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g711 fallback on calls where t38 negotiation fails
by Bruce Lefko 12 Jan '15

12 Jan '15

Is there a way in mod_spandsp to allow for g711 fallback in calls where t38 negotiation fails? I would like to not have to call a number back with fax_enable_t38=false and have the call continue with audio fallback. Thanks!

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Record sound quality not good but pcap is quite good
by Ali Pey 12 Jan '15

12 Jan '15

Hello, I use record to record part of a call and sometimes I get some static noises and distortion; however when I listen to the call on pcap capture, it's quite clear and no noise. I record to a wav file and then use sox to convert it to ulaw. Any suggestions? Any help would be greatly appreciated. Thanks, Ali Pey

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files.freeswitch.org
by Brian West 12 Jan '15

12 Jan '15

We migrated this instance to a new container today, If you see any weirdness in downloads please report them ASAP, I've also re-enabled IPv6 on files dot. Our allocation is accessible from both HE and Cogent. If anyone has Cogent allocations please do what you can to apply pressure so that Cogent will peer with HE solving the issues with the partitioned IPv6 network. HE is willing to peer, its Cogent that is refusing. Thanks, -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest

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Auto Changing stun/rtp/dtls port every second?
by Stephen Dame 12 Jan '15

12 Jan '15

Auto Changing stun/rtp/dtls port every second? Just built from master FreeSWITCH Version 1.5.15b+git~20150109T000119Z~d199060249~64bit (git d199060 2015-01-09 00:01:19Z 64bit) And when connecting with chrome web-rtc thru sip.js I noticed switch appears to be changing or flipping ports every second bouncing between them. Happens with Chrome, But not Firefox. 2015-01-10 01:57:22.095932 [NOTICE] switch_rtp.c:1139 Auto Changing stun/rtp/dtls port from xx.xx.xx.xx:42748 to xx.xx.xx.xx:42751 2015-01-10 01:57:22.535937 [NOTICE] switch_rtp.c:1139 Auto Changing stun/rtp/dtls port from xx.xx.xx.xx:42751 to xx.xx.xx.xx:42748 2015-01-10 01:57:23.095959 [NOTICE] switch_rtp.c:1139 Auto Changing stun/rtp/dtls port from xx.xx.xx.xx:42748 to xx.xx.xx.xx:42751 2015-01-10 01:57:23.495890 [NOTICE] switch_rtp.c:1139 Auto Changing stun/rtp/dtls port from xx.xx.xx.xx:42751 to xx.xx.xx.xx:42748 2015-01-10 01:57:23.995945 [NOTICE] switch_rtp.c:1139 Auto Changing stun/rtp/dtls port from xx.xx.xx.xx:42748 to xx.xx.xx.xx:42751 ..continues on every second for duration of call. Audio connection seems to be working fine, but have never seen this in prior builds. Not clear to me if this is a real issue, or just annoyance in fs_cli console. Thanks in advance for any suggestions. I have included the SDPs below for both Chrome and FireFox. Regards, Stephen CHROME REMOTE 2015-01-10 02:40:03.535934 [DEBUG] sofia.c:6624 Remote SDP: v=0 o=- 944480236656755000 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS YHfGEaiCgFgkdGUVwVMwsXG2j4QFHe1h13N4 m=audio 44646 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 xx.xx.xx.xx a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=rtcp:44646 IN IP4 xx.xx.xx.xx a=candidate:331141188 1 udp 2122260223 192.168.99.220 57065 typ host generation 0 a=candidate:331141188 2 udp 2122260223 192.168.99.220 57065 typ host generation 0 a=candidate:693976877 1 udp 2122194687 192.168.52.1 57066 typ host generation 0 a=candidate:693976877 2 udp 2122194687 192.168.52.1 57066 typ host generation 0 a=candidate:2448668656 1 udp 2122129151 192.168.142.1 57067 typ host generation 0 a=candidate:2448668656 2 udp 2122129151 192.168.142.1 57067 typ host generation 0 a=candidate:4031593493 1 udp 2122063615 192.168.99.193 57068 typ host generation 0 a=candidate:4031593493 2 udp 2122063615 192.168.99.193 57068 typ host generation 0 a=candidate:1564421300 1 tcp 1518280447 192.168.99.220 0 typ host tcptype active generation 0 a=candidate:1564421300 2 tcp 1518280447 192.168.99.220 0 typ host tcptype active generation 0 a=candidate:1742652381 1 tcp 1518214911 192.168.52.1 0 typ host tcptype active generation 0 a=candidate:1742652381 2 tcp 1518214911 192.168.52.1 0 typ host tcptype active generation 0 a=candidate:3748678400 1 tcp 1518149375 192.168.142.1 0 typ host tcptype active generation 0 a=candidate:3748678400 2 tcp 1518149375 192.168.142.1 0 typ host tcptype active generation 0 a=candidate:3201220837 1 tcp 1518083839 192.168.99.193 0 typ host tcptype active generation 0 a=candidate:3201220837 2 tcp 1518083839 192.168.99.193 0 typ host tcptype active generation 0 a=candidate:3744591831 1 udp 1686052607 xx.xx.xx.xx 44646 typ srflx raddr 192.168.99.220 rport 57065 generation 0 a=candidate:3744591831 2 udp 1686052607 xx.xx.xx.xx 44646 typ srflx raddr 192.168.99.220 rport 57065 generation 0 a=candidate:1019216774 1 udp 1685855999 xx.xx.xx.xx 44647 typ srflx raddr 192.168.99.193 rport 57068 generation 0 a=candidate:1019216774 2 udp 1685855999 xx.xx.xx.xx 44647 typ srflx raddr 192.168.99.193 rport 57068 generation 0 a=ice-ufrag:vWJYtjRU7/AK7HTV a=ice-pwd:tPowc1rLFI4tlG9OuSJsZmhm a=ice-options:google-ice a=fingerprint:sha-256 60:DB:2E:2E:4C:EC:C8:E4:E8:9A:0A:DD:9E:5B:F5:72:7A:AA:E2:BA:4C:B5:6E:77:1B:0 8:CA:D0:33:69:76:18 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=rtcp-mux a=maxptime:60 a=ssrc:207143798 cname:RQHJDeTxW6AqEXek a=ssrc:207143798 msid:YHfGEaiCgFgkdGUVwVMwsXG2j4QFHe1h13N4 4c7d7f12-cdc6-4fc7-8fde-ab184c4374c6 a=ssrc:207143798 mslabel:YHfGEaiCgFgkdGUVwVMwsXG2j4QFHe1h13N4 a=ssrc:207143798 label:4c7d7f12-cdc6-4fc7-8fde-ab184c4374c6 2015-01-10 02:40:03.555959 [DEBUG] sofia.c:6890 (sofia/external/ID-llll(a)sb.domain.com) State Change CS_NEW -> CS_INIT LOCAL CHROME 2015-01-10 02:40:03.555959 [DEBUG] mod_sofia.c:780 Local SDP sofia/external/ID-llll(a)sb.domain.com: v=0 o=FreeSWITCH 1420828455 1420828456 IN IP4 126.254.27.18 s=FreeSWITCH c=IN IP4 126.254.27.18 t=0 0 a=msid-semantic: WMS nURTbPNjtYX6b7AHGQff8WLdypK5IlTV m=audio 29148 UDP/TLS/RTP/SAVPF 111 126 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:126 telephone-event/8000 a=ptime:20 a=sendrecv a=fingerprint:sha-256 C5:4F:B0:97:3E:08:F7:4D:28:D4:DC:66:B9:CB:40:D1:43:14:00:7F:7E:F9:0B:FE:3D:A D:AC:11:1C:09:44:76 a=rtcp-mux a=rtcp:29148 IN IP4 126.254.27.18 a=ssrc:1689569427 cname:5AF6gjgz74w81WFy a=ssrc:1689569427 msid:nURTbPNjtYX6b7AHGQff8WLdypK5IlTV a0 a=ssrc:1689569427 mslabel:nURTbPNjtYX6b7AHGQff8WLdypK5IlTV a=ssrc:1689569427 label:nURTbPNjtYX6b7AHGQff8WLdypK5IlTVa0 a=ice-ufrag:DLDCxeGVugR3rAOm a=ice-pwd:1kj7tqxqkTpbGVAaw3CU78Gs a=candidate:4398516811 1 udp 659136 126.254.27.18 29148 typ host generation 0 ---------------------------------------------------------------------------- ---------------------------------------------------------------- REMOTE FIREFOX 2015-01-10 02:54:48.315868 [DEBUG] sofia.c:6624 Remote SDP: v=0 o=Mozilla-SIPUA-34.0.5 24651 0 IN IP4 0.0.0.0 s=SIP Call t=0 0 a=ice-ufrag:0c83e766 a=ice-pwd:3e4fe41a976d4d295b87ef47898f1bb1 a=fingerprint:sha-256 A4:64:42:CB:31:42:04:21:4B:23:4A:44:93:73:2C:64:73:AC:CB:64:DD:0B:00:CA:ED:1 C:35:5A:F8:19:D2:2B m=audio 9 UDP/TLS/RTP/SAVPF 109 9 0 8 101 c=IN IP4 127.0.0.1 a=rtpmap:109 opus/48000/2 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=setup:actpass a=rtcp-mux a=candidate:0 1 UDP 2128609535 192.168.99.193 50443 typ host a=candidate:2 1 UDP 2128543999 192.168.99.220 50444 typ host a=candidate:4 1 UDP 2128478463 192.168.142.1 50445 typ host a=candidate:6 1 UDP 2128412927 192.168.52.1 50446 typ host a=candidate:0 2 UDP 2128609534 192.168.99.193 50447 typ host a=candidate:2 2 UDP 2128543998 192.168.99.220 50448 typ host a=candidate:4 2 UDP 2128478462 192.168.142.1 50449 typ host a=candidate:6 2 UDP 2128412926 192.168.52.1 50450 typ host a=candidate:3 1 UDP 1692401663 xx.xx.xx.xx 41316 typ srflx raddr 192.168.99.220 rport 50444 a=candidate:1 1 UDP 1692467199 xx.xx.xx.xx 41315 typ srflx raddr 192.168.99.193 rport 50443 a=candidate:3 2 UDP 1692401662 xx.xx.xx.xx 41318 typ srflx raddr 192.168.99.220 rport 50448 a=candidate:1 2 UDP 1692467198 xx.xx.xx.xx 41317 typ srflx raddr 192.168.99.193 rport 50447 2015-01-10 02:54:48.315868 [DEBUG] sofia.c:6890 (sofia/external/ID-sss(a)sb.domain.com) State Change CS_NEW -> CS_INIT LOCAL FIREFOX 2015-01-10 02:54:48.335813 [DEBUG] mod_sofia.c:780 Local SDP sofia/external/ID-sss(a)sb.domain.com: v=0 o=FreeSWITCH 1420830508 1420830509 IN IP4 126.254.27.18 s=FreeSWITCH c=IN IP4 126.254.27.18 t=0 0 a=msid-semantic: WMS 8zLyY4o5MRbtWMdeKRpKz22ilDyNZwPX m=audio 27980 UDP/TLS/RTP/SAVPF 109 101 a=rtpmap:109 opus/48000/2 a=fmtp:109 useinbandfec=1; usedtx=1; maxaveragebitrate=30000; maxplaybackrate=48000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv a=fingerprint:sha-256 C5:4F:B0:97:3E:08:F7:4D:28:D4:DC:66:B9:CB:40:D1:43:14:00:7F:7E:F9:0B:FE:3D:A D:AC:11:1C:09:44:76 a=rtcp-mux a=rtcp:27980 IN IP4 126.254.27.18 a=ssrc:1689570312 cname:VqmJkBFXjGg5ik9T a=ssrc:1689570312 msid:8zLyY4o5MRbtWMdeKRpKz22ilDyNZwPX a0 a=ssrc:1689570312 mslabel:8zLyY4o5MRbtWMdeKRpKz22ilDyNZwPX a=ssrc:1689570312 label:8zLyY4o5MRbtWMdeKRpKz22ilDyNZwPXa0 a=ice-ufrag:1otrhmztPY6EkLs6 a=ice-pwd:2FGc6bZfI6aRhaA9oUqw6QPL a=candidate:1819979940 1 udp 659136 126.254.27.18 27980 typ host generation 0

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Strange noise when using conference_set_auto_outcall
by Oleg Stolyar 12 Jan '15

12 Jan '15

Hi guys, When I use conference_set_auto_outcall to dial a PSTN number via a SIP trunk (Level3 is the provider), the existing conference participants start hearing noise after a couple of seconds. It happens whether at that time the line is still ringing or the PSTN party picked up.. The noise persists through the session when the PSTN party is silent. The noise does not happen when I use the simple bridge to dial the PSTN number instead of a conference, so I don't think it's the PSTN provider's fault. On the other hand, when I use the same settings to outcall another FS server, the noise is not there. Is there some conference setting I am missing? Has anyone been able to successfully use set_auto_outcall to call PSTN numbers? I am on FreeSWITCH Version 1.5.14b+git~20140730T171747Z~5075d4af0d~64bit (git 5075d4a 2014-07-30 17:17:47Z 64bit) I am trying to reproduce this on the latest master but having trouble building on my old CentOS 5.9. I'll keep trying.

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Freeswitch + serial forking at opensips
by John Nash 12 Jan '15

12 Jan '15

Hello, I am facing an issue when using opensips to serial fork Invites to some SIP gateways and using freeswitch in the middle as b2bua to send calls. That means if first Invite fails opensips will pick next available gateway and append branch to send this again to freeswitch. But I have noticed when first gateway sends a failed response (and freeswitch log shows call destroyed) and second invite is sent (Same call id, cseq but different branch tag) , freeswitch seem to have old transaction alive and sends 482 Request merged response. I saw in mail archives that it was discussed few years back but there was no conclusion. Is there a reason freeswitch keeps transaction alive (for 4 seconds I think) even if it received "ACK" for failed response? Regards John

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Doc-Sprint Friday 16 January 2015
by Areski 12 Jan '15

12 Jan '15

Hi Everyone, We are organizing an other Sprint to work on FreeSWITCH Documentation on *Friday 16 January 2015 at 10am CT*. It will be 4 hours long but you can join for less time. The Doc-sprint will focus on migrating the remaining pages from old FS Wiki (https://wiki.freeswitch.org) to Confluence ( https://freeswitch.org/confluence) We will use this IRC channel during the sprint: #freeswitch-docs and we will be tracking our work on a spreadsheet: https://docs.google.com/spreadsheets/d/1qsG-kRymvKlNBapnBLw86W130VdbnK6naYa… to avoid working on the same content. So during the sprint, please change the page "Status" you are working on to "Editing" with your name next to it. Some extra information: - https://freeswitch.org/confluence/display/FREESWITCH/Wiki+Migration - https://freeswitch.org/confluence/display/FREESWITCH/Contributing+Documenta… Let us know if you have any question. We hope to get a maximum number of people signed up! So, who is in? -- Kind regards, /Areski <http://www.linkedin.com/in/areski>

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Re: [Freeswitch-users] REg IVR and codec negotiation
by indra sena 12 Jan '15

12 Jan '15

Hi , I have observed in freeswitch that, In case of IVR scenario prior to the bridge FreeSwitch plays IVR with the 1st priority codec then invite to the bridge endpoint (B-leg) with SDP containing only the codec negotiated for IVR. Also the answer (183 ) to originator(A-leg) with SDP containing only with IVR negotiated codec. I am having one issue here for example. Originator(A-leg) sends invite with G729, G711 and Freeswitch negotiated with G729 and starts playing IVR with G729 and answered(183) to originator with SDP containing only G729. And Invited to termination endpoint with SDP having only G729 and termination gateway having support of only G711, and it is rejecting call after receiving invite with only G729. It should be work like after IVR , invite can be having sdp with all the originator supported codecs and if termination (b-leg) having different priority codecs then it can send re-Invite sanding the same. I have observed one issue in freeswitch Jeera ( https://freeswitch.org/jira/browse/FS-880) , but there is no solution in that page. Do we have any solution for this right now ? Your answers will be very much helpful for me. I appreciate if you can quick response or give some solution for this. Thanks in advance. Thanks & Regards, GISR..

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Set From User in SIP request
by jeremy ardley 12 Jan '15

12 Jan '15

Hi, I have a DID provider with multiple incoming numbers. I can send outgoing calls on particular DID numbers by sending a SIP request with the From header formatted as From: dddddddddd <sip:dddddddd@sip.provider.com>;tag=yyy Where the From name is the DID in full national number form (10 digit) and the From user is the 8 digit account number. Each unique DID has a matching unique account number. I have internal extensions mapped to different DID numbers. I need a mechanism to feed different DID number and account number pairs to the gateway for each outgoing call. I hope to do that via the internal extension xml files e.g. <include> <user id="200"> <params> <param name="password" value="xxxxxx"/> <param name="vm-password" value="200"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="200"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Michael J Fox"/> <variable name="effective_caller_id_number" value="200"/> <variable name="outbound_caller_id_name" value="0000000001"/> <variable name="outbound_caller_id_number" value="0000000001"/> <variable name="outbound_caller_from_user" value="01234567"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> The question is how to pass the new variable outbound_caller_from_user to the gateway and get it substituted in the user field of the SIP From header? I have a secondary issue in that I can set the outbound DID part of the >From Header but it has quotes around it From: "dddddddddd" <sip:dddddddd@sip.provider.com>;tag=yyy Which is undesirable and it would be helpful to suppress that in Freeswitch itself. Thanks,

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Call back through SIP trunk
by Sina Owolabi 10 Jan '15

10 Jan '15

Hi List! FreeSWITCHNewbie here. Please can I have some guidance on how to setup call back? I would like to be able to dial the DID attached to the SIP trunk Freeswitch is registered to, and then have freeSWITCH hang up the call and dial the caller id number back through any other SIP trunk FreeSWITCH Is registered with, but with the origination number set to the DID that the first call came through in the first place. Please is this possible just through the dial plan? Thanks for any help!

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