Version 1.6.17~64bit ( 64bit) I am attaching the opus.conf.xml for the reference. Any idea, why start_dtmf_generate is not working with opus codec ?. Do i need to set any setting for opus ?. Regards Paul

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Re: [Freeswitch-users] spandsp_start_tone_detect is compatible with G729 codec???
by Raimundo Pérez Nieves 29 Jan '18

29 Jan '18

So if I change to G711 spandsp module will detect each tone??? Enviado desde mi iPhone El 25/01/2018, a la(s) 00:29, Alexey Sibyakin <alex(a)freeswitch.com> escribió: > As Brian said most of tones cannot be carried via g729 only some of them. You are just didn't encounter this problem yet. I believe that it's a good idea to convert inband -> rfc2833 right on the border with analog lines (which are g711). > >> On Thu, Jan 25, 2018 at 2:13 AM, Raimundo Pérez Nieves <raimundo.perez.cuba(a)gmail.com> wrote: >> Thanks Brian, but I am using tone_detect with g729 (commercial license) and works very good. That was the reason I tried with spandsp module, because it allows detect tone using cadence. >> What is the meaning of f1 and f2 in segment tone debug, nobody can answer me this question. >> Thanks >> >> Enviado desde mi iPhone >> >> El 24/01/2018, a la(s) 16:40, Brian West <brian(a)freeswitch.com> escribió: >> >>> You can't do tone detect on G729, the bandwidth of the audio is insufficient! >>> >>>> On Wed, Jan 24, 2018 at 5:58 AM, Raimundo Pérez Nieves <raimundo.perez.cuba(a)gmail.com> wrote: >>>> Hello, >>>> spandsp_start_tone_detect is compatible with G729 codec??? >>>> Because Iam testing and I get no result. >>>> >>>> Thanks >>>> >>>> _________________________________________________________________________ >>>> 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 >>> >>> >>> >>> -- >>> >>> Brian West | Co-founder and Developer >>> Need Commercial support? email sales(a)freeswitch.com >>> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 >>> Email: brian(a)freeswitch.com >>> Mobile: 918-424-9378 >>> Website: https://www.FreeSWITCH.com >>> >> >> _________________________________________________________________________ >> 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 > > > > -- > > Alex Sibyakin | Support Engineer > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > Email: alex(a)freeswitch.com > Website: https://www.FreeSWITCH.com > Need commercial support? Contact sales(a)freeswitch.com for details.

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