From mrjoli021 at gmail.com Tue Feb 1 15:31:47 2022 From: mrjoli021 at gmail.com (Joli Martinez) Date: Tue, 1 Feb 2022 10:31:47 -0500 Subject: [Freeswitch-users] cidlookup not working Message-ID: Hello, I have enabled the module cidlookup and added my config to the cidlookup.conf.xml file. I have also reloaded the module as well as restarted freeswitch. Freeswitch does not attempt to send any queries out to opencnam. I am doing a tcpdump on their IP's from the box and it is not even attempting to go out to them on port 443 or 53. I am running FS 1.10.7 on Debian 11. Any suggestions? &auth_token=”/> -------------- next part -------------- An HTML attachment was scrubbed... URL: From dragos at freeswitch.org Wed Feb 2 12:18:37 2022 From: dragos at freeswitch.org (Dragos Oancea) Date: Wed, 2 Feb 2022 14:18:37 +0200 Subject: [Freeswitch-users] cidlookup not working In-Reply-To: References: Message-ID: Do you have the "cidlookup" command available in fs_cli ? Are you using the "cidlookup" app ? On Tue, Feb 1, 2022 at 5:32 PM Joli Martinez wrote: > Hello, > > I have enabled the module cidlookup and added my config to the > cidlookup.conf.xml file. I have also reloaded the module as well as > restarted freeswitch. Freeswitch does not attempt to send any queries out > to opencnam. I am doing a tcpdump on their IP's from the box and it is not > even attempting to go out to them on port 443 or 53. I am running FS > 1.10.7 on Debian 11. Any suggestions? > > > > > phone/+${caller_id_number}?account_sid=&auth_token=”/> > > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From shahnawaj.khan1990 at gmail.com Wed Feb 2 13:21:37 2022 From: shahnawaj.khan1990 at gmail.com (Shahnawaj Khan) Date: Wed, 2 Feb 2022 18:51:37 +0530 Subject: [Freeswitch-users] Playback file stopping abruptly from freeswitch 1.10.5 and onwards Message-ID: Hi All, Recently we have moved from freeswitch 1.10.3 to 1.10.5. and we observed that while executing the following command through ESL Command Execute playback(random.wav) freeswitch is stopping the playback earlier than the file playback duration.In this setup we are using freeswitch to generate the call as well. Our setup looks similar to this. Freeswitch_1------>opensips------>Freeswitch_2 Freeswitch_1: a) Originate and park b) playback some file on park (Using command though ESL) Freeswitch_2: a) playback some file on park (Using command though ESL) In Attached logs freeswitch has stopped the playback after 3 seconds but the file is of 10 seconds. But in another scenario where we replace the freeswitch_1 with a manual call using a softphone, Freeswitch_2 plays the file for the whole 10 seconds. Please let me know if anyone has faced the same issue or any insight regarding it. We observed the same issue with 1.10.7 but working fine in our old 1.10.3 setup. Thanks & Regards, Shahnawaj -------------- next part -------------- 2021-10-20 12:18:03.036893 [DEBUG] switch_ivr_originate.c:2242 Parsing global variables 2021-10-20 12:18:03.036893 [NOTICE] switch_channel.c:1118 New Channel sofia/3clogic_external/009911 at randomdomain.com:5507 [c5d6f887-319f-11ec-bf87-0265fdb72ff7] 2021-10-20 12:18:03.036893 [DEBUG] mod_sofia.c:5089 (sofia/3clogic_external/009911 at randomdomain.com:5507) State Change CS_NEW -> CS_INIT 2021-10-20 12:18:03.036893 [DEBUG] switch_core_session.c:641 sofia/3clogic_external/009911 at randomdomain.com:5507 set UUID=c5d6f887-319f-11ec-bf87-0265fdb72ff7 2021-10-20 12:18:03.036893 [DEBUG] switch_core_state_machine.c:585 (sofia/3clogic_external/009911 at randomdomain.com:5507) Running State Change CS_INIT (Cur 1 Tot 6) 2021-10-20 12:18:03.036893 [DEBUG] switch_core_state_machine.c:628 (sofia/3clogic_external/009911 at randomdomain.com:5507) State INIT 2021-10-20 12:18:03.036893 [DEBUG] mod_sofia.c:93 sofia/3clogic_external/009911 at randomdomain.com:5507 SOFIA INIT 2021-10-20 12:18:03.036893 [DEBUG] sofia_glue.c:1618 sofia/3clogic_external/009911 at randomdomain.com:5507 sending invite version: 1.10.5-release git 25569c1 2020-08-18 18:51:21Z 64bit Local SDP: v=0 o=FreeSWITCH 1634702247 1634702248 IN IP4 X.X.X.X s=FreeSWITCH c=IN IP4 X.X.X.X t=0 0 m=audio 30036 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2021-10-20 12:18:03.036893 [DEBUG] switch_core_state_machine.c:40 sofia/3clogic_external/009911 at randomdomain.com:5507 Standard INIT 2021-10-20 12:18:03.036893 [DEBUG] switch_core_state_machine.c:48 (sofia/3clogic_external/009911 at randomdomain.com:5507) State Change CS_INIT -> CS_ROUTING 2021-10-20 12:18:03.036893 [DEBUG] switch_core_state_machine.c:628 (sofia/3clogic_external/009911 at randomdomain.com:5507) State INIT going to sleep send 1053 bytes to udp/[X.X.X.X]:5507 at 12:18:03.041869: ------------------------------------------------------------------------ INVITE sip:009911 at randomdomain.com:5507 SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5080;rport;branch=z9hG4bKvp1p031Krm5Za Max-Forwards: 70 From: ;tag=jeK6trDeyj31g To: Call-ID: 9d356e34-ac42-123a-0fa2-0265fdb72ff7 CSeq: 42812349 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.10.5-release+git~20200818T185121Z~25569c1631~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 220 X-FS-Support: update_display,send_info Remote-Party-ID: ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1634702247 1634702248 IN IP4 X.X.X.X s=FreeSWITCH c=IN IP4 X.X.X.X t=0 0 m=audio 30036 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 2021-10-20 12:18:03.036893 [DEBUG] switch_core_state_machine.c:585 (sofia/3clogic_external/009911 at randomdomain.com:5507) Running State Change CS_ROUTING (Cur 1 Tot 6) 2021-10-20 12:18:03.036893 [DEBUG] sofia.c:7326 Channel sofia/3clogic_external/009911 at randomdomain.com:5507 entering state [calling][0] 2021-10-20 12:18:03.036893 [DEBUG] switch_core_state_machine.c:644 (sofia/3clogic_external/009911 at randomdomain.com:5507) State ROUTING 2021-10-20 12:18:03.036893 [DEBUG] mod_sofia.c:154 sofia/3clogic_external/009911 at randomdomain.com:5507 SOFIA ROUTING 2021-10-20 12:18:03.036893 [DEBUG] switch_ivr_originate.c:67 (sofia/3clogic_external/009911 at randomdomain.com:5507) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2021-10-20 12:18:03.036893 [DEBUG] switch_core_state_machine.c:644 (sofia/3clogic_external/009911 at randomdomain.com:5507) State ROUTING going to sleep 2021-10-20 12:18:03.036893 [DEBUG] switch_core_state_machine.c:585 (sofia/3clogic_external/009911 at randomdomain.com:5507) Running State Change CS_CONSUME_MEDIA (Cur 1 Tot 6) 2021-10-20 12:18:03.036893 [DEBUG] switch_core_state_machine.c:663 (sofia/3clogic_external/009911 at randomdomain.com:5507) State CONSUME_MEDIA 2021-10-20 12:18:03.036893 [DEBUG] switch_core_state_machine.c:663 (sofia/3clogic_external/009911 at randomdomain.com:5507) State CONSUME_MEDIA going to sleep recv 361 bytes from udp/[X.X.X.X]:5507 at 12:18:03.042319: ------------------------------------------------------------------------ SIP/2.0 100 Giving a try Via: SIP/2.0/UDP X.X.X.X:5080;received=X.X.X.X;rport=5080;branch=z9hG4bKvp1p031Krm5Za From: ;tag=jeK6trDeyj31g To: Call-ID: 9d356e34-ac42-123a-0fa2-0265fdb72ff7 CSeq: 42812349 INVITE Server: OpenSIPS (2.2.4 (x86_64/linux)) Content-Length: 0 recv 1353 bytes from udp/[X.X.X.X]:5507 at 12:18:03.047406: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP X.X.X.X:5080;received=X.X.X.X;rport=5080;branch=z9hG4bKvp1p031Krm5Za Record-Route: Record-Route: From: ;tag=jeK6trDeyj31g To: ;tag=Z8aHy6Xjc6jBe Call-ID: 9d356e34-ac42-123a-0fa2-0265fdb72ff7 CSeq: 42812349 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.10.5-release+git~20200818T185121Z~25569c1631~64bit 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 Content-Type: application/sdp Content-Disposition: session Content-Length: 218 X-FS-Display-Name: 009911 X-FS-Display-Number: sip:009911 at randomdomain.com X-FS-Support: update_display,send_info Remote-Party-ID: "009911" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1634701115 1634701116 IN IP4 X.X.X.X s=FreeSWITCH c=IN IP4 X.X.X.X t=0 0 m=audio 31168 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 2021-10-20 12:18:03.036893 [INFO] sofia.c:1369 sofia/3clogic_external/009911 at randomdomain.com:5507 Update Callee ID to "009911" 2021-10-20 12:18:03.036893 [DEBUG] sofia.c:7326 Channel sofia/3clogic_external/009911 at randomdomain.com:5507 entering state [proceeding][183] 2021-10-20 12:18:03.036893 [DEBUG] sofia.c:7336 Remote SDP: v=0 o=FreeSWITCH 1634701115 1634701116 IN IP4 X.X.X.X s=FreeSWITCH c=IN IP4 X.X.X.X t=0 0 m=audio 31168 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 2021-10-20 12:18:03.036893 [NOTICE] sofia.c:7339 Pre-Answer sofia/3clogic_external/009911 at randomdomain.com:5507! 2021-10-20 12:18:03.036893 [DEBUG] switch_channel.c:3565 (sofia/3clogic_external/009911 at randomdomain.com:5507) Callstate Change DOWN -> EARLY 2021-10-20 12:18:03.036893 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2021-10-20 12:18:03.036893 [DEBUG] switch_core_media.c:5649 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 2021-10-20 12:18:03.036893 [DEBUG] switch_core_media.c:5510 Set telephone-event payload to 101 at 8000 2021-10-20 12:18:03.036893 [DEBUG] switch_core_media.c:3839 Set Codec sofia/3clogic_external/009911 at randomdomain.com:5507 PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2021-10-20 12:18:03.036893 [DEBUG] switch_core_codec.c:111 sofia/3clogic_external/009911 at randomdomain.com:5507 Original read codec set to PCMU:0 2021-10-20 12:18:03.036893 [DEBUG] switch_core_media.c:5853 Set telephone-event payload to 101 at 8000 2021-10-20 12:18:03.036893 [DEBUG] switch_core_media.c:5911 sofia/3clogic_external/009911 at randomdomain.com:5507 Set 2833 dtmf send payload to 101 recv payload to 101 2021-10-20 12:18:03.036893 [DEBUG] switch_core_media.c:8663 AUDIO RTP [sofia/3clogic_external/009911 at randomdomain.com:5507] 20.1.155.222 port 30036 -> X.X.X.X port 31168 codec: 0 ms: 20 2021-10-20 12:18:03.036893 [DEBUG] switch_rtp.c:4472 Not using a timer 2021-10-20 12:18:03.036893 [DEBUG] switch_core_media.c:2554 Setting Jitterbuffer to 60ms (3 frames) (150 max frames) 2021-10-20 12:18:03.036893 [DEBUG] switch_core_media.c:8977 sofia/3clogic_external/009911 at randomdomain.com:5507 Set 2833 dtmf send payload to 101 2021-10-20 12:18:03.036893 [DEBUG] switch_core_media.c:8984 sofia/3clogic_external/009911 at randomdomain.com:5507 Set 2833 dtmf receive payload to 101 2021-10-20 12:18:03.036893 [DEBUG] switch_core_media.c:9007 sofia/3clogic_external/009911 at randomdomain.com:5507 Set rtp dtmf delay to 40 2021-10-20 12:18:03.056895 [DEBUG] switch_ivr_originate.c:3852 Originate Resulted in Success: [sofia/3clogic_external/009911 at randomdomain.com:5507] 2021-10-20 12:18:03.056895 [INFO] switch_channel.c:3213 sofia/3clogic_external/009911 at randomdomain.com:5507 Flipping CID from "" to "009911" <009911> 2021-10-20 12:18:03.056895 [DEBUG] mod_commands.c:5094 (sofia/3clogic_external/009911 at randomdomain.com:5507) State Change CS_CONSUME_MEDIA -> CS_EXECUTE 2021-10-20 12:18:03.056895 [DEBUG] switch_core_state_machine.c:585 (sofia/3clogic_external/009911 at randomdomain.com:5507) Running State Change CS_EXECUTE (Cur 1 Tot 6) 2021-10-20 12:18:03.056895 [DEBUG] switch_core_state_machine.c:651 (sofia/3clogic_external/009911 at randomdomain.com:5507) State EXECUTE 2021-10-20 12:18:03.056895 [DEBUG] mod_sofia.c:209 sofia/3clogic_external/009911 at randomdomain.com:5507 SOFIA EXECUTE 2021-10-20 12:18:03.056895 [DEBUG] switch_core_state_machine.c:329 sofia/3clogic_external/009911 at randomdomain.com:5507 Standard EXECUTE EXECUTE [depth=0] sofia/3clogic_external/009911 at randomdomain.com:5507 park() 2021-10-20 12:18:03.076894 [DEBUG] switch_ivr.c:632 sofia/3clogic_external/009911 at randomdomain.com:5507 Command Execute [depth=1] set(parent_verb=1) EXECUTE [depth=1] sofia/3clogic_external/009911 at randomdomain.com:5507 set(parent_verb=1) 2021-10-20 12:18:03.076894 [DEBUG] mod_dptools.c:1672 SET sofia/3clogic_external/009911 at randomdomain.com:5507 [parent_verb]=[1] 2021-10-20 12:18:03.076894 [DEBUG] switch_ivr.c:632 sofia/3clogic_external/009911 at randomdomain.com:5507 Command Execute [depth=1] set(playback_terminators=none) EXECUTE [depth=1] sofia/3clogic_external/009911 at randomdomain.com:5507 set(playback_terminators=none) 2021-10-20 12:18:03.076894 [DEBUG] mod_dptools.c:1672 SET sofia/3clogic_external/009911 at randomdomain.com:5507 [playback_terminators]=[none] 2021-10-20 12:18:03.076894 [DEBUG] switch_ivr.c:632 sofia/3clogic_external/009911 at randomdomain.com:5507 Command Execute [depth=1] set(queue_id=c5dabd54-319f-11ec-bf87-0265fdb72ff7) EXECUTE [depth=1] sofia/3clogic_external/009911 at randomdomain.com:5507 set(queue_id=c5dabd54-319f-11ec-bf87-0265fdb72ff7) 2021-10-20 12:18:03.076894 [DEBUG] mod_dptools.c:1672 SET sofia/3clogic_external/009911 at randomdomain.com:5507 [queue_id]=[c5dabd54-319f-11ec-bf87-0265fdb72ff7] 2021-10-20 12:18:03.076894 [DEBUG] switch_ivr.c:632 sofia/3clogic_external/009911 at randomdomain.com:5507 Command Execute [depth=1] playback(random.wav) EXECUTE [depth=1] sofia/3clogic_external/009911 at randomdomain.com:5507 playback(random.wav) 2021-10-20 12:18:03.076894 [WARNING] switch_core_file.c:424 File has 2 channels, muxing to 1 channel will occur. 2021-10-20 12:18:03.076894 [DEBUG] switch_ivr_play_say.c:1488 Codec Activated L16 at 8000hz 1 channels 20ms recv 1314 bytes from udp/[X.X.X.X]:5507 at 12:18:03.450121: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP X.X.X.X:5080;received=X.X.X.X;rport=5080;branch=z9hG4bKvp1p031Krm5Za Record-Route: Record-Route: From: ;tag=jeK6trDeyj31g To: ;tag=Z8aHy6Xjc6jBe Call-ID: 9d356e34-ac42-123a-0fa2-0265fdb72ff7 CSeq: 42812349 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.10.5-release+git~20200818T185121Z~25569c1631~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 218 X-FS-Display-Name: 009911 X-FS-Display-Number: sip:009911 at randomdomain.com X-FS-Support: update_display,send_info Remote-Party-ID: "009911" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1634701115 1634701116 IN IP4 X.X.X.X s=FreeSWITCH c=IN IP4 X.X.X.X t=0 0 m=audio 31168 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 2021-10-20 12:18:03.436893 [INFO] sofia.c:1358 sofia/3clogic_external/009911 at randomdomain.com:5507 Update Caller ID to "009911" 2021-10-20 12:18:03.436893 [DEBUG] sofia.c:7326 Channel sofia/3clogic_external/009911 at randomdomain.com:5507 entering state [completing][200] 2021-10-20 12:18:03.436893 [DEBUG] sofia.c:7333 Duplicate SDP v=0 o=FreeSWITCH 1634701115 1634701116 IN IP4 X.X.X.X s=FreeSWITCH c=IN IP4 X.X.X.X t=0 0 m=audio 31168 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 send 540 bytes to udp/[X.X.X.X]:5507 at 12:18:03.450901: ------------------------------------------------------------------------ ACK sip:009911 at X.X.X.X:5080;transport=udp SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5080;rport;branch=z9hG4bKXZtF2yjQNXUjp Route: Route: Max-Forwards: 70 From: ;tag=jeK6trDeyj31g To: ;tag=Z8aHy6Xjc6jBe Call-ID: 9d356e34-ac42-123a-0fa2-0265fdb72ff7 CSeq: 42812349 ACK Contact: Content-Length: 0 2021-10-20 12:18:03.436893 [DEBUG] sofia.c:7326 Channel sofia/3clogic_external/009911 at randomdomain.com:5507 entering state [ready][200] 2021-10-20 12:18:03.436893 [NOTICE] sofia.c:8445 Channel [sofia/3clogic_external/009911 at randomdomain.com:5507] has been answered 2021-10-20 12:18:03.436893 [DEBUG] switch_channel.c:3865 (sofia/3clogic_external/009911 at randomdomain.com:5507) Callstate Change EARLY -> ACTIVE 2021-10-20 12:18:05.936893 [DEBUG] switch_rtp.c:7759 Correct audio ip/port confirmed. 2021-10-20 12:18:06.036893 [DEBUG] switch_ivr_play_say.c:1933 done playing file random.wav 2021-10-20 12:18:06.416892 [DEBUG] switch_ivr.c:632 sofia/3clogic_external/009911 at randomdomain.com:5507 Command Execute [depth=1] hangup(Queue is Empty) EXECUTE [depth=1] sofia/3clogic_external/009911 at randomdomain.com:5507 hangup(Queue is Empty) 2021-10-20 12:18:06.416892 [NOTICE] mod_dptools.c:1380 Hangup sofia/3clogic_external/009911 at randomdomain.com:5507 [CS_EXECUTE] [NORMAL_CLEARING] 2021-10-20 12:18:06.416892 [DEBUG] switch_core_session.c:2905 sofia/3clogic_external/009911 at randomdomain.com:5507 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2021-10-20 12:18:06.416892 [DEBUG] switch_core_session.c:2905 sofia/3clogic_external/009911 at randomdomain.com:5507 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2021-10-20 12:18:06.416892 [DEBUG] switch_core_state_machine.c:651 (sofia/3clogic_external/009911 at randomdomain.com:5507) State EXECUTE going to sleep 2021-10-20 12:18:06.416892 [DEBUG] switch_core_state_machine.c:585 (sofia/3clogic_external/009911 at randomdomain.com:5507) Running State Change CS_HANGUP (Cur 1 Tot 6) 2021-10-20 12:18:06.416892 [DEBUG] switch_ivr.c:679 sofia/3clogic_external/009911 at randomdomain.com:5507 skip receive message [AUDIO_SYNC] (channel is hungup already) 2021-10-20 12:18:06.416892 [DEBUG] switch_core_state_machine.c:848 (sofia/3clogic_external/009911 at randomdomain.com:5507) Callstate Change ACTIVE -> HANGUP 2021-10-20 12:18:06.416892 [DEBUG] switch_core_state_machine.c:850 (sofia/3clogic_external/009911 at randomdomain.com:5507) State HANGUP 2021-10-20 12:18:06.416892 [DEBUG] mod_sofia.c:453 Channel sofia/3clogic_external/009911 at randomdomain.com:5507 hanging up, cause: NORMAL_CLEARING 2021-10-20 12:18:06.416892 [DEBUG] mod_sofia.c:507 Sending BYE to sofia/3clogic_external/009911 at randomdomain.com:5507 2021-10-20 12:18:06.416892 [DEBUG] switch_core_state_machine.c:60 sofia/3clogic_external/009911 at randomdomain.com:5507 Standard HANGUP, cause: NORMAL_CLEARING 2021-10-20 12:18:06.416892 [DEBUG] switch_core_state_machine.c:850 (sofia/3clogic_external/009911 at randomdomain.com:5507) State HANGUP going to sleep 2021-10-20 12:18:06.416892 [DEBUG] switch_core_state_machine.c:620 (sofia/3clogic_external/009911 at randomdomain.com:5507) State Change CS_HANGUP -> CS_REPORTING 2021-10-20 12:18:06.416892 [DEBUG] switch_core_state_machine.c:585 (sofia/3clogic_external/009911 at randomdomain.com:5507) Running State Change CS_REPORTING (Cur 1 Tot 6) 2021-10-20 12:18:06.416892 [DEBUG] switch_core_state_machine.c:936 (sofia/3clogic_external/009911 at randomdomain.com:5507) State REPORTING send 754 bytes to udp/[X.X.X.X]:5507 at 12:18:06.432892: ------------------------------------------------------------------------ BYE sip:009911 at X.X.X.X:5080;transport=udp SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5080;rport;branch=z9hG4bKy8K83S3tj6H5H Route: Route: Max-Forwards: 70 From: ;tag=jeK6trDeyj31g To: ;tag=Z8aHy6Xjc6jBe Call-ID: 9d356e34-ac42-123a-0fa2-0265fdb72ff7 CSeq: 42812350 BYE User-Agent: FreeSWITCH-mod_sofia/1.10.5-release+git~20200818T185121Z~25569c1631~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 2021-10-20 12:18:06.416892 [DEBUG] switch_core_state_machine.c:174 sofia/3clogic_external/009911 at randomdomain.com:5507 Standard REPORTING, cause: NORMAL_CLEARING 2021-10-20 12:18:06.416892 [DEBUG] switch_core_state_machine.c:936 (sofia/3clogic_external/009911 at randomdomain.com:5507) State REPORTING going to sleep 2021-10-20 12:18:06.416892 [DEBUG] switch_core_state_machine.c:611 (sofia/3clogic_external/009911 at randomdomain.com:5507) State Change CS_REPORTING -> CS_DESTROY 2021-10-20 12:18:06.416892 [DEBUG] switch_core_session.c:1726 Session 6 (sofia/3clogic_external/009911 at randomdomain.com:5507) Locked, Waiting on external entities 2021-10-20 12:18:06.416892 [NOTICE] switch_core_session.c:1744 Session 6 (sofia/3clogic_external/009911 at randomdomain.com:5507) Ended 2021-10-20 12:18:06.416892 [NOTICE] switch_core_session.c:1748 Close Channel sofia/3clogic_external/009911 at randomdomain.com:5507 [CS_DESTROY] 2021-10-20 12:18:06.416892 [DEBUG] switch_core_state_machine.c:739 (sofia/3clogic_external/009911 at randomdomain.com:5507) Running State Change CS_DESTROY (Cur 0 Tot 6) 2021-10-20 12:18:06.416892 [DEBUG] switch_core_state_machine.c:749 (sofia/3clogic_external/009911 at randomdomain.com:5507) State DESTROY 2021-10-20 12:18:06.416892 [DEBUG] mod_sofia.c:364 sofia/3clogic_external/009911 at randomdomain.com:5507 SOFIA DESTROY 2021-10-20 12:18:06.416892 [DEBUG] switch_core_state_machine.c:181 sofia/3clogic_external/009911 at randomdomain.com:5507 Standard DESTROY 2021-10-20 12:18:06.416892 [DEBUG] switch_core_state_machine.c:749 (sofia/3clogic_external/009911 at randomdomain.com:5507) State DESTROY going to sleep recv 536 bytes from udp/[X.X.X.X]:5507 at 12:18:06.454758: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP X.X.X.X:5080;received=X.X.X.X;rport=5080;branch=z9hG4bKy8K83S3tj6H5H From: ;tag=jeK6trDeyj31g To: ;tag=Z8aHy6Xjc6jBe Call-ID: 9d356e34-ac42-123a-0fa2-0265fdb72ff7 CSeq: 42812350 BYE User-Agent: FreeSWITCH-mod_sofia/1.10.5-release+git~20200818T185121Z~25569c1631~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Content-Length: 0 From gledimusa08 at gmail.com Tue Feb 1 18:22:33 2022 From: gledimusa08 at gmail.com (Gledi Musa) Date: Tue, 1 Feb 2022 19:22:33 +0100 Subject: [Freeswitch-users] SRTP calls randomly have no audio In-Reply-To: References: Message-ID: Hello, I'm using an instance of FreeSWITCH Version 1.10.7-release-19 in a Debian 11 VM, with OpenSSL 1.1.1k. The carrier only supports TLSv1, so I have hardcoded it in vars.xml and openssl.cnf to use only it, instead of TLSv1.2. On random calls there is no way audio, and without any changes, the next calls work, and then some other calls have no way audio. The logs don't throw any errors, but in the cases where there is no audio, I see that the DTLS negotiation doesn't go past HANDSHAKE, and stays like that for minutes, without throwing any errors. I have tried to set "legacyDTLS=1" as a channel variable, but that doesn't seem to have any effect. Any idea how to approach this? -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Thu Feb 3 12:36:17 2022 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 3 Feb 2022 12:36:17 +0000 Subject: [Freeswitch-users] Leg 480 kills all other legs? Message-ID: Hello all, I’m calling multiple destinations separated by | One of the destinations is returning 480 and fs is canceling all other legs. I have continue_on_fail to true (I also tried with 480 and “Temp Unava…”) It was my understanding fs would just ignore that leg (no media is received) and allow the other legs continue… What am I missing? Thanks all! David -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From fs at szmidt.org Thu Feb 3 15:19:06 2022 From: fs at szmidt.org (fs) Date: Thu, 3 Feb 2022 10:19:06 -0500 Subject: [Freeswitch-users] No hang up, transfers or hold Message-ID: <8b5d6e36-edf3-38ab-c999-8c271ff65cd6@szmidt.org> My thought is that others may have run into this and have some pointers to share. I have Freeswitch 4.2.5 Branch 4.2 Switch 1.6.20 64 bit on Linux FusionPBX 3.16.0-11-amd64 #1 SMP Debian 3.16.84-1 (2020-06-09) x86_64 GNU/Linux. I'm using default settings as far as I know/recall. Obviously something must not be as it should be. My phones are a Polycom 650 and a Yealink T48G. My external test phone is a cell phone. It had only been used with a single extension for years until now when I added a second phone and extension. Before today I had noticed that hold was not working properly but did not use it. Attempting to transfer external calls between the two fails in the middle of the transfer. I press the transfer button and type the extension and then hit send but the call does not arrive and the call is hung up on the internal side but the external party does not know until something times out. If I put a call on hold it may be lost as well. Brief holds work. If I call out and then try to transfer that call the same thing happens. If I conference the other extension on an inbound call the external party is not getting any audio once the conference is initiated. If I then hangup on one extension the other is not aware that the call is over. The same is true for the external party. I can call the other extension and answer, when I hang up on either one the other does not know it's gone. It does not matter which phone originates or receives. At least audio works well. :) I've placed a FS log file here: https://drive.google.com/file/d/1VdoLqoPhr5mUE1ky1i2SJjXFChp9w-nM/view?usp=sharing The call duration is 2 minutes from 9:30 to 9:32. IP's and phone numbers have been replaced and placed within < > brackets. Internal IP's retains the last 8-bit number. --- fs -------------- next part -------------- A non-text attachment was scrubbed... Name: OpenPGP_signature Type: application/pgp-signature Size: 840 bytes Desc: OpenPGP digital signature URL: From david.villasmil.work at gmail.com Thu Feb 3 15:58:09 2022 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 3 Feb 2022 15:58:09 +0000 Subject: [Freeswitch-users] Leg 480 kills all other legs? In-Reply-To: References: Message-ID: Thoughts? Ideas anyone? On Thu, 3 Feb 2022 at 12:36, David Villasmil wrote: > Hello all, > > I’m calling multiple destinations separated by | > One of the destinations is returning 480 and fs is canceling all other > legs. I have continue_on_fail to true (I also tried with 480 and “Temp > Unava…”) > > It was my understanding fs would just ignore that leg (no media is > received) and allow the other legs continue… > > What am I missing? > > Thanks all! > > David > -- > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From mrjoli021 at gmail.com Thu Feb 3 15:58:17 2022 From: mrjoli021 at gmail.com (Joli Martinez) Date: Thu, 3 Feb 2022 10:58:17 -0500 Subject: [Freeswitch-users] cidlookup not working In-Reply-To: References: Message-ID: Hello, I am able to type cidlookup on the fs_cli and it returns None. The module is loaded, but at no point does it reach out to query the OpenCnam servers. I am doing a tcpdump to their IP's and nothing attempts goes out to them. On Wed, Feb 2, 2022 at 7:43 AM Dragos Oancea wrote: > Do you have the "cidlookup" command available in fs_cli ? > > Are you using the "cidlookup" app ? > > On Tue, Feb 1, 2022 at 5:32 PM Joli Martinez wrote: > >> Hello, >> >> I have enabled the module cidlookup and added my config to the >> cidlookup.conf.xml file. I have also reloaded the module as well as >> restarted freeswitch. Freeswitch does not attempt to send any queries out >> to opencnam. I am doing a tcpdump on their IP's from the box and it is not >> even attempting to go out to them on port 443 or 53. I am running FS >> 1.10.7 on Debian 11. Any suggestions? >> >> >> >> >> > phone/+${caller_id_number}?account_sid=&auth_token=”/> >> >> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From d at d-man.org Thu Feb 3 16:00:00 2022 From: d at d-man.org (Darren) Date: Thu, 3 Feb 2022 16:00:00 +0000 Subject: [Freeswitch-users] cidlookup not working In-Reply-To: References: Message-ID: OpenCNAM was acquired by Neustar and I thought they shutdown the old API server/URL, are you sure you’re hitting the right URL? From: FreeSWITCH-users On Behalf Of Joli Martinez Sent: Thursday, February 3, 2022 7:58 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] cidlookup not working Hello, I am able to type cidlookup on the fs_cli and it returns None. The module is loaded, but at no point does it reach out to query the OpenCnam servers. I am doing a tcpdump to their IP's and nothing attempts goes out to them. On Wed, Feb 2, 2022 at 7:43 AM Dragos Oancea > wrote: Do you have the "cidlookup" command available in fs_cli ? Are you using the "cidlookup" app ? On Tue, Feb 1, 2022 at 5:32 PM Joli Martinez > wrote: Hello, I have enabled the module cidlookup and added my config to the cidlookup.conf.xml file. I have also reloaded the module as well as restarted freeswitch. Freeswitch does not attempt to send any queries out to opencnam. I am doing a tcpdump on their IP's from the box and it is not even attempting to go out to them on port 443 or 53. I am running FS 1.10.7 on Debian 11. Any suggestions? &auth_token=”/> _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From saied.tazari at gmail.com Thu Feb 3 17:08:50 2022 From: saied.tazari at gmail.com (Mohammad-Reza Tazari) Date: Thu, 3 Feb 2022 18:08:50 +0100 Subject: [Freeswitch-users] debian repos for Raspberry Pi broken? Message-ID: Dear Andrey, any news here? Thanks and kind regards, -- Saied Andrey Volk wrote on 21-Dec-2021 21:36: > > Let me see what I can do. May take some time. > > вт, 21 дек. 2021 г. в 23:19, Johan Helsingius >: > > > >* Hi, *> >> >* Trying to install freeswitch on a RPi 4 (Debian 11 Bullseye), *> >* my /etc/apt/sources.list.d/freeswitch.list has: *> >> >* deb http://files.freeswitch.org/repo/deb/rpi/debian-release/ bullseye main *> >* deb-src http://files.freeswitch.org/repo/deb/rpi/debian-release/ bullseye *> >* main *> >> >* but when I try *> >> >* apt-get update && apt-get install -y freeswitch-meta-all *> >> >* I get *> >> >* Unable to locate package freeswitch-meta-all *> >> >* apt-cache search freeswitch only gives me: *> >* libsipwitch-dev - secure peer-to-peer SIP VoIP server - development files *> >* libsipwitch1 - secure peer-to-peer SIP VoIP server - shared libraries *> >* libsipwitch1-dbg - secure peer-to-peer SIP VoIP server - debug symbols *> >* sipwitch - secure peer-to-peer VoIP server for the SIP protocol *> >* sipwitch-cgi - secure peer-to-peer SIP VoIP server - CGI XML-RPC interface *> >> >* Any suggestions? *> >> >* Julf * -------------- next part -------------- An HTML attachment was scrubbed... URL: From mrjoli021 at gmail.com Thu Feb 3 17:10:40 2022 From: mrjoli021 at gmail.com (Joli Martinez) Date: Thu, 3 Feb 2022 12:10:40 -0500 Subject: [Freeswitch-users] cidlookup not working In-Reply-To: References: Message-ID: That is the document they sent me. It might be outdated. ill go back and check with them. Thanks On Thu, Feb 3, 2022 at 11:57 AM Darren wrote: > OpenCNAM was acquired by Neustar and I thought they shutdown the old API > server/URL, are you sure you’re hitting the right URL? > > > > *From:* FreeSWITCH-users *On > Behalf Of *Joli Martinez > *Sent:* Thursday, February 3, 2022 7:58 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] cidlookup not working > > > > Hello, > > > > I am able to type cidlookup on the fs_cli and it returns > None. The module is loaded, but at no point does it reach out to query the > OpenCnam servers. I am doing a tcpdump to their IP's and nothing attempts > goes out to them. > > > > On Wed, Feb 2, 2022 at 7:43 AM Dragos Oancea > wrote: > > Do you have the "cidlookup" command available in fs_cli ? > > > > Are you using the "cidlookup" app ? > > > > On Tue, Feb 1, 2022 at 5:32 PM Joli Martinez wrote: > > Hello, > > > > I have enabled the module cidlookup and added my config to the > cidlookup.conf.xml file. I have also reloaded the module as well as > restarted freeswitch. Freeswitch does not attempt to send any queries out > to opencnam. I am doing a tcpdump on their IP's from the box and it is not > even attempting to go out to them on port 443 or 53. I am running FS > 1.10.7 on Debian 11. Any suggestions? > > > > > > > phone/+${caller_id_number}?account_sid=&auth_token=”/> > > > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From mrjoli021 at gmail.com Thu Feb 3 17:15:35 2022 From: mrjoli021 at gmail.com (Joli Martinez) Date: Thu, 3 Feb 2022 12:15:35 -0500 Subject: [Freeswitch-users] cidlookup not working In-Reply-To: References: Message-ID: Hello, I just tested it on an old Debian 8 server we had with them and it is working, but not with this new server. This is literally the config I have on the old server and from the old server's cli I can type cidlookup and it returns the correct number but from the new server, it returns None. Entire file, everything else has been deleted from the file on the old server and it is working. &auth_token=”/> On Thu, Feb 3, 2022 at 12:10 PM Joli Martinez wrote: > That is the document they sent me. It might be outdated. ill go back and > check with them. > > Thanks > > > On Thu, Feb 3, 2022 at 11:57 AM Darren wrote: > >> OpenCNAM was acquired by Neustar and I thought they shutdown the old API >> server/URL, are you sure you’re hitting the right URL? >> >> >> >> *From:* FreeSWITCH-users *On >> Behalf Of *Joli Martinez >> *Sent:* Thursday, February 3, 2022 7:58 AM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] cidlookup not working >> >> >> >> Hello, >> >> >> >> I am able to type cidlookup on the fs_cli and it returns >> None. The module is loaded, but at no point does it reach out to query the >> OpenCnam servers. I am doing a tcpdump to their IP's and nothing attempts >> goes out to them. >> >> >> >> On Wed, Feb 2, 2022 at 7:43 AM Dragos Oancea >> wrote: >> >> Do you have the "cidlookup" command available in fs_cli ? >> >> >> >> Are you using the "cidlookup" app ? >> >> >> >> On Tue, Feb 1, 2022 at 5:32 PM Joli Martinez wrote: >> >> Hello, >> >> >> >> I have enabled the module cidlookup and added my config to the >> cidlookup.conf.xml file. I have also reloaded the module as well as >> restarted freeswitch. Freeswitch does not attempt to send any queries out >> to opencnam. I am doing a tcpdump on their IP's from the box and it is not >> even attempting to go out to them on port 443 or 53. I am running FS >> 1.10.7 on Debian 11. Any suggestions? >> >> >> >> >> >> >> > phone/+${caller_id_number}?account_sid=&auth_token=”/> >> >> >> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com >> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com >> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From d at d-man.org Thu Feb 3 17:17:23 2022 From: d at d-man.org (Darren) Date: Thu, 3 Feb 2022 17:17:23 +0000 Subject: [Freeswitch-users] cidlookup not working In-Reply-To: References: Message-ID: <939F8ECA-A178-45C0-BB14-D240153331EB@d-man.org> Would really need to see the logs, but based on your comment of it working on one OS but not another I wonder if you have an SSL certificate problem where https://api.opencnam.com/ is actually failing to connect or resolve in one OS versus the other. Kind of guessing without logs. From: FreeSWITCH-users on behalf of Joli Martinez Reply-To: FreeSWITCH Users Help Date: Thursday, February 3, 2022 at 9:16 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] cidlookup not working Hello, I just tested it on an old Debian 8 server we had with them and it is working, but not with this new server. This is literally the config I have on the old server and from the old server's cli I can type cidlookup and it returns the correct number but from the new server, it returns None. Entire file, everything else has been deleted from the file on the old server and it is working. &auth_token=”/> On Thu, Feb 3, 2022 at 12:10 PM Joli Martinez > wrote: That is the document they sent me. It might be outdated. ill go back and check with them. Thanks On Thu, Feb 3, 2022 at 11:57 AM Darren > wrote: OpenCNAM was acquired by Neustar and I thought they shutdown the old API server/URL, are you sure you’re hitting the right URL? From: FreeSWITCH-users > On Behalf Of Joli Martinez Sent: Thursday, February 3, 2022 7:58 AM To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] cidlookup not working Hello, I am able to type cidlookup on the fs_cli and it returns None. The module is loaded, but at no point does it reach out to query the OpenCnam servers. I am doing a tcpdump to their IP's and nothing attempts goes out to them. On Wed, Feb 2, 2022 at 7:43 AM Dragos Oancea > wrote: Do you have the "cidlookup" command available in fs_cli ? Are you using the "cidlookup" app ? On Tue, Feb 1, 2022 at 5:32 PM Joli Martinez > wrote: Hello, I have enabled the module cidlookup and added my config to the cidlookup.conf.xml file. I have also reloaded the module as well as restarted freeswitch. Freeswitch does not attempt to send any queries out to opencnam. I am doing a tcpdump on their IP's from the box and it is not even attempting to go out to them on port 443 or 53. I am running FS 1.10.7 on Debian 11. Any suggestions? &auth_token=”/> _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From botelist at gmail.com Thu Feb 3 17:40:21 2022 From: botelist at gmail.com (Bote Man) Date: Thu, 3 Feb 2022 12:40:21 -0500 Subject: [Freeswitch-users] No hang up, transfers or hold In-Reply-To: <8b5d6e36-edf3-38ab-c999-8c271ff65cd6@szmidt.org> References: <8b5d6e36-edf3-38ab-c999-8c271ff65cd6@szmidt.org> Message-ID: <005a01d81925$1e49b160$5add1420$@gmail.com> All I can tell you is that FreeSWITCH has no way to know how to process your Transfer button on your phone. The defaults that come with the Vanilla configuration files are merely to get it as close as possible to working in a basic PBX mode so that you can play with FreeSWITCH quickly, but they are by no means the end of your configuration tasks. You are fortunate that it has worked so well for you with so little configuration needed thus far. This all sounds like a chance to learn more about how your phones handle transfers, tweak some FreeSWITCH dialplan codes, and perhaps you might need to tweak your firewall to allow the conference audio to pass to external users. I think you are not far off since it mostly works right now. The Wiki is in need of some updating on various topics, but it holds a wealth of information nevertheless. Here is one starting point that might help you: https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+transfer Best success! John Boteler Bote Communications -----Original Message----- From: FreeSWITCH-users On Behalf Of fs Sent: Thursday, 3 February, 2022 10:19 To: FreeSWITCH Users Help Subject: [Freeswitch-users] No hang up, transfers or hold My thought is that others may have run into this and have some pointers to share. I have Freeswitch 4.2.5 Branch 4.2 Switch 1.6.20 64 bit on Linux FusionPBX 3.16.0-11-amd64 #1 SMP Debian 3.16.84-1 (2020-06-09) x86_64 GNU/Linux. I'm using default settings as far as I know/recall. Obviously something must not be as it should be. My phones are a Polycom 650 and a Yealink T48G. My external test phone is a cell phone. It had only been used with a single extension for years until now when I added a second phone and extension. Before today I had noticed that hold was not working properly but did not use it. Attempting to transfer external calls between the two fails in the middle of the transfer. I press the transfer button and type the extension and then hit send but the call does not arrive and the call is hung up on the internal side but the external party does not know until something times out. If I put a call on hold it may be lost as well. Brief holds work. If I call out and then try to transfer that call the same thing happens. If I conference the other extension on an inbound call the external party is not getting any audio once the conference is initiated. If I then hangup on one extension the other is not aware that the call is over. The same is true for the external party. I can call the other extension and answer, when I hang up on either one the other does not know it's gone. It does not matter which phone originates or receives. At least audio works well. :) I've placed a FS log file here: https://drive.google.com/file/d/1VdoLqoPhr5mUE1ky1i2SJjXFChp9w-nM/view?usp=sharing The call duration is 2 minutes from 9:30 to 9:32. IP's and phone numbers have been replaced and placed within < > brackets. Internal IP's retains the last 8-bit number. --- fs From fs at szmidt.org Thu Feb 3 18:02:50 2022 From: fs at szmidt.org (fs) Date: Thu, 3 Feb 2022 13:02:50 -0500 Subject: [Freeswitch-users] No hang up, transfers or hold In-Reply-To: <005a01d81925$1e49b160$5add1420$@gmail.com> References: <8b5d6e36-edf3-38ab-c999-8c271ff65cd6@szmidt.org> <005a01d81925$1e49b160$5add1420$@gmail.com> Message-ID: On 2/3/22 12:40 PM, Bote Man wrote: > All I can tell you is that FreeSWITCH has no way to know how to > process your Transfer button on your phone. Sorry but that seems odd. Both are using standard SIP protocol, and FS recognizes both phones and can configure them. > The defaults that come > with the Vanilla configuration files are merely to get it as close as > possible to working in a basic PBX mode so that you can play with > FreeSWITCH quickly, but they are by no means the end of your > configuration tasks. You are fortunate that it has worked so well for > you with so little configuration needed thus far. When I said default values I did not mean that it was not changed but I did not try to make an unusual setup. It has the standard 5060 and 5080 ports as suggested with a minimum of changes. > This all sounds like a chance to learn more about how your phones > handle transfers, tweak some FreeSWITCH dialplan codes, and perhaps > you might need to tweak your firewall to allow the conference audio > to pass to external users. I think you are not far off since it > mostly works right now. Audio works just fine. The problem lies elsewhere. > The Wiki is in need of some updating on various topics, but it holds > a wealth of information nevertheless. Here is one starting point that > might help you: > https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+transfer OK, I've not found anything yet, but thanks. --- fs -------------- next part -------------- A non-text attachment was scrubbed... Name: OpenPGP_signature Type: application/pgp-signature Size: 840 bytes Desc: OpenPGP digital signature URL: From gregor at infomedia.si Fri Feb 4 09:29:42 2022 From: gregor at infomedia.si (Gregor Nanger) Date: Fri, 4 Feb 2022 10:29:42 +0100 Subject: [Freeswitch-users] Leg 480 kills all other legs? In-Reply-To: References: Message-ID: I think we had a similar problem. From then, I prefer to use enterprise originate as it creates thread for each destination. Separate destinations with :_: On Thu, 3 Feb 2022 at 16:59, David Villasmil wrote: > Thoughts? Ideas anyone? > > On Thu, 3 Feb 2022 at 12:36, David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> Hello all, >> >> I’m calling multiple destinations separated by | >> One of the destinations is returning 480 and fs is canceling all other >> legs. I have continue_on_fail to true (I also tried with 480 and “Temp >> Unava…”) >> >> It was my understanding fs would just ignore that leg (no media is >> received) and allow the other legs continue… >> >> What am I missing? >> >> Thanks all! >> >> David >> -- >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 >> > -- > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From mircea.huh at gmail.com Fri Feb 4 11:49:38 2022 From: mircea.huh at gmail.com (Mircea Botoca-Huh) Date: Fri, 4 Feb 2022 13:49:38 +0200 Subject: [Freeswitch-users] Multi-tenant not working Message-ID: An HTML attachment was scrubbed... URL: From len at freeswitch.org Fri Feb 4 14:53:54 2022 From: len at freeswitch.org (Len Graham) Date: Fri, 4 Feb 2022 09:53:54 -0500 Subject: [Freeswitch-users] cidlookup not working In-Reply-To: <939F8ECA-A178-45C0-BB14-D240153331EB@d-man.org> References: <939F8ECA-A178-45C0-BB14-D240153331EB@d-man.org> Message-ID: What do you get when you curl from the OS's cli? curl https://api.opencnam.com/v3/ phone/ +18883158356?account_sid=&auth_token= On Thu, Feb 3, 2022 at 12:17 PM Darren wrote: > Would really need to see the logs, but based on your comment of it working > on one OS but not another I wonder if you have an SSL certificate problem > where https://api.opencnam.com/ is actually failing to connect or resolve > in one OS versus the other. Kind of guessing without logs. > > > > *From: *FreeSWITCH-users > on behalf of Joli Martinez > *Reply-To: *FreeSWITCH Users Help > *Date: *Thursday, February 3, 2022 at 9:16 AM > *To: *FreeSWITCH Users Help > *Subject: *Re: [Freeswitch-users] cidlookup not working > > > > Hello, > > > > I just tested it on an old Debian 8 server we had with them and it is > working, but not with this new server. This is literally the config I have > on the old server and from the old server's cli I can type cidlookup > and it returns the correct number but from the new server, it > returns None. > > > > Entire file, everything else has been deleted from the file on the old > server and it is working. > > > > > > phone/+${caller_id_number}?account_sid=&auth_token=”/> > > > > > > On Thu, Feb 3, 2022 at 12:10 PM Joli Martinez wrote: > > That is the document they sent me. It might be outdated. ill go back and > check with them. > > > > Thanks > > > > > > On Thu, Feb 3, 2022 at 11:57 AM Darren wrote: > > OpenCNAM was acquired by Neustar and I thought they shutdown the old API > server/URL, are you sure you’re hitting the right URL? > > > > *From:* FreeSWITCH-users *On > Behalf Of *Joli Martinez > *Sent:* Thursday, February 3, 2022 7:58 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] cidlookup not working > > > > Hello, > > > > I am able to type cidlookup on the fs_cli and it returns > None. The module is loaded, but at no point does it reach out to query the > OpenCnam servers. I am doing a tcpdump to their IP's and nothing attempts > goes out to them. > > > > On Wed, Feb 2, 2022 at 7:43 AM Dragos Oancea > wrote: > > Do you have the "cidlookup" command available in fs_cli ? > > > > Are you using the "cidlookup" app ? > > > > On Tue, Feb 1, 2022 at 5:32 PM Joli Martinez wrote: > > Hello, > > > > I have enabled the module cidlookup and added my config to the > cidlookup.conf.xml file. I have also reloaded the module as well as > restarted freeswitch. Freeswitch does not attempt to send any queries out > to opencnam. I am doing a tcpdump on their IP's from the box and it is not > even attempting to go out to them on port 443 or 53. I am running FS > 1.10.7 on Debian 11. Any suggestions? > > > > > > > phone/+${caller_id_number}?account_sid=&auth_token=”/> > > > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Fri Feb 4 15:21:57 2022 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 4 Feb 2022 15:21:57 +0000 Subject: [Freeswitch-users] Leg 480 kills all other legs? In-Reply-To: References: Message-ID: Hello Gregor, Mmm, interesting point. I solved the problem setting continue_on_fail On every leg (in [] instead of the global {}) and setting specifically 480 on the offending leg. On Fri, 4 Feb 2022 at 09:30, Gregor Nanger wrote: > I think we had a similar problem. From then, I prefer to use enterprise > originate as it creates thread for each destination. Separate destinations > with :_: > > On Thu, 3 Feb 2022 at 16:59, David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> Thoughts? Ideas anyone? >> >> On Thu, 3 Feb 2022 at 12:36, David Villasmil < >> david.villasmil.work at gmail.com> wrote: >> >>> Hello all, >>> >>> I’m calling multiple destinations separated by | >>> One of the destinations is returning 480 and fs is canceling all other >>> legs. I have continue_on_fail to true (I also tried with 480 and “Temp >>> Unava…”) >>> >>> It was my understanding fs would just ignore that leg (no media is >>> received) and allow the other legs continue… >>> >>> What am I missing? >>> >>> Thanks all! >>> >>> David >>> -- >>> Regards, >>> >>> David Villasmil >>> email: david.villasmil.work at gmail.com >>> phone: +34669448337 >>> >> -- >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 >> > _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From sdevoy at bizfocused.com Fri Feb 4 19:32:38 2022 From: sdevoy at bizfocused.com (Sean Devoy) Date: Fri, 4 Feb 2022 19:32:38 +0000 Subject: [Freeswitch-users] Outbound Gateway hang up at 30 seconds with no answer In-Reply-To: <202201072255.53485.Antony.Stone@freeswitch.open.source.it> References: <202201072255.53485.Antony.Stone@freeswitch.open.source.it> Message-ID: I have a PCAP with a failed call. It is 100K. where can I put it or send it? Should I attach it here? I don't understand what I am looking at. Hopefully, someone can look at it and help me out. Thanks, Sean -----Original Message----- From: FreeSWITCH-users On Behalf Of Antony Stone Sent: Friday, January 7, 2022 4:56 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Outbound Gateway hang up at 30 seconds with no answer On Friday 07 January 2022 at 22:33:31, David Villasmil wrote: > yeah, if you're using tshark, something like: > > tshark -f "host [GATEWAY-IP]" -i any > > will show you anything going/coming from that ip address. > > you can also install sngrep, which is a fantastic cli tool. it's in > debian's repos. You start it with "sngrep -c" to show only [c]alls I would also advocate doing both. 1. Install sngrep, and if you don't already have them, tshark and/or tcpdump 2. Use tshark (or tcpdump) to capture SIP traffic (which normally means "anything on port 5060, in either direction"), so: tshark -f "port 5060" -i any -w capturefile.pcap 3. Then you can use sngrep to read the file which tshark created, and at your leisure, scroll through the traffic and try to understand what happened when: sngrep -I capturefile.pcap sngrep is a splendid tool, but SIP is not a simple protocol, so having a capture file you can go back to later, whenever you want to, and understand a bit more about it (or answer other people's questions if they ask "well, what was the response to this, then?") is very useful. Antony. -- Some things the German language doesn't easily distinguish between: - slugs and snails - cucumbers and gherkins - snakes and queues - wearing something, or carrying it - mothers and nuts - driving a car, riding a bicycle, or travelling by train - a man and a husband - a woman and a wife - changing clothes and moving house - pockets and bags Please reply to the list; please *don't* CC me. _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com From david.villasmil.work at gmail.com Fri Feb 4 20:39:28 2022 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 4 Feb 2022 20:39:28 +0000 Subject: [Freeswitch-users] ice_lite dynamically? Message-ID: Hello all, I'm trying to enable/disable ice_lite dynamically by setting it on the xml dialplan, but nothing's happening. If i set it globally on vars.xml it does work, but i can't modify it on the dialplan... is this not possible? Thanks and Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Fri Feb 4 22:13:01 2022 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 4 Feb 2022 22:13:01 +0000 Subject: [Freeswitch-users] ice_lite dynamically? In-Reply-To: References: Message-ID: I was missing the inline On Fri, 4 Feb 2022 at 20:39, David Villasmil wrote: > Hello all, > > I'm trying to enable/disable ice_lite dynamically by setting it on the xml > dialplan, but nothing's happening. > > If i set it globally on vars.xml it does work, but i can't modify it on > the dialplan... > > is this not possible? > > Thanks and Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Fri Feb 4 23:46:05 2022 From: krice at freeswitch.org (Ken Rice) Date: Fri, 4 Feb 2022 17:46:05 -0600 Subject: [Freeswitch-users] Outbound Gateway hang up at 30 seconds with no answer In-Reply-To: References: Message-ID: <442469A3-F282-4FB2-B8CD-0F1453479342@freeswitch.org> NAT or similar related issue. let me guess called party is sending a bye to you 30 seconds after the they send the 200 OK? they are most likely not getting the ACK for the 200 OK. look at your trance and see where the ACK is getting sent to Sent from my iPhone > On Feb 4, 2022, at 13:33, Sean Devoy wrote: > > I have a PCAP with a failed call. It is 100K. where can I put it or send it? Should I attach it here? > I don't understand what I am looking at. Hopefully, someone can look at it and help me out. > > Thanks, > Sean > > -----Original Message----- > From: FreeSWITCH-users On Behalf Of Antony Stone > Sent: Friday, January 7, 2022 4:56 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Outbound Gateway hang up at 30 seconds with no answer > >> On Friday 07 January 2022 at 22:33:31, David Villasmil wrote: >> >> yeah, if you're using tshark, something like: >> >> tshark -f "host [GATEWAY-IP]" -i any >> >> will show you anything going/coming from that ip address. >> >> you can also install sngrep, which is a fantastic cli tool. it's in >> debian's repos. You start it with "sngrep -c" to show only [c]alls > > I would also advocate doing both. > > 1. Install sngrep, and if you don't already have them, tshark and/or tcpdump > > 2. Use tshark (or tcpdump) to capture SIP traffic (which normally means "anything on port 5060, in either direction"), so: > > tshark -f "port 5060" -i any -w capturefile.pcap > > 3. Then you can use sngrep to read the file which tshark created, and at your leisure, scroll through the traffic and try to understand what happened when: > > sngrep -I capturefile.pcap > > sngrep is a splendid tool, but SIP is not a simple protocol, so having a capture file you can go back to later, whenever you want to, and understand a bit more about it (or answer other people's questions if they ask "well, what was the response to this, then?") is very useful. > > > Antony. > > -- > Some things the German language doesn't easily distinguish between: > - slugs and snails > - cucumbers and gherkins > - snakes and queues > - wearing something, or carrying it > - mothers and nuts > - driving a car, riding a bicycle, or travelling by train > - a man and a husband > - a woman and a wife > - changing clothes and moving house > - pockets and bags > > Please reply to the list; > please *don't* CC me. > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com From david.villasmil.work at gmail.com Fri Feb 4 23:56:30 2022 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 4 Feb 2022 23:56:30 +0000 Subject: [Freeswitch-users] ICE_LITE FS-to-FS issues Message-ID: Hello folks, I enabled ice_lite on FS and when the caller and callee are on the same FS, all is good. But if they are on different FS there’s no audio, although signaling “looks” good. The path is: A->FS->FS->B Anyone else seeing this? Thanks all! David -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From csadi at hotmail.com Mon Feb 7 13:55:30 2022 From: csadi at hotmail.com (Adiseshu Channasamudhram) Date: Mon, 7 Feb 2022 13:55:30 +0000 Subject: [Freeswitch-users] =?utf-8?q?=28no_subject=29?= Message-ID: Hello Freeswitch Team Can one of you please let me know how i can copy the P-Asserted-Identity header from the incoming INVITE to the outgoing INVITE? I did go through most of the articles but none of them helped. I have tried the below so far in my dialplan Thank you in advance Regards, Adi -------------- next part -------------- An HTML attachment was scrubbed... URL: From shaun at sysconfig.cloud Mon Feb 7 15:13:51 2022 From: shaun at sysconfig.cloud (Shaun Stokes) Date: Mon, 7 Feb 2022 15:13:51 +0000 Subject: [Freeswitch-users] (no subject) In-Reply-To: References: Message-ID: If you're setting the PAID manually using 'sip_h' then CID Type should be set to none, or you'll likely get the two PAID headers. ________________________________ From: FreeSWITCH-users on behalf of Adiseshu Channasamudhram Sent: 07 February 2022 14:55 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] (no subject) Hello Freeswitch Team Can one of you please let me know how i can copy the P-Asserted-Identity header from the incoming INVITE to the outgoing INVITE? I did go through most of the articles but none of them helped. I have tried the below so far in my dialplan Thank you in advance Regards, Adi -------------- next part -------------- An HTML attachment was scrubbed... URL: From Antony.Stone at freeswitch.open.source.it Mon Feb 7 15:18:04 2022 From: Antony.Stone at freeswitch.open.source.it (Antony Stone) Date: Mon, 7 Feb 2022 16:18:04 +0100 Subject: [Freeswitch-users] (no subject) In-Reply-To: References: Message-ID: <202202071618.04809.Antony.Stone@freeswitch.open.source.it> On Monday 07 February 2022 at 14:55:30, Adiseshu Channasamudhram wrote: > Can one of you please let me know how i can copy the P-Asserted-Identity > header from the incoming INVITE to the outgoing INVITE? > data="sip_h_P-Asserted-Identity=${sip_P-Asserted_Identity}"/> Do you not simply have a - vs. _ typo there? Antony. -- Most people have more than the average number of legs. Please reply to the list; please *don't* CC me. From Zvonimir.Buzanic at asseco-see.hr Mon Feb 7 15:29:19 2022 From: Zvonimir.Buzanic at asseco-see.hr (=?iso-8859-2?Q?Zvonimir_Bu=BEani=E6?=) Date: Mon, 7 Feb 2022 15:29:19 +0000 Subject: [Freeswitch-users] (no subject) In-Reply-To: References: Message-ID: <4c912ea0d599497c9b67c9a5c3868565@asseco-see.hr> We are using something like this: Br, Zvonimir From: FreeSWITCH-users On Behalf Of Adiseshu Channasamudhram Sent: 07. veljače 2022. 14:56 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] (no subject) Hello Freeswitch Team Can one of you please let me know how i can copy the P-Asserted-Identity header from the incoming INVITE to the outgoing INVITE? I did go through most of the articles but none of them helped. I have tried the below so far in my dialplan Thank you in advance Regards, Adi This communication is for informational purposes only. All market prices, data and other information are not warranted as to completeness or accuracy and are subject to change without notice. Present message and any attached files may be or contain privileged information and is the property exclusive of ASSECO SEE CAPITAL GROUP. This transmission may contain information that is privileged, confidential, legally privileged, and/or exempt from disclosure under applicable law. The information contained in this message is solely intended for the physical or legal person to whom it is addressed and to the authorized persons for receiving it. In the case you are not the intended recipient or the authorized person to receive this message, we inform that disclosure, duplicate, distribution or taking up any actions on information contained in this message are strictly forbidden and are under civil and legal responsibility. In case you received it by error, you are requested to notify the sender and to destroy the original e-mail message from your system. Opinions, conclusions or any other information contained into this message, which are not related to ASSECO SEE CAPITAL GROUP activity must not be understood to be expressed or should be endorsed by ASSECO SEE CAPITAL GROUP. The interpretation expressed in the present message did not reflect ASSECO SEE CAPITAL GROUP opinion. -------------- next part -------------- An HTML attachment was scrubbed... URL: From heedfeld at gmail.com Mon Feb 7 16:46:51 2022 From: heedfeld at gmail.com (Henning Heedfeld) Date: Mon, 7 Feb 2022 17:46:51 +0100 Subject: [Freeswitch-users] (no subject) In-Reply-To: References: Message-ID: <164C7962-FCBF-4A71-B982-A624F8F03735@gmail.com> Hi, that’s the way I do it: | INVITE | | | no RPID (correct) | | | ------------------------------->| | | | | | 200 OK | | | contains RPID | | 200 OK |<--------------------------------| | contains RPID, copied from B-leg | | | I want this RPID removed | | |<-----------------------------------| | Cheers, Marcin -------------- next part -------------- An HTML attachment was scrubbed... URL: From dragos at freeswitch.org Wed Feb 16 12:49:52 2022 From: dragos at freeswitch.org (Dragos Oancea) Date: Wed, 16 Feb 2022 14:49:52 +0200 Subject: [Freeswitch-users] Media timeout In-Reply-To: References: Message-ID: please open a github issue. https://github.com/signalwire/freeswitch/issues you could also check if switch_rtp_set_max_missed_packets() and/or switch_rtp_set_media_timeout() are called in your particular call setup, with your configs. On Tue, Feb 15, 2022 at 11:33 PM Alexander Haugg wrote: > Hi Shaun, > > > > I have noticed the same thing. "media_timout" in conjunction with WebRTC > does not work at all. > > If someone has a solution, that would be really fine. > > > > Currently I have helped myself by re-enabling the following in the code: > > > > diff --git a/src/switch_rtp.c b/src/switch_rtp.c > > index 40d8978..aada64a 100644 > > --- a/src/switch_rtp.c > > +++ b/src/switch_rtp.c > > @@ -854,8 +854,8 @@ static switch_status_t ice_out(switch_rtp_t > *rtp_session, switch_rtp_ice_t *ice) > > if (elapsed > 30000) { > > > switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), > SWITCH_LOG_WARNING, "No %s rtp_session->last_stun > = switch_micro_time_now(); > > - //status = SWITCH_STATUS_GENERR; > > - //goto end; > > + status = SWITCH_STATUS_GENERR; > > + goto end; > > } > > } > > > > @@ -897,7 +897,7 @@ static switch_status_t ice_out(switch_rtp_t > *rtp_session, switch_rtp_ice_t *ice) > > > > ice->sending = 3; > > > > - // end: > > + end: > > READ_DEC(rtp_session); > > > > return status; > > (END) > > > > The code has the consequence that the session is cleared if no more media > comes for 30 seconds. > > > > With kind regards > > Alex > > > > *Von:* FreeSWITCH-users *Im > Auftrag von *Shaun Stokes > *Gesendet:* Donnerstag, 10. Februar 2022 14:39 > *An:* FreeSWITCH Users Help > *Betreff:* [Freeswitch-users] Media timeout > > > > Hi All, > > > > I'm not sure if others have had similar experiences but for us the > media_timeout variable does not work as expected. > > - media_timeout on a call that supports video but with-out video will > fail. > - media_timeout_audio works in some instances, in others the timeout > period is ignored so the call will timeout almost immediately after RTP > stops. > - media_hold_timeout_audio doesn't seem to work at all, calls that are > on hold never timeout. > > Why is the SIP profile parameter 'rtp-timeout-sec' depreciated? It's much > simpler to apply this per SIP profile than it is per call. > > > > Thanks, > > Shaun > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From botelist at gmail.com Wed Feb 16 18:21:00 2022 From: botelist at gmail.com (Bote Man) Date: Wed, 16 Feb 2022 13:21:00 -0500 Subject: [Freeswitch-users] mod_sms: use different context In-Reply-To: <09D27263-D03D-42FB-84E7-44178ABFA463@wirelessmundi.com> References: <09D27263-D03D-42FB-84E7-44178ABFA463@wirelessmundi.com> Message-ID: <005b01d82361$f3464c20$d9d2e460$@gmail.com> I guess that the “transfer” dialplan app is only for voice calls, but I do not know what apps or commands work for SMS/chat? I have never understood how the chatplan works, so I am following this to learn. I want to send SMS out of FreeSWITCH somehow in the near future. John Boteler Bote Communications From: António Silva Sent: Tuesday, 15 February, 2022 14:17 To: freeswitch-users at lists.freeswitch.org Subject: mod_sms: use different context Hi, Is it possible to set a different context in chatplan.xml? Or when receiving a chat in the default context send it to a different context like transfer app? I can set the context in Sofia profile, but is not possible to set it per user… The following didn't work: Got the following error: Chatplan: 6 parsing [default->tapi] continue=false Chatplan: 6 at lab2.local Regex (FAIL) [tapi] to(6 at lab2.local ) =~ /^tapi.*$/ break=on-false Chatplan: 6 parsing [default->tel message sensor] continue=false Chatplan: 6 at lab2.local Regex (FAIL) [tel message sensor] to(6 at lab2.local ) =~ /^tel_message_sensor/ break=on-false Chatplan: 6 parsing [default->default_catch_all] continue=false Chatplan: 6 at lab2.local Regex (PASS) [default_catch_all] to(6 at lab2.local ) =~ /^(.*)$/ break=on-false Chatplan: 6 at lab2.local Action transfer(${to} XML outgoing) 2022-02-15 09:45:14.759408 83.57% [ERR] switch_loadable_module.c:1000 Invalid chat application interface [transfer]! 2022-02-15 09:45:14.759408 83.57% [DEBUG] sofia_presence.c:225 Can't find registered user 6 at lab2.local -- Saludos / Regards / Cumprimentos António Silva -------------- next part -------------- An HTML attachment was scrubbed... URL: From mahmood.alkhalil at outlook.com Thu Feb 17 05:12:55 2022 From: mahmood.alkhalil at outlook.com (Mahmood Alkhalil) Date: Thu, 17 Feb 2022 09:12:55 +0400 Subject: [Freeswitch-users] Single RTP port to handle all RTP streams Message-ID: Hi all, Is it possible to configure FS to use a single port for all RTP streams? I face tons of issue with our ISP blocking most of the ports you might think of, so is it possible to do such with FS or any other software i can use in front of freeswitch? From botelist at gmail.com Thu Feb 17 07:04:12 2022 From: botelist at gmail.com (Bote Man) Date: Thu, 17 Feb 2022 02:04:12 -0500 Subject: [Freeswitch-users] Single RTP port to handle all RTP streams In-Reply-To: References: Message-ID: <00f001d823cc$91e7d5a0$b5b780e0$@gmail.com> Each FreeSWITCH SIP profile represents one port for SIP signaling. However, each voice leg or session will use its own unique port. That is the only way for FreeSWITCH to identify each RTP stream. John Boteler Bote Communications -----Original Message----- From: FreeSWITCH-users On Behalf Of Mahmood Alkhalil Sent: Thursday, 17 February, 2022 00:13 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Single RTP port to handle all RTP streams Hi all, Is it possible to configure FS to use a single port for all RTP streams? I face tons of issue with our ISP blocking most of the ports you might think of, so is it possible to do such with FS or any other software i can use in front of freeswitch? _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com From gilles at sauvaire.com Thu Feb 17 07:36:24 2022 From: gilles at sauvaire.com (Gilles SAUVAIRE) Date: Thu, 17 Feb 2022 08:36:24 +0100 Subject: [Freeswitch-users] Single RTP port to handle all RTP streams In-Reply-To: <00f001d823cc$91e7d5a0$b5b780e0$@gmail.com> References: <00f001d823cc$91e7d5a0$b5b780e0$@gmail.com> Message-ID: yes, You can use only one port... but... you will only have one simultaneous communication. -----Message d'origine----- De : FreeSWITCH-users De la part de Bote Man Envoyé : jeudi 17 février 2022 08:04 À : 'FreeSWITCH Users Help' Objet : Re: [Freeswitch-users] Single RTP port to handle all RTP streams Each FreeSWITCH SIP profile represents one port for SIP signaling. However, each voice leg or session will use its own unique port. That is the only way for FreeSWITCH to identify each RTP stream. John Boteler Bote Communications -----Original Message----- From: FreeSWITCH-users On Behalf Of Mahmood Alkhalil Sent: Thursday, 17 February, 2022 00:13 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Single RTP port to handle all RTP streams Hi all, Is it possible to configure FS to use a single port for all RTP streams? I face tons of issue with our ISP blocking most of the ports you might think of, so is it possible to do such with FS or any other software i can use in front of freeswitch? _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com From david.villasmil.work at gmail.com Thu Feb 17 15:39:51 2022 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 17 Feb 2022 15:39:51 +0000 Subject: [Freeswitch-users] Single RTP port to handle all RTP streams In-Reply-To: References: Message-ID: use a vpn Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Thu, Feb 17, 2022 at 5:45 AM Mahmood Alkhalil < mahmood.alkhalil at outlook.com> wrote: > Hi all, > > > Is it possible to configure FS to use a single port for all RTP streams? > > I face tons of issue with our ISP blocking most of the ports you might > think of, so is it possible to do such with FS or any other software i > can use in front of freeswitch? > > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From lists at telium.io Thu Feb 17 16:43:48 2022 From: lists at telium.io (TTT) Date: Thu, 17 Feb 2022 16:43:48 +0000 Subject: [Freeswitch-users] Single RTP port to handle all RTP streams In-Reply-To: References: Message-ID: <0100017f089182d1-5e0f1b2f-0629-42cd-93a3-fc7cf638a5fd-000000@email.amazonses.com> Do you control boths end of the SIP/RTP connection? Something as simple as an IPIP / GRE tunnel may be a simple solution without the overhead of encryption, negotiations, authentication, etc. Just ads a bit to each header. If you don’t control the far end of the connection then you are out of luck, unless they support tunnels (or VPN as suggested below) Jason From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Villasmil Sent: Thursday, February 17, 2022 10:40 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Single RTP port to handle all RTP streams use a vpn Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Thu, Feb 17, 2022 at 5:45 AM Mahmood Alkhalil > wrote: Hi all, Is it possible to configure FS to use a single port for all RTP streams? I face tons of issue with our ISP blocking most of the ports you might think of, so is it possible to do such with FS or any other software i can use in front of freeswitch? _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From csadi at hotmail.com Thu Feb 17 17:05:18 2022 From: csadi at hotmail.com (Adiseshu Channasamudhram) Date: Thu, 17 Feb 2022 17:05:18 +0000 Subject: [Freeswitch-users] How to copy from header Message-ID: Hello Team I have a requirement to copy from header from the incoming INVITE to the outgoing INVITE right now, below is the from header from incomign invite From: "123-456-7788" ;tag=2bf345ee but the outgoing INVITE has the from header as below From: "123-456-7788" ;tag=yaXFUFjD666ym How can i copy the cpc and isup-oli from the incoming from header to the outgoing header? Thanks in advance Regards Adi -------------- next part -------------- An HTML attachment was scrubbed... URL: From mahmood.alkhalil at outlook.com Fri Feb 18 08:42:56 2022 From: mahmood.alkhalil at outlook.com (Mahmood Alkhalil) Date: Fri, 18 Feb 2022 08:42:56 +0000 Subject: [Freeswitch-users] Single RTP port to handle all RTP streams In-Reply-To: <0100017f089182d1-5e0f1b2f-0629-42cd-93a3-fc7cf638a5fd-000000@email.amazonses.com> References: <0100017f089182d1-5e0f1b2f-0629-42cd-93a3-fc7cf638a5fd-000000@email.amazonses.com> Message-ID: I am using webrtc and SIP over WSS, and I don't want to add any extra requirements such as VPN or tunneling as users will be using different machines at any moment, so i guess i will need to check with the ISP why UDP ports are blocked. Mahmood Alkhalil. +971552235220 ________________________________ From: FreeSWITCH-users on behalf of TTT Sent: Thursday, February 17, 2022 8:43:48 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] Single RTP port to handle all RTP streams Do you control boths end of the SIP/RTP connection? Something as simple as an IPIP / GRE tunnel may be a simple solution without the overhead of encryption, negotiations, authentication, etc. Just ads a bit to each header. If you don’t control the far end of the connection then you are out of luck, unless they support tunnels (or VPN as suggested below) Jason From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Villasmil Sent: Thursday, February 17, 2022 10:40 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Single RTP port to handle all RTP streams use a vpn Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Thu, Feb 17, 2022 at 5:45 AM Mahmood Alkhalil > wrote: Hi all, Is it possible to configure FS to use a single port for all RTP streams? I face tons of issue with our ISP blocking most of the ports you might think of, so is it possible to do such with FS or any other software i can use in front of freeswitch? _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From shaun at sysconfig.cloud Fri Feb 18 09:00:08 2022 From: shaun at sysconfig.cloud (Shaun Stokes) Date: Fri, 18 Feb 2022 09:00:08 +0000 Subject: [Freeswitch-users] Media timeout In-Reply-To: References: Message-ID: I've figured out our issue with media_timeout_audio, this is using ms not seconds. Is this a bug, or is the documentation out of date? ________________________________ From: FreeSWITCH-users on behalf of Dragos Oancea Sent: 16 February 2022 13:49 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Media timeout please open a github issue. https://github.com/signalwire/freeswitch/issues you could also check if switch_rtp_set_max_missed_packets() and/or switch_rtp_set_media_timeout() are called in your particular call setup, with your configs. On Tue, Feb 15, 2022 at 11:33 PM Alexander Haugg > wrote: Hi Shaun, I have noticed the same thing. "media_timout" in conjunction with WebRTC does not work at all. If someone has a solution, that would be really fine. Currently I have helped myself by re-enabling the following in the code: diff --git a/src/switch_rtp.c b/src/switch_rtp.c index 40d8978..aada64a 100644 --- a/src/switch_rtp.c +++ b/src/switch_rtp.c @@ -854,8 +854,8 @@ static switch_status_t ice_out(switch_rtp_t *rtp_session, switch_rtp_ice_t *ice) if (elapsed > 30000) { switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_WARNING, "No %s rtp_session->last_stun = switch_micro_time_now(); - //status = SWITCH_STATUS_GENERR; - //goto end; + status = SWITCH_STATUS_GENERR; + goto end; } } @@ -897,7 +897,7 @@ static switch_status_t ice_out(switch_rtp_t *rtp_session, switch_rtp_ice_t *ice) ice->sending = 3; - // end: + end: READ_DEC(rtp_session); return status; (END) The code has the consequence that the session is cleared if no more media comes for 30 seconds. With kind regards Alex Von: FreeSWITCH-users > Im Auftrag von Shaun Stokes Gesendet: Donnerstag, 10. Februar 2022 14:39 An: FreeSWITCH Users Help > Betreff: [Freeswitch-users] Media timeout Hi All, I'm not sure if others have had similar experiences but for us the media_timeout variable does not work as expected. * media_timeout on a call that supports video but with-out video will fail. * media_timeout_audio works in some instances, in others the timeout period is ignored so the call will timeout almost immediately after RTP stops. * media_hold_timeout_audio doesn't seem to work at all, calls that are on hold never timeout. Why is the SIP profile parameter 'rtp-timeout-sec' depreciated? It's much simpler to apply this per SIP profile than it is per call. Thanks, Shaun _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From dragos at freeswitch.org Fri Feb 18 10:48:00 2022 From: dragos at freeswitch.org (Dragos Oancea) Date: Fri, 18 Feb 2022 12:48:00 +0200 Subject: [Freeswitch-users] Media timeout In-Reply-To: References: Message-ID: This has been discussed on this ML a while ago. Documentation out of date. On Fri, Feb 18, 2022 at 11:00 AM Shaun Stokes wrote: > I've figured out our issue with media_timeout_audio, this is using ms not > seconds. > > Is this a bug, or is the documentation out of date? > ------------------------------ > *From:* FreeSWITCH-users > on behalf of Dragos Oancea > *Sent:* 16 February 2022 13:49 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Media timeout > > please open a github issue. > https://github.com/signalwire/freeswitch/issues > > you could also check if switch_rtp_set_max_missed_packets() > and/or switch_rtp_set_media_timeout() are called in your particular call > setup, with your configs. > > > > On Tue, Feb 15, 2022 at 11:33 PM Alexander Haugg > wrote: > > Hi Shaun, > > > > I have noticed the same thing. "media_timout" in conjunction with WebRTC > does not work at all. > > If someone has a solution, that would be really fine. > > > > Currently I have helped myself by re-enabling the following in the code: > > > > diff --git a/src/switch_rtp.c b/src/switch_rtp.c > > index 40d8978..aada64a 100644 > > --- a/src/switch_rtp.c > > +++ b/src/switch_rtp.c > > @@ -854,8 +854,8 @@ static switch_status_t ice_out(switch_rtp_t > *rtp_session, switch_rtp_ice_t *ice) > > if (elapsed > 30000) { > > > switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), > SWITCH_LOG_WARNING, "No %s rtp_session->last_stun > = switch_micro_time_now(); > > - //status = SWITCH_STATUS_GENERR; > > - //goto end; > > + status = SWITCH_STATUS_GENERR; > > + goto end; > > } > > } > > > > @@ -897,7 +897,7 @@ static switch_status_t ice_out(switch_rtp_t > *rtp_session, switch_rtp_ice_t *ice) > > > > ice->sending = 3; > > > > - // end: > > + end: > > READ_DEC(rtp_session); > > > > return status; > > (END) > > > > The code has the consequence that the session is cleared if no more media > comes for 30 seconds. > > > > With kind regards > > Alex > > > > *Von:* FreeSWITCH-users *Im > Auftrag von *Shaun Stokes > *Gesendet:* Donnerstag, 10. Februar 2022 14:39 > *An:* FreeSWITCH Users Help > *Betreff:* [Freeswitch-users] Media timeout > > > > Hi All, > > > > I'm not sure if others have had similar experiences but for us the > media_timeout variable does not work as expected. > > - media_timeout on a call that supports video but with-out video will > fail. > - media_timeout_audio works in some instances, in others the timeout > period is ignored so the call will timeout almost immediately after RTP > stops. > - media_hold_timeout_audio doesn't seem to work at all, calls that are > on hold never timeout. > > Why is the SIP profile parameter 'rtp-timeout-sec' depreciated? It's much > simpler to apply this per SIP profile than it is per call. > > > > Thanks, > > Shaun > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From csadi at hotmail.com Fri Feb 18 12:23:29 2022 From: csadi at hotmail.com (Adiseshu Channasamudhram) Date: Fri, 18 Feb 2022 12:23:29 +0000 Subject: [Freeswitch-users] How to copy from header In-Reply-To: References: Message-ID: Hello FS experts Can someone please help me with this? Thanks Adi Sent from my iPhone On Feb 17, 2022, at 12:05 PM, Adiseshu Channasamudhram wrote:  Hello Team I have a requirement to copy from header from the incoming INVITE to the outgoing INVITE right now, below is the from header from incomign invite From: "123-456-7788" ;tag=2bf345ee but the outgoing INVITE has the from header as below From: "123-456-7788" ;tag=yaXFUFjD666ym How can i copy the cpc and isup-oli from the incoming from header to the outgoing header? Thanks in advance Regards Adi -------------- next part -------------- An HTML attachment was scrubbed... URL: From Jaan.Kaja at enghouse.com Fri Feb 18 09:40:21 2022 From: Jaan.Kaja at enghouse.com (Jaan Kaja) Date: Fri, 18 Feb 2022 09:40:21 +0000 Subject: Sv: [Freeswitch-users] Single RTP port to handle all RTP streams In-Reply-To: References: <0100017f089182d1-5e0f1b2f-0629-42cd-93a3-fc7cf638a5fd-000000@email.amazonses.com> Message-ID: In switch.conf.xml, you can limit the range of ports that are used: Might make it easier to negotiate with your ISP. /Jaan Från: FreeSWITCH-users För Mahmood Alkhalil Skickat: den 18 februari 2022 09:43 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Single RTP port to handle all RTP streams I am using webrtc and SIP over WSS, and I don't want to add any extra requirements such as VPN or tunneling as users will be using different machines at any moment, so i guess i will need to check with the ISP why UDP ports are blocked. Mahmood Alkhalil. +971552235220 ________________________________ From: FreeSWITCH-users > on behalf of TTT > Sent: Thursday, February 17, 2022 8:43:48 PM To: 'FreeSWITCH Users Help' > Subject: Re: [Freeswitch-users] Single RTP port to handle all RTP streams Do you control boths end of the SIP/RTP connection? Something as simple as an IPIP / GRE tunnel may be a simple solution without the overhead of encryption, negotiations, authentication, etc. Just ads a bit to each header. If you don't control the far end of the connection then you are out of luck, unless they support tunnels (or VPN as suggested below) Jason From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Villasmil Sent: Thursday, February 17, 2022 10:40 AM To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Single RTP port to handle all RTP streams use a vpn Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Thu, Feb 17, 2022 at 5:45 AM Mahmood Alkhalil > wrote: Hi all, Is it possible to configure FS to use a single port for all RTP streams? I face tons of issue with our ISP blocking most of the ports you might think of, so is it possible to do such with FS or any other software i can use in front of freeswitch? _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From botelist at gmail.com Fri Feb 18 17:04:03 2022 From: botelist at gmail.com (Bote Man) Date: Fri, 18 Feb 2022 12:04:03 -0500 Subject: [Freeswitch-users] Media timeout In-Reply-To: References: Message-ID: <018901d824e9$88c60b90$9a5222b0$@gmail.com> I have updated the Confluence page https://freeswitch.org/confluence/display/FREESWITCH/Sofia+Configuration+Files but that variable is mentioned on other pages because it’s a wiki so… We always welcome help with updating the wiki, just ask for edit permission. Thank you. John Boteler Bote Communications From: FreeSWITCH-users On Behalf Of Dragos Oancea Sent: Friday, 18 February, 2022 05:48 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Media timeout This has been discussed on this ML a while ago. Documentation out of date. On Fri, Feb 18, 2022 at 11:00 AM Shaun Stokes > wrote: I've figured out our issue with media_timeout_audio, this is using ms not seconds. Is this a bug, or is the documentation out of date? _____ From: FreeSWITCH-users > on behalf of Dragos Oancea > Sent: 16 February 2022 13:49 To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Media timeout please open a github issue. https://github.com/signalwire/freeswitch/issues you could also check if switch_rtp_set_max_missed_packets() and/or switch_rtp_set_media_timeout() are called in your particular call setup, with your configs. On Tue, Feb 15, 2022 at 11:33 PM Alexander Haugg > wrote: Hi Shaun, I have noticed the same thing. "media_timout" in conjunction with WebRTC does not work at all. If someone has a solution, that would be really fine. Currently I have helped myself by re-enabling the following in the code: diff --git a/src/switch_rtp.c b/src/switch_rtp.c index 40d8978..aada64a 100644 --- a/src/switch_rtp.c +++ b/src/switch_rtp.c @@ -854,8 +854,8 @@ static switch_status_t ice_out(switch_rtp_t *rtp_session, switch_rtp_ice_t *ice) if (elapsed > 30000) { switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_WARNING, "No %s rtp_session->last_stun = switch_micro_time_now(); - //status = SWITCH_STATUS_GENERR; - //goto end; + status = SWITCH_STATUS_GENERR; + goto end; } } @@ -897,7 +897,7 @@ static switch_status_t ice_out(switch_rtp_t *rtp_session, switch_rtp_ice_t *ice) ice->sending = 3; - // end: + end: READ_DEC(rtp_session); return status; (END) The code has the consequence that the session is cleared if no more media comes for 30 seconds. With kind regards Alex Von: FreeSWITCH-users > Im Auftrag von Shaun Stokes Gesendet: Donnerstag, 10. Februar 2022 14:39 An: FreeSWITCH Users Help > Betreff: [Freeswitch-users] Media timeout Hi All, I'm not sure if others have had similar experiences but for us the media_timeout variable does not work as expected. * media_timeout on a call that supports video but with-out video will fail. * media_timeout_audio works in some instances, in others the timeout period is ignored so the call will timeout almost immediately after RTP stops. * media_hold_timeout_audio doesn't seem to work at all, calls that are on hold never timeout. Why is the SIP profile parameter 'rtp-timeout-sec' depreciated? It's much simpler to apply this per SIP profile than it is per call. Thanks, Shaun _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From sdevoy at bizfocused.com Fri Feb 18 21:19:35 2022 From: sdevoy at bizfocused.com (Sean Devoy) Date: Fri, 18 Feb 2022 21:19:35 +0000 Subject: Why did the user phone send a BYE. Message-ID: I have reviewed my pcap file MANY times. I have a FAILED call and a GOOD call from the same extension. I will attempt you list the signaling below. One oddity that jumps out at me, there are no SIP 183 packets in the call that WORKED. It is clear that the user is hanging up, but they state the call rings 4 times, then goes to busy. The tshark command I am using is: tshark -f "port 5060" -i any -w capturefile.pcap How can I include RTP messages? Will that let me see if there is early media, then busy or failure? Call that failed signaling: U is the User extension behind NAT. FS is the server with public IP. GW is gateway with public IP (Voip Innovations). U-> FS INVITE FS->U AUTH REQ U->FS Request: ACK U->FS INVITE FS->U Trying FS->GW INVITE GW->FS TRYING U->FS Status 200 GW->FS Status: 183 Session Progress FS->U Status: 183 Session Progress 1/10 Second Later GW->FS Status: 183 Session Progress 21 Seconds Later GW->FS Status: 183 Session Progress GW->FS Status: 200 FS->GW Request: ACK FS->U Status 200 U->FS Request ACK FS->U Request: UPDATE U->FS Status 200 3 Seconds Later - 26 seconds total U->FS BYE FS->U Status: 200 FS->GW BYE GW->FS Status: 200 Regards, Sean Devoy VP Operations and Development Business Focused Internet Systems, Inc. From w8hdkim at gmail.com Sat Feb 19 15:41:00 2022 From: w8hdkim at gmail.com (Kim Culhan) Date: Sat, 19 Feb 2022 10:41:00 -0500 Subject: [Freeswitch-users] voicemail from domain Message-ID: Running FreeSWITCH Version 1.10.3-release~64bit (-release 64bit) Emails sent with voicemail messages have a from line: extension at ip_address_of_external interface. In voicemail.conf.xml have: This did not change it, still has @external_ip_address Any help is greatly appreciated. thanks -kim -------------- next part -------------- An HTML attachment was scrubbed... URL: From aryeklt at gmail.com Sun Feb 20 18:08:48 2022 From: aryeklt at gmail.com (=?UTF-8?B?15DXqNeZ15Qg16fXnNeY16g=?=) Date: Sun, 20 Feb 2022 20:08:48 +0200 Subject: [Freeswitch-users] freeswitch stuck at Push codec L16:100 In-Reply-To: References: Message-ID: just update it happened again one day after the last post, 2022-02-16 but it stucked only on one specific conference, on other conferences it is working as expected. any idea what to search for? i enabled sofia debug, here is the logs. working - https://pastebin.com/eJDRmfyq not working - https://pastebin.com/YT3dFUi3 You can see that in the not working the sipp trying again and again to send BYE, while in the working conference we receive the hangup after the first BYE try. Any idea? Regards, Arye ‫בתאריך יום ג׳, 15 בפבר׳ 2022 ב-11:31 מאת אריה קלטר <‪aryeklt at gmail.com‬‏>:‬ > Freeswitch version 1.10.7-release~64bit on centos 7 server > The load is something around 200 - 800 sessions in conference > > > ‫בתאריך יום ג׳, 15 בפבר׳ 2022 ב-11:19 מאת ‪Brian :‬‏ <‪brians at iptel.co > ‬‏>:‬ > >> What version of freeswitch and OS? >> >> What type of load? >> >> On Monday, February 14, 2022, אריה קלטר wrote: >> > Hello group >> > >> > I have problem with one of our servers, that sometimes it starting to >> stuck, on all new calls the last log line is "push codec L16:100" and >> nothing helps to end the call, no response to bye from the calling server, >> no response to sip info and even no response to hupall, the only thing that >> solves the problem is restart the freeswitch itself. >> > It happened two times. >> > Any idea what is going on? >> > Example from one of the calls? >> > fe4849de-cb96-4a4a-ad02-dc7485cc87ee 2022-02-14 08:42:36.832268 92.10% >> [NOTICE] mod_dptools.c:1419 Channel [sofia/external/@] has been >> answered >> > fe4849de-cb96-4a4a-ad02-dc7485cc87ee 2022-02-14 08:42:36.832268 92.10% >> [DEBUG] switch_channel.c:3950 (sofia/external/ @ ) Callstate >> Change EARLY -> ACTIVE >> > fe4849de-cb96-4a4a-ad02-dc7485cc87ee 2022-02-14 08:42:36.832268 92.10% >> [DEBUG] sofia.c:7499 Channel sofia/external/ @ entering state >> [completed][200] >> > fe4849de-cb96-4a4a-ad02-dc7485cc87ee 2022-02-14 08:42:36.832268 92.10% >> [DEBUG] switch_ivr.c:632 sofia/external/ @ Command Execute >> [depth=1] conference(conf10 at default+flags{}) >> > fe4849de-cb96-4a4a-ad02-dc7485cc87ee EXECUTE [depth=1] sofia/external/ >> @ conference(conf10 at default+flags{}) >> > fe4849de-cb96-4a4a-ad02-dc7485cc87ee 2022-02-14 08:42:36.832268 92.10% >> [DEBUG] sofia.c:7499 Channel sofia/external/ @ entering state >> [ready][200] >> > fe4849de-cb96-4a4a-ad02-dc7485cc87ee 2022-02-14 08:42:36.832268 92.10% >> [DEBUG] conference_member.c:1794 Raw Codec Activation Success L16 at 8000hz >> 1 channel 20ms >> > fe4849de-cb96-4a4a-ad02-dc7485cc87ee 2022-02-14 08:42:36.832268 92.10% >> [DEBUG] conference_member.c:1841 Raw Codec Activation Success L16 at 8000hz >> 1 channel 20ms >> > and the next line is >> > fe4849de-cb96-4a4a-ad02-dc7485cc87ee 2022-02-14 18:15:31.312291 95.10% >> [NOTICE] switch_core_session.c:407 Hangup sofia/external/ @ >> [CS_EXECUTE] [MANAGER_REQUEST] >> > because i did hupall, but the call is still there until restart of the >> freeswitch itself, even after the hupall. >> > Regards, >> > Arye >> > >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From sdevoy at bizfocused.com Mon Feb 21 03:28:04 2022 From: sdevoy at bizfocused.com (Sean Devoy) Date: Mon, 21 Feb 2022 03:28:04 +0000 Subject: [Freeswitch-users] voicemail from domain In-Reply-To: References: Message-ID: Hi, I finally found that setting these values in the directory file, specifically the voicemail directory for the user works. Here is an example: /> /> I first made the mistake of adding these to the user’s extension, but it must be that user’s voicemail extension. HTH, Sean From: FreeSWITCH-users On Behalf Of Kim Culhan Sent: Saturday, February 19, 2022 10:41 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] voicemail from domain Running FreeSWITCH Version 1.10.3-release~64bit (-release 64bit) Emails sent with voicemail messages have a from line: extension at ip_address_of_external interface. In voicemail.conf.xml have: This did not change it, still has @external_ip_address Any help is greatly appreciated. thanks -kim -------------- next part -------------- An HTML attachment was scrubbed... URL: From Alexander.Haugg at c4b.de Mon Feb 21 10:41:35 2022 From: Alexander.Haugg at c4b.de (Alexander Haugg) Date: Mon, 21 Feb 2022 10:41:35 +0000 Subject: [Freeswitch-users] Media timeout In-Reply-To: References: Message-ID: <02ca24091c3d436d86e682d3c8a530e1@c4b.de> OK, I need to correct my statement. If I configure a timer for the sip profile configuration and specify the value for media_timeout correctly in milliseconds, everything seems to work. Thanks a lot. Von: FreeSWITCH-users Im Auftrag von Dragos Oancea Gesendet: Freitag, 18. Februar 2022 11:48 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] Media timeout This has been discussed on this ML a while ago. Documentation out of date. On Fri, Feb 18, 2022 at 11:00 AM Shaun Stokes > wrote: I've figured out our issue with media_timeout_audio, this is using ms not seconds. Is this a bug, or is the documentation out of date? ________________________________ From: FreeSWITCH-users > on behalf of Dragos Oancea > Sent: 16 February 2022 13:49 To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Media timeout please open a github issue. https://github.com/signalwire/freeswitch/issues you could also check if switch_rtp_set_max_missed_packets() and/or switch_rtp_set_media_timeout() are called in your particular call setup, with your configs. On Tue, Feb 15, 2022 at 11:33 PM Alexander Haugg > wrote: Hi Shaun, I have noticed the same thing. "media_timout" in conjunction with WebRTC does not work at all. If someone has a solution, that would be really fine. Currently I have helped myself by re-enabling the following in the code: diff --git a/src/switch_rtp.c b/src/switch_rtp.c index 40d8978..aada64a 100644 --- a/src/switch_rtp.c +++ b/src/switch_rtp.c @@ -854,8 +854,8 @@ static switch_status_t ice_out(switch_rtp_t *rtp_session, switch_rtp_ice_t *ice) if (elapsed > 30000) { switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_WARNING, "No %s rtp_session->last_stun = switch_micro_time_now(); - //status = SWITCH_STATUS_GENERR; - //goto end; + status = SWITCH_STATUS_GENERR; + goto end; } } @@ -897,7 +897,7 @@ static switch_status_t ice_out(switch_rtp_t *rtp_session, switch_rtp_ice_t *ice) ice->sending = 3; - // end: + end: READ_DEC(rtp_session); return status; (END) The code has the consequence that the session is cleared if no more media comes for 30 seconds. With kind regards Alex Von: FreeSWITCH-users > Im Auftrag von Shaun Stokes Gesendet: Donnerstag, 10. Februar 2022 14:39 An: FreeSWITCH Users Help > Betreff: [Freeswitch-users] Media timeout Hi All, I'm not sure if others have had similar experiences but for us the media_timeout variable does not work as expected. * media_timeout on a call that supports video but with-out video will fail. * media_timeout_audio works in some instances, in others the timeout period is ignored so the call will timeout almost immediately after RTP stops. * media_hold_timeout_audio doesn't seem to work at all, calls that are on hold never timeout. Why is the SIP profile parameter 'rtp-timeout-sec' depreciated? It's much simpler to apply this per SIP profile than it is per call. Thanks, Shaun _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From rasheed.kalapurackal at gmail.com Mon Feb 21 12:58:24 2022 From: rasheed.kalapurackal at gmail.com (Rasheed Kalapurackal) Date: Mon, 21 Feb 2022 18:28:24 +0530 Subject: [Freeswitch-users] Limit concurrent calls for inbound + outbound on a specific DID number Message-ID: Hello , Is it possible to limit the total number of inbound + outbound calls made or received on a specific DID number to 10 ? Thanks Rasheed -------------- next part -------------- An HTML attachment was scrubbed... URL: From avi at avimarcus.net Mon Feb 21 13:15:53 2022 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 21 Feb 2022 13:15:53 +0000 Subject: [Freeswitch-users] Limit concurrent calls for inbound + outbound on a specific DID number In-Reply-To: References: Message-ID: <0100017f1c6c997f-cd5fad63-28a1-4ae4-b4fc-005812418fbc-000000@email.amazonses.com> Use mod_limit Set the same name for the inbound and outbound routes if you want to count them together. On Mon, Feb 21, 2022, 2:59 PM Rasheed Kalapurackal < rasheed.kalapurackal at gmail.com> wrote: > Hello , > > Is it possible to limit the total number of inbound + outbound calls made > or received on a specific DID number to 10 ? > > Thanks > Rasheed > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From Antony.Stone at freeswitch.open.source.it Mon Feb 21 13:21:17 2022 From: Antony.Stone at freeswitch.open.source.it (Antony Stone) Date: Mon, 21 Feb 2022 13:21:17 +0000 Subject: [Freeswitch-users] Limit concurrent calls for inbound + outbound on a specific DID number In-Reply-To: References: Message-ID: <202202211321.18000.Antony.Stone@freeswitch.open.source.it> On Monday 21 February 2022 at 12:58:24, Rasheed Kalapurackal wrote: > Hello , > > Is it possible to limit the total number of inbound + outbound calls made > or received on a specific DID number to 10 ? 1. How is this DID connected to your system? Is it on a trunk where other DIDs also receive calls, or is it on a dedicated connection for the DID only? 2. How does a DID relate to outbound calls? If you mean "restrict the number of simultaneous calls presenting a certain caller ID" then I doubt that this is possible. Antony. -- If the human brain were so simple that we could understand it, we'd be so simple that we couldn't. Please reply to the list; please *don't* CC me. From rasheed.kalapurackal at gmail.com Mon Feb 21 13:49:12 2022 From: rasheed.kalapurackal at gmail.com (Rasheed Kalapurackal) Date: Mon, 21 Feb 2022 19:19:12 +0530 Subject: [Freeswitch-users] Limit concurrent calls for inbound + outbound on a specific DID number In-Reply-To: <202202211321.18000.Antony.Stone@freeswitch.open.source.it> References: <202202211321.18000.Antony.Stone@freeswitch.open.source.it> Message-ID: Hello Antony, Following the responses to your queries. 1. How is this DID connected to your system? Is it on a trunk where other DIDs also receive calls, or is it on a dedicated connection for the DID only? - DID is connected on a single trunk where other DIDs also will receive the calls. it is not a dedicated connection. 2. How does a DID relate to outbound calls? If you mean "restrict the number of simultaneous calls presenting a certain caller ID" then I doubt that this is possible. - I just want to limit the total number of calls (inbound + outbound) using the same DID . For example: If my DID is 920034576 , if i receive a call to this DID then one channel is counted , at the same time if i make a call using this DID like the following This case it will count 2 calls as total - 1 incoming and 1 outgoing using same DID. This total calls should be limited to 10 . Thanks and Regards Rasheed On Mon, Feb 21, 2022 at 7:12 PM Antony Stone < Antony.Stone at freeswitch.open.source.it> wrote: > On Monday 21 February 2022 at 12:58:24, Rasheed Kalapurackal wrote: > > > Hello , > > > > Is it possible to limit the total number of inbound + outbound calls made > > or received on a specific DID number to 10 ? > > 1. How is this DID connected to your system? Is it on a trunk where other > DIDs also receive calls, or is it on a dedicated connection for the DID > only? > > 2. How does a DID relate to outbound calls? If you mean "restrict the > number > of simultaneous calls presenting a certain caller ID" then I doubt that > this > is possible. > > > Antony. > > -- > If the human brain were so simple that we could understand it, > we'd be so simple that we couldn't. > > Please reply to the > list; > please *don't* CC > me. > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From alexanderhenryperkins at gmail.com Mon Feb 21 14:25:05 2022 From: alexanderhenryperkins at gmail.com (Alexander Perkins) Date: Mon, 21 Feb 2022 09:25:05 -0500 Subject: [Freeswitch-users] Read Header Message-ID: Hi All. I have a custom header that will be passed down to be by our provider and I would like to read that header. How can I do that? It is one of those X- headers. Thank you, Alex -------------- next part -------------- An HTML attachment was scrubbed... URL: From thomas.peterseil at mine-project.eu Mon Feb 21 16:56:45 2022 From: thomas.peterseil at mine-project.eu (thomas peterseil) Date: Mon, 21 Feb 2022 17:56:45 +0100 Subject: [Freeswitch-users] call duration for bash script Message-ID: hello freeswitch-users, i would like to run a bash script after a call and for this script i need the duration of that call. i found the variable "billsec", but i couldn´t find any dialplan examples how to use this variable. can someone give me a hint how i can hand over the duration of the call to a bash script in the dialplan. thank you very much! best regards, thomas From mrjoli021 at gmail.com Mon Feb 21 19:29:08 2022 From: mrjoli021 at gmail.com (Joli Martinez) Date: Mon, 21 Feb 2022 14:29:08 -0500 Subject: [Freeswitch-users] Python Development Message-ID: Hello, I have written some scripts in Lua for freeswitch, but I would like to switch over to Python since I am more familiar with the language. I am not sure how to install the freeswitch module. I would be running Debian 10 or 11. I see that there are examples online but they all require importing freeswitch. When I run pip, all the available pip packages with the word Freeswitch is ESL or Eventsocket. When researching the ESL setup is different. So I am sort of confused. Could someone point me in the right direction as to how to install the correct packages and a sample "Hello World" script. I should be able to take it from there. -------------- next part -------------- An HTML attachment was scrubbed... URL: From mgpx38 at gmail.com Mon Feb 21 20:13:07 2022 From: mgpx38 at gmail.com (MG PX) Date: Mon, 21 Feb 2022 21:13:07 +0100 Subject: [Freeswitch-users] Unable to establish end-to-end OPUS communication with installation freeswitch configuration In-Reply-To: References: Message-ID: Hello all, I did a test with a softphone configured in G722 only and a softphone configured in OPUS only and there everything works fine. I hope it can give an idea to someone who has more expertise than me in FS to understand what happens with pure OPUS communications. Thank you in advance for your help. BR Le mar. 15 févr. 2022 à 12:06, MG PX a écrit : > Freeswitch is at ip address 192.168.0.27 > My softphone #1 is registered with freeswitch as user 1005. It is at ip > address 192.168.0.30 > My softphone #2 is registered with freeswitch as user 1006. It is at ip > address 192.168.0.16 > 1005 calls 1006 using OPUS codec. > The call does not succeed because the media stream (audio only in my case) > initially established between FS and the caller 1005 is interrupted after > FS has setup the communication with callee 1006 and sent the SIP OK to > caller 1005. All other media streams are correctly established. > I got this behaviour only with OPUS codec. It works correctly with G722 > codec for instance. > I tried with freeswitch 1.8.7 under windows and freeswitch 1.10.7 under > linux and got the same result. > I tried different freeswitch settings with the same result. > Can someone help me to solve this problem? I don't know where to look > anymore. > > I attached a wireshark capture. It shows that FS is not sending media > packets anymore to 1005 (192.168.0.3) after it has sent the SIP OK (packet > #2360). > I also attached freeswitch log. > -------------- next part -------------- An HTML attachment was scrubbed... URL: From dragos at freeswitch.org Mon Feb 21 20:22:39 2022 From: dragos at freeswitch.org (Dragos Oancea) Date: Mon, 21 Feb 2022 22:22:39 +0200 Subject: [Freeswitch-users] Unable to establish end-to-end OPUS communication with installation freeswitch configuration In-Reply-To: References: Message-ID: Try: We can look on it if there's a github issue. On Mon, Feb 21, 2022 at 10:14 PM MG PX wrote: > Hello all, > I did a test with a softphone configured in G722 only and a softphone > configured in OPUS only and there everything works fine. I hope it can give > an idea to someone who has more expertise than me in FS to understand what > happens with pure OPUS communications. > Thank you in advance for your help. > BR > > Le mar. 15 févr. 2022 à 12:06, MG PX a écrit : > >> Freeswitch is at ip address 192.168.0.27 >> My softphone #1 is registered with freeswitch as user 1005. It is at ip >> address 192.168.0.30 >> My softphone #2 is registered with freeswitch as user 1006. It is at ip >> address 192.168.0.16 >> 1005 calls 1006 using OPUS codec. >> The call does not succeed because the media stream (audio only in my >> case) initially established between FS and the caller 1005 is interrupted >> after FS has setup the communication with callee 1006 and sent the SIP OK >> to caller 1005. All other media streams are correctly established. >> I got this behaviour only with OPUS codec. It works correctly with G722 >> codec for instance. >> I tried with freeswitch 1.8.7 under windows and freeswitch 1.10.7 under >> linux and got the same result. >> I tried different freeswitch settings with the same result. >> Can someone help me to solve this problem? I don't know where to look >> anymore. >> >> I attached a wireshark capture. It shows that FS is not sending media >> packets anymore to 1005 (192.168.0.3) after it has sent the SIP OK (packet >> #2360). >> I also attached freeswitch log. >> > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From aryeklt at gmail.com Mon Feb 21 23:25:30 2022 From: aryeklt at gmail.com (=?UTF-8?B?15DXqNeZ15Qg16fXnNeY16g=?=) Date: Tue, 22 Feb 2022 01:25:30 +0200 Subject: [Freeswitch-users] freeswitch stuck at Push codec L16:100 In-Reply-To: References: Message-ID: i found how to use the debug tools i saw lot of sessions with the same locked trace as in this paste below. any idea why it happens? https://pastebin.com/YZRdmeAQ #0 0x00002afdf6c5f54d in __lll_lock_wait () from /usr/lib64/libpthread.so.0 #1 0x00002afdf6c5aeb6 in _L_lock_941 () from /usr/lib64/libpthread.so.0 #2 0x00002afdf6c5adaf in pthread_mutex_lock () from /usr/lib64/libpthread.so.0 #3 0x00002afdf358c5ff in apr_thread_mutex_lock () from /usr/lib64/libfreeswitch.so.1 #4 0x00002afdf3221bb2 in switch_mutex_lock (lock=0x38a9cae8) at src/switch_apr.c:301 #5 0x00002afe1001c7da in conference_member_add (nce=0x38a9c108, member=0x2afe4f326f40) at conference_member.c:732 ‫בתאריך יום א׳, 20 בפבר׳ 2022 ב-20:08 מאת אריה קלטר <‪aryeklt at gmail.com‬‏>:‬ > just update > it happened again one day after the last post, 2022-02-16 > but it stucked only on one specific conference, on other conferences it is > working as expected. > any idea what to search for? > > i enabled sofia debug, here is the logs. > > working - https://pastebin.com/eJDRmfyq > not working - https://pastebin.com/YT3dFUi3 > > You can see that in the not working the sipp trying again and again to > send BYE, while in the working conference we receive the hangup after the > first BYE try. > > Any idea? > > Regards, > Arye > > ‫בתאריך יום ג׳, 15 בפבר׳ 2022 ב-11:31 מאת אריה קלטר <‪aryeklt at gmail.com > ‬‏>:‬ > >> Freeswitch version 1.10.7-release~64bit on centos 7 server >> The load is something around 200 - 800 sessions in conference >> >> >> ‫בתאריך יום ג׳, 15 בפבר׳ 2022 ב-11:19 מאת ‪Brian :‬‏ <‪brians at iptel.co >> ‬‏>:‬ >> >>> What version of freeswitch and OS? >>> >>> What type of load? >>> >>> On Monday, February 14, 2022, אריה קלטר wrote: >>> > Hello group >>> > >>> > I have problem with one of our servers, that sometimes it starting to >>> stuck, on all new calls the last log line is "push codec L16:100" and >>> nothing helps to end the call, no response to bye from the calling server, >>> no response to sip info and even no response to hupall, the only thing that >>> solves the problem is restart the freeswitch itself. >>> > It happened two times. >>> > Any idea what is going on? >>> > Example from one of the calls? >>> > fe4849de-cb96-4a4a-ad02-dc7485cc87ee 2022-02-14 08:42:36.832268 92.10% >>> [NOTICE] mod_dptools.c:1419 Channel [sofia/external/@] has been >>> answered >>> > fe4849de-cb96-4a4a-ad02-dc7485cc87ee 2022-02-14 08:42:36.832268 92.10% >>> [DEBUG] switch_channel.c:3950 (sofia/external/ @ ) Callstate >>> Change EARLY -> ACTIVE >>> > fe4849de-cb96-4a4a-ad02-dc7485cc87ee 2022-02-14 08:42:36.832268 92.10% >>> [DEBUG] sofia.c:7499 Channel sofia/external/ @ entering state >>> [completed][200] >>> > fe4849de-cb96-4a4a-ad02-dc7485cc87ee 2022-02-14 08:42:36.832268 92.10% >>> [DEBUG] switch_ivr.c:632 sofia/external/ @ Command Execute >>> [depth=1] conference(conf10 at default+flags{}) >>> > fe4849de-cb96-4a4a-ad02-dc7485cc87ee EXECUTE [depth=1] sofia/external/ >>> @ conference(conf10 at default+flags{}) >>> > fe4849de-cb96-4a4a-ad02-dc7485cc87ee 2022-02-14 08:42:36.832268 92.10% >>> [DEBUG] sofia.c:7499 Channel sofia/external/ @ entering state >>> [ready][200] >>> > fe4849de-cb96-4a4a-ad02-dc7485cc87ee 2022-02-14 08:42:36.832268 92.10% >>> [DEBUG] conference_member.c:1794 Raw Codec Activation Success L16 at 8000hz >>> 1 channel 20ms >>> > fe4849de-cb96-4a4a-ad02-dc7485cc87ee 2022-02-14 08:42:36.832268 92.10% >>> [DEBUG] conference_member.c:1841 Raw Codec Activation Success L16 at 8000hz >>> 1 channel 20ms >>> > and the next line is >>> > fe4849de-cb96-4a4a-ad02-dc7485cc87ee 2022-02-14 18:15:31.312291 95.10% >>> [NOTICE] switch_core_session.c:407 Hangup sofia/external/ @ >>> [CS_EXECUTE] [MANAGER_REQUEST] >>> > because i did hupall, but the call is still there until restart of the >>> freeswitch itself, even after the hupall. >>> > Regards, >>> > Arye >>> > >>> _________________________________________________________________________ >>> >>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>> services. >>> Build your next product on our scalable cloud platform. >>> >>> Join our online community to chat in real time >>> https://signalwire.community >>> >>> Professional FreeSWITCH Services >>> sales at freeswitch.com >>> https://freeswitch.com >>> >>> Official FreeSWITCH Sites >>> https://freeswitch.com/oss >>> https://freeswitch.org/confluence >>> https://cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: From csadi at hotmail.com Tue Feb 22 16:33:45 2022 From: csadi at hotmail.com (Adiseshu Channasamudhram) Date: Tue, 22 Feb 2022 16:33:45 +0000 Subject: [Freeswitch-users] How to copy from header In-Reply-To: References: Message-ID: Hello Team As I have not got any response, trying this a different way How can I pass aniii to outgoing from header? Regards Adi Sent from my iPhone On Feb 18, 2022, at 7:23 AM, Adiseshu Channasamudhram wrote:  Hello FS experts Can someone please help me with this? Thanks Adi Sent from my iPhone On Feb 17, 2022, at 12:05 PM, Adiseshu Channasamudhram wrote:  Hello Team I have a requirement to copy from header from the incoming INVITE to the outgoing INVITE right now, below is the from header from incomign invite From: "123-456-7788" ;tag=2bf345ee but the outgoing INVITE has the from header as below From: "123-456-7788" ;tag=yaXFUFjD666ym How can i copy the cpc and isup-oli from the incoming from header to the outgoing header? Thanks in advance Regards Adi -------------- next part -------------- An HTML attachment was scrubbed... URL: From sdevoy at bizfocused.com Tue Feb 22 16:56:15 2022 From: sdevoy at bizfocused.com (Sean Devoy) Date: Tue, 22 Feb 2022 16:56:15 +0000 Subject: PCAP help and many call failures PLEASE HELP Message-ID: Hi, This is a very serious problem for my users AT this site. PLEASE help. I suspect it is a NAT issue reaching the extension. I have tried with no change. I have tried TCP instead of UDP. Details of the problem: User extension initiates a call (INVITE after auth required). User hears ringing for approximately 20 seconds, then a busy signal. Today a user called her own cell phone and said it WAS ringing when the call failed. I see some RTP packets between 183 session progress from FS to extension and CANCEL, but can't tell what extension hey are to/from. I see a very single strange RTP from FS to itself: unknown RTP protocol version 3. The call CANCEL is initiated from FreeSwitch at 22.75 seconds after the invite to FS. PCAP issue: I am trying to trace down a problem with tshark. The short version of my question is how do I tell which NAT extension RTP packets are going to and why did FS cancel the call? The full explanation is: I have identified the failed call based on the INVITE from the NAT extension's IP and PORT number. I am able to follow the SIP packets is OK. I see RTP packets to and from that IP address, but I cannot determine what extension it is going to. I also see an ICMP from the FSserver to the extension's external IP aftern the CANCEL. We block ICMP. Is that a problem? I have the full pcap file if someone could PLEASE help Pcacp file: http://brianbunce.com/capturefile.pcap (11 MB) It is a customer site, be kind. The invite from the ext is packet# 2334. The invite to the GW is packet# 2357. That will show you all IPs involved. Thanks for any help, Sean From mayamatakeshi at gmail.com Tue Feb 22 22:08:54 2022 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Wed, 23 Feb 2022 07:08:54 +0900 Subject: [Freeswitch-users] PCAP help and many call failures PLEASE HELP In-Reply-To: References: Message-ID: On Wed, Feb 23, 2022 at 2:20 AM Sean Devoy via FreeSWITCH-users < freeswitch-users at lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Sean Devoy > To: FreeSWITCH Users Help > Cc: > Bcc: > Date: Tue, 22 Feb 2022 16:56:15 +0000 > Subject: PCAP help and many call failures PLEASE HELP > Hi, > This is a very serious problem for my users AT this site. PLEASE help. > > I suspect it is a NAT issue reaching the extension. I have tried name="sip-force-contact" value="NDLB-connectile-dysfunction"/> with no > change. > I have tried TCP instead of UDP. > > Details of the problem: > User extension initiates a call (INVITE after auth required). > User hears ringing for approximately 20 seconds, then a busy signal. Today > a user called her own cell phone and said it WAS ringing when the call > failed. > I see some RTP packets between 183 session progress from FS to extension > and CANCEL, but can't tell what extension hey are to/from. > I see a very single strange RTP from FS to itself: unknown RTP protocol > version 3. > The call CANCEL is initiated from FreeSwitch at 22.75 seconds after the > invite to FS. > > PCAP issue: > I am trying to trace down a problem with tshark. > The short version of my question is how do I tell which NAT extension RTP > packets are going to and why did FS cancel the call? > > The full explanation is: > I have identified the failed call based on the INVITE from the NAT > extension's IP and PORT number. I am able to follow the SIP packets is > OK. I see RTP packets to and from that IP address, but I cannot determine > what extension it is going to. > > I also see an ICMP from the FSserver to the extension's external IP aftern > the CANCEL. We block ICMP. Is that a problem? > > I have the full pcap file if someone could PLEASE help > Pcacp file: http://brianbunce.com/capturefile.pcap (11 MB) It is a > customer site, be kind. > The invite from the ext is packet# 2334. The invite to the GW is packet# > 2357. That will show you all IPs involved. > > I am not a security expert and I am not sure if it is wise to share capture files with unredacted information. Anyway, I took a look and I think what matters is to understand why FS decided to cancel the call. If you have the XML CDRs of the channels involved, they might give some clue about it. If you don't have them, please show the complete dialplan you are executing as there might be something there like call_timeout etc. Then even if with those things the cause of CANCEL cannot be found, one alternative if you are able to reproduce this problem at will or if the problem happens frequently enough would be to try to enable debug logging during low traffic period by doing: fs_cli -x "fsctl loglevel DEBUG" +OK log level: DEBUG [7] Then make a call that recreates the problem and switch back to the original log level (usually, ERR): fs_cli -x "fsctl loglevel ERR" +OK log level: ERR [3] Then with the logs, you should be able to understand why FS is deciding to cancel the call. -------------- next part -------------- An HTML attachment was scrubbed... URL: From mgpx38 at gmail.com Wed Feb 23 10:16:30 2022 From: mgpx38 at gmail.com (MG PX) Date: Wed, 23 Feb 2022 11:16:30 +0100 Subject: [Freeswitch-users] Unable to establish end-to-end OPUS communication with installation freeswitch configuration In-Reply-To: References: Message-ID: Hello Dragos, Unfortunately it didn't change anything. I will create a github issue. Thanks Le lun. 21 févr. 2022 à 21:56, Dragos Oancea a écrit : > Try: > > > > We can look on it if there's a github issue. > > > > On Mon, Feb 21, 2022 at 10:14 PM MG PX wrote: > >> Hello all, >> I did a test with a softphone configured in G722 only and a softphone >> configured in OPUS only and there everything works fine. I hope it can give >> an idea to someone who has more expertise than me in FS to understand what >> happens with pure OPUS communications. >> Thank you in advance for your help. >> BR >> >> Le mar. 15 févr. 2022 à 12:06, MG PX a écrit : >> >>> Freeswitch is at ip address 192.168.0.27 >>> My softphone #1 is registered with freeswitch as user 1005. It is at ip >>> address 192.168.0.30 >>> My softphone #2 is registered with freeswitch as user 1006. It is at ip >>> address 192.168.0.16 >>> 1005 calls 1006 using OPUS codec. >>> The call does not succeed because the media stream (audio only in my >>> case) initially established between FS and the caller 1005 is interrupted >>> after FS has setup the communication with callee 1006 and sent the SIP OK >>> to caller 1005. All other media streams are correctly established. >>> I got this behaviour only with OPUS codec. It works correctly with G722 >>> codec for instance. >>> I tried with freeswitch 1.8.7 under windows and freeswitch 1.10.7 under >>> linux and got the same result. >>> I tried different freeswitch settings with the same result. >>> Can someone help me to solve this problem? I don't know where to look >>> anymore. >>> >>> I attached a wireshark capture. It shows that FS is not sending media >>> packets anymore to 1005 (192.168.0.3) after it has sent the SIP OK (packet >>> #2360). >>> I also attached freeswitch log. >>> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From dragos at freeswitch.org Wed Feb 23 10:20:11 2022 From: dragos at freeswitch.org (Dragos Oancea) Date: Wed, 23 Feb 2022 12:20:11 +0200 Subject: [Freeswitch-users] How to copy from header In-Reply-To: References: Message-ID: maybe reading chan var "sip_full_from" and putting its contents in a special sip "X-" header. On Tue, Feb 22, 2022 at 6:34 PM Adiseshu Channasamudhram wrote: > Hello Team > > As I have not got any response, trying this a different way > > How can I pass aniii to outgoing from header? > > Regards > > Adi > > Sent from my iPhone > > On Feb 18, 2022, at 7:23 AM, Adiseshu Channasamudhram > wrote: > >  Hello FS experts > > Can someone please help me with this? > > Thanks > > Adi > > Sent from my iPhone > > On Feb 17, 2022, at 12:05 PM, Adiseshu Channasamudhram > wrote: > >  > Hello Team > > I have a requirement to copy from header from the incoming INVITE to the > outgoing INVITE > > right now, below is the from header from incomign invite > > From: "123-456-7788" ;user=phone;isup-oli=0>;tag=2bf345ee > > but the outgoing INVITE has the from header as below > > From: "123-456-7788" ;tag=yaXFUFjD666ym > > How can i copy the cpc and isup-oli from the incoming from header to the > outgoing header? > > Thanks in advance > > Regards > > Adi > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From kaiduanx at yahoo.ca Wed Feb 23 15:08:19 2022 From: kaiduanx at yahoo.ca (kaiduan xie) Date: Wed, 23 Feb 2022 15:08:19 +0000 (UTC) Subject: [Freeswitch-users] How to copy from header In-Reply-To: References: Message-ID: <1280002399.817722.1645628899707@mail.yahoo.com> Adiseshu, If you want to copy the cpc and isup-oli part from the incoming From header to outgoing From header, you have to change source code. Regards, /Kaiduan On Wednesday, February 23, 2022, 05:21:06 a.m. EST, Dragos Oancea wrote: maybe reading chan var "sip_full_from"  and putting its contents in a special sip "X-"  header.  On Tue, Feb 22, 2022 at 6:34 PM Adiseshu Channasamudhram wrote: Hello Team As I have not got any response, trying this a different way How can I pass aniii to outgoing from header? Regards Adi Sent from my iPhone On Feb 18, 2022, at 7:23 AM, Adiseshu Channasamudhram wrote:  Hello FS experts Can someone please help me with this? Thanks Adi Sent from my iPhone On Feb 17, 2022, at 12:05 PM, Adiseshu Channasamudhram wrote: Hello Team I have a requirement to copy from header from the incoming INVITE to the outgoing INVITE right now, below is the from header from incomign invite From: "123-456-7788" ;tag=2bf345ee but the outgoing INVITE has the from header as below From: "123-456-7788" ;tag=yaXFUFjD666ym How can i copy the cpc and isup-oli from the incoming from header to the outgoing header? Thanks in advance Regards Adi _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From kaduww at gmail.com Wed Feb 23 15:13:56 2022 From: kaduww at gmail.com (Carlos Eduardo) Date: Wed, 23 Feb 2022 12:13:56 -0300 Subject: [Freeswitch-users] How to copy from header In-Reply-To: References: Message-ID: you can try copying the from header received export sip_from_uri=${sip_from_uri} Em qua., 23 de fev. de 2022 às 07:20, Dragos Oancea escreveu: > maybe reading chan var "sip_full_from" and putting its contents in a > special sip "X-" header. > > On Tue, Feb 22, 2022 at 6:34 PM Adiseshu Channasamudhram < > csadi at hotmail.com> wrote: > >> Hello Team >> >> As I have not got any response, trying this a different way >> >> How can I pass aniii to outgoing from header? >> >> Regards >> >> Adi >> >> Sent from my iPhone >> >> On Feb 18, 2022, at 7:23 AM, Adiseshu Channasamudhram >> wrote: >> >>  Hello FS experts >> >> Can someone please help me with this? >> >> Thanks >> >> Adi >> >> Sent from my iPhone >> >> On Feb 17, 2022, at 12:05 PM, Adiseshu Channasamudhram >> wrote: >> >>  >> Hello Team >> >> I have a requirement to copy from header from the incoming INVITE to the >> outgoing INVITE >> >> right now, below is the from header from incomign invite >> >> From: "123-456-7788" > ;user=phone;isup-oli=0>;tag=2bf345ee >> >> but the outgoing INVITE has the from header as below >> >> From: "123-456-7788" ;tag=yaXFUFjD666ym >> >> How can i copy the cpc and isup-oli from the incoming from header to the >> outgoing header? >> >> Thanks in advance >> >> Regards >> >> Adi >> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- *Carlos E. Wagner* *Tecnólogo em Telecomunicações, Opensips Certified Professional* *Fone: +55 48 99981-0894* *E-mail:* kaduww at gmail.com *LinkedIn:* https://www.linkedin.com/in/carlos-eduardo-wagner-96bbb433/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From botelist at gmail.com Wed Feb 23 15:55:51 2022 From: botelist at gmail.com (Bote Man) Date: Wed, 23 Feb 2022 10:55:51 -0500 Subject: [Freeswitch-users] voicemail from domain In-Reply-To: References: Message-ID: <006001d828cd$d5a00370$80e00a50$@gmail.com> Sean, what is the difference between the user's extension and the voicemail extension? They should be the same, no? I found a plethora of variables and parameters that you can set in: https://freeswitch.org/confluence/display/FREESWITCH/mod_voicemail Hope this helps. John Boteler Bote Communications -----Original Message----- From: FreeSWITCH-users On Behalf Of Sean Devoy via FreeSWITCH-users Sent: Sunday, 20 February, 2022 22:50 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] voicemail from domain Hi, I finally found that setting these values in the directory file, specifically the voicemail directory for the user works. Here is an example: I first made the mistake of adding these to the user’s extension, but it must be that user’s voicemail extension. HTH, Sean From csadi at hotmail.com Wed Feb 23 16:14:42 2022 From: csadi at hotmail.com (Adiseshu Channasamudhram) Date: Wed, 23 Feb 2022 16:14:42 +0000 Subject: [Freeswitch-users] How to copy from header In-Reply-To: References: Message-ID: Thank you very mush - this helped Sent from my iPhone On Feb 23, 2022, at 10:13 AM, Carlos Eduardo wrote:  you can try copying the from header received export sip_from_uri=${sip_from_uri} Em qua., 23 de fev. de 2022 às 07:20, Dragos Oancea > escreveu: maybe reading chan var "sip_full_from" and putting its contents in a special sip "X-" header. On Tue, Feb 22, 2022 at 6:34 PM Adiseshu Channasamudhram > wrote: Hello Team As I have not got any response, trying this a different way How can I pass aniii to outgoing from header? Regards Adi Sent from my iPhone On Feb 18, 2022, at 7:23 AM, Adiseshu Channasamudhram > wrote:  Hello FS experts Can someone please help me with this? Thanks Adi Sent from my iPhone On Feb 17, 2022, at 12:05 PM, Adiseshu Channasamudhram > wrote:  Hello Team I have a requirement to copy from header from the incoming INVITE to the outgoing INVITE right now, below is the from header from incomign invite From: "123-456-7788" ;tag=2bf345ee but the outgoing INVITE has the from header as below From: "123-456-7788" >;tag=yaXFUFjD666ym How can i copy the cpc and isup-oli from the incoming from header to the outgoing header? Thanks in advance Regards Adi _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -- Carlos E. Wagner Tecnólogo em Telecomunicações, Opensips Certified Professional Fone: +55 48 99981-0894 E-mail: kaduww at gmail.com LinkedIn: https://www.linkedin.com/in/carlos-eduardo-wagner-96bbb433/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From botelist at gmail.com Wed Feb 23 16:17:55 2022 From: botelist at gmail.com (Bote Man) Date: Wed, 23 Feb 2022 11:17:55 -0500 Subject: [Freeswitch-users] Read Header In-Reply-To: References: Message-ID: <006c01d828d0$ea971c70$bfc55550$@gmail.com> Read the channel variable ${sip_h_X-Custom-Header-Value} to read “X-Custom-Header-Value” from the SIP header. Read all about it: https://freeswitch.org/confluence/display/FREESWITCH/Sofia+SIP+Stack John Boteler Bote Communications From: FreeSWITCH-users On Behalf Of Alexander Perkins Sent: Monday, 21 February, 2022 09:25 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Read Header Hi All. I have a custom header that will be passed down to be by our provider and I would like to read that header. How can I do that? It is one of those X- headers. Thank you, Alex -------------- next part -------------- An HTML attachment was scrubbed... URL: From botelist at gmail.com Wed Feb 23 16:28:57 2022 From: botelist at gmail.com (Bote Man) Date: Wed, 23 Feb 2022 11:28:57 -0500 Subject: [Freeswitch-users] call duration for bash script In-Reply-To: References: Message-ID: <007101d828d2$756339f0$6029add0$@gmail.com> I don't know a good answer to your question, but over the years I understand that the best way to solve these kinds of problems is to read the call detail log and parse out the desired values from there. This is best done outside of the dialplan, so you would trigger your script to start its processing while the dialplan completes the call. John Boteler Bote Communications -----Original Message----- From: FreeSWITCH-users On Behalf Of thomas peterseil Sent: Monday, 21 February, 2022 11:57 To: FreeSWITCH Users Help Subject: [Freeswitch-users] call duration for bash script hello freeswitch-users, i would like to run a bash script after a call and for this script i need the duration of that call. i found the variable "billsec", but i couldn´t find any dialplan examples how to use this variable. can someone give me a hint how i can hand over the duration of the call to a bash script in the dialplan. thank you very much! best regards, thomas _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com From botelist at gmail.com Wed Feb 23 16:40:55 2022 From: botelist at gmail.com (Bote Man) Date: Wed, 23 Feb 2022 11:40:55 -0500 Subject: [Freeswitch-users] How to copy from header In-Reply-To: References: Message-ID: <007201d828d4$213aa960$63affc20$@gmail.com> Perhaps there is an answer on this wiki page: https://freeswitch.org/confluence/display/FREESWITCH/Sofia+SIP+Stack John Boteler Bote Communications From: Carlos Eduardo Sent: Wednesday, 23 February, 2022 10:14 Subject: Re: [Freeswitch-users] How to copy from header you can try copying the from header received export sip_from_uri=${sip_from_uri} Em qua., 23 de fev. de 2022 às 07:20, Dragos Oancea > escreveu: maybe reading chan var "sip_full_from" and putting its contents in a special sip "X-" header. On Tue, Feb 22, 2022 at 6:34 PM Adiseshu Channasamudhram > wrote: Hello Team As I have not got any response, trying this a different way How can I pass aniii to outgoing from header? Regards Adi Sent from my iPhone On Feb 18, 2022, at 7:23 AM, Adiseshu Channasamudhram > wrote:  Hello FS experts Can someone please help me with this? Thanks Adi Sent from my iPhone On Feb 17, 2022, at 12:05 PM, Adiseshu Channasamudhram > wrote:  Hello Team I have a requirement to copy from header from the incoming INVITE to the outgoing INVITE right now, below is the from header from incomign invite From: "123-456-7788" ;tag=2bf345ee but the outgoing INVITE has the from header as below From: "123-456-7788" >;tag=yaXFUFjD666ym How can i copy the cpc and isup-oli from the incoming from header to the outgoing header? Thanks in advance Regards Adi _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -- Carlos E. Wagner Tecnólogo em Telecomunicações, Opensips Certified Professional Fone: +55 48 99981-0894 E-mail: kaduww at gmail.com LinkedIn: https://www.linkedin.com/in/carlos-eduardo-wagner-96bbb433/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From sdevoy at bizfocused.com Wed Feb 23 19:14:18 2022 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 23 Feb 2022 19:14:18 +0000 Subject: [Freeswitch-users] PCAP help and many call failures PLEASE HELP In-Reply-To: References: Message-ID: Since about half of the calls fail and half work, could it be that the calls that are answered quicker are making it under some time limit and the others are not? What parameters should I look at in FS for outbound call answer time limit?? Thanks, Sean -----Original Message----- From: FreeSWITCH-users On Behalf Of Sean Devoy via FreeSWITCH-users Sent: Tuesday, February 22, 2022 11:57 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] PCAP help and many call failures PLEASE HELP _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com From sdevoy at bizfocused.com Wed Feb 23 19:43:44 2022 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 23 Feb 2022 19:43:44 +0000 Subject: Mitel/Astra phone UDP and NAT issue Message-ID: Hi, At a customer site behind NAT we have three brands of phones: Mitel, CISCO, and Yealink. Cisco and Yealink work like a charm. If the Mitel phone is set to UDP it does not respond to the "Not Authorized" response to its "Registration", presumably because it did not make it back through NAT. Using TCP it works. Any ideas for NAT support with Mitel? Regards, Sean Devoy VP Operations and Development Business Focused Internet Systems, Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL: From mayamatakeshi at gmail.com Wed Feb 23 22:19:15 2022 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Thu, 24 Feb 2022 07:19:15 +0900 Subject: [Freeswitch-users] PCAP help and many call failures PLEASE HELP In-Reply-To: References: Message-ID: On Thu, Feb 24, 2022 at 12:27 AM Sean Devoy wrote: > > > I have debug level FS logs. There are so many lines it is difficult to > follow a single call. > > > Do you have the logs related to the capture file? If yes, then search for the Call-IDs of the troubled call: Leg1: Call-ID: f0544e98-378154a6 at 192.168.2.111 Leg2: Call-ID: 26e3398f-0e97-123b-3b83-16f3d912a996 For example, I did a test call with Call-ID: 090abe39-188c-4f06-b371-e8ffe3d862f8 So I grepped the freeswitch.log file with: [root at lab002 ~]$ grep 090abe39-188c-4f06-b371-e8ffe3d862f8 /usr/local/freeswitch/log/freeswitch.log 4b860aaf-b481-4e26-8886-37321c2fc3f3 2022-02-24 06:49:17.131915 [INFO] sofia.c:10414 sofia/external_in/123412341234 at test.com receiving invite from 127.0.0.1:5060 version: 1.10.7-dev git dd24113 2021-08-25 17:37:19Z 64bit call-id: 090abe39-188c-4f06-b371-e8ffe3d862f8 The 4b860aaf-b481-4e26-8886-37321c2fc3f3 above is the channel uuid (the unique identifier of the channel in freeswitch). The freeswitch log file will include that channel uuid in all log lines related to that channel. So you can grep your freeswitch.log file for it. If no clue about the problem is found, do the same for the other leg. > This happens are about 50% of ALL calls. > > Here is the dial plan: > > > > > > > > > > > > > > > > sip_from_uri=${outbound_caller_id_number}@ > REDACTED/> > > > > > > > > > > > > > > > > > > {sip_from_uri=${outbound_caller_id_number}@REDACTED,sip_from_user=${outbound_caller_id_number},sip_from_host=REDACTED}sofia/gateway/voip-innovations-outbound/1$1 > <%7bsip_from_uri=$%7boutbound_caller_id_number%7d at 64.136.173.30,sip_from_user=$%7boutbound_caller_id_number%7d,sip_from_host=64.136.173.30%7dsofia/gateway/voip-innovations-outbound/1$1> > /> > > > > > > > OK. So there is nothing setting an answer timeout (so it would default to 60 seconds). The variable controlling it would be this one: https://freeswitch.org/confluence/display/FREESWITCH/call_timeout -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Wed Feb 23 22:45:14 2022 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 23 Feb 2022 22:45:14 +0000 Subject: [Freeswitch-users] call duration for bash script In-Reply-To: <007101d828d2$756339f0$6029add0$@gmail.com> References: <007101d828d2$756339f0$6029add0$@gmail.com> Message-ID: look into https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+set+zombie+exec billsec is not available until the call actually hangs up. Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Wed, Feb 23, 2022 at 4:54 PM Bote Man wrote: > I don't know a good answer to your question, but over the years I > understand that the best way to solve these kinds of problems is to read > the call detail log and parse out the desired values from there. This is > best done outside of the dialplan, so you would trigger your script to > start its processing while the dialplan completes the call. > > > John Boteler > Bote Communications > > > > -----Original Message----- > From: FreeSWITCH-users On > Behalf Of thomas peterseil > Sent: Monday, 21 February, 2022 11:57 > To: FreeSWITCH Users Help > Subject: [Freeswitch-users] call duration for bash script > > hello freeswitch-users, > i would like to run a bash script after a call and for this script i need > the duration of that call. i found the variable "billsec", but i couldn´t > find any dialplan examples how to use this variable. can someone give me a > hint how i can hand over the duration of the call to a bash script in the > dialplan. > thank you very much! > best regards, > thomas > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Thu Feb 24 00:33:27 2022 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 24 Feb 2022 00:33:27 +0000 Subject: [Freeswitch-users] PCAP help and many call failures PLEASE HELP In-Reply-To: References: Message-ID: try adding right before the bridge. Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Wed, Feb 23, 2022 at 10:46 PM mayamatakeshi wrote: > > > On Thu, Feb 24, 2022 at 12:27 AM Sean Devoy wrote: > >> >> >> I have debug level FS logs. There are so many lines it is difficult to >> follow a single call. >> >> >> > > Do you have the logs related to the capture file? > If yes, then search for the Call-IDs of the troubled call: > > Leg1: > Call-ID: f0544e98-378154a6 at 192.168.2.111 > > Leg2: > Call-ID: 26e3398f-0e97-123b-3b83-16f3d912a996 > > For example, I did a test call with > Call-ID: 090abe39-188c-4f06-b371-e8ffe3d862f8 > So I grepped the freeswitch.log file with: > > [root at lab002 ~]$ grep 090abe39-188c-4f06-b371-e8ffe3d862f8 > /usr/local/freeswitch/log/freeswitch.log > 4b860aaf-b481-4e26-8886-37321c2fc3f3 2022-02-24 06:49:17.131915 [INFO] > sofia.c:10414 sofia/external_in/123412341234 at test.com receiving invite > from 127.0.0.1:5060 version: 1.10.7-dev git dd24113 2021-08-25 17:37:19Z > 64bit call-id: 090abe39-188c-4f06-b371-e8ffe3d862f8 > > The 4b860aaf-b481-4e26-8886-37321c2fc3f3 above is the channel uuid (the > unique identifier of the channel in freeswitch). > The freeswitch log file will include that channel uuid in all log lines > related to that channel. > So you can grep your freeswitch.log file for it. > > If no clue about the problem is found, do the same for the other leg. > > >> This happens are about 50% of ALL calls. >> >> Here is the dial plan: >> >> >> >> >> >> >> >> >> >> >> >> >> >> > /> >> >> > sip_from_uri=${outbound_caller_id_number}@ >> REDACTED/> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> > {sip_from_uri=${outbound_caller_id_number}@REDACTED,sip_from_user=${outbound_caller_id_number},sip_from_host=REDACTED}sofia/gateway/voip-innovations-outbound/1$1 >> <%7bsip_from_uri=$%7boutbound_caller_id_number%7d at 64.136.173.30,sip_from_user=$%7boutbound_caller_id_number%7d,sip_from_host=64.136.173.30%7dsofia/gateway/voip-innovations-outbound/1$1> >> /> >> >> >> >> >> >> >> > > OK. So there is nothing setting an answer timeout (so it would default to > 60 seconds). > The variable controlling it would be this one: > https://freeswitch.org/confluence/display/FREESWITCH/call_timeout > > > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From andynewlands at gmail.com Mon Feb 28 14:12:56 2022 From: andynewlands at gmail.com (Andy Newlands) Date: Mon, 28 Feb 2022 14:12:56 +0000 Subject: [Freeswitch-users] Spurious DTMF characters when using RFC2833 with SRTP Message-ID: Hi, We have a reproducible problem where spurious DTMF characters are introduced on calls from Freeswitch to a remote (3rd party) IVR, but ONLY when using SRTP (not with RTP) When the spurious character is generated it is: 1. Only ever following a valid/legitimate character (one that was genuinely keyed) 2. Often has a very long duration (see below) 3. Appears to be a random character - often 0x00, but sometimes 0-9/A-F,#,* (and that's when we get problems with the IVR - FS automatically drops the 0x00 values). We have been able to reproduce this from mobile/cell phones and traditional land-lines. I added some code to: switch_rtp.c, in static handle_rfc2833_result_t handle_rfc2833(switch_rtp_t *rtp_session, switch_size_t bytes, int *do_cng), to try to address this. It logs an error (with the character hex value and duration) for DTMF characters with "long" durations (over 10000) and then discards them. As you can see, below, this is a fairly frequent occurrence (although most of the time, the character value is null, so FS drps it, anyway - but sometimes, "real" characters are introduced). 2022-02-28 14:00:53.743665 [NOTICE] switch_vpx.c:599 VPX encoder reset (WxH/BW) from 0x0/0 to 352x288/1024 2022-02-28 14:00:56.103665 [INFO] switch_channel.c:522 RECV DTMF 4:2400 2022-02-28 14:00:56.303664 [ERR] switch_channel.c:557 Ignored invalid DTMF on Call-ID: 3b5d08d3-d898-40b6-ae00-5886713ca6dd, DTMF: 0x0 Duration: 5063 2022-02-28 14:00:56.383661 [INFO] switch_channel.c:522 RECV DTMF 4:2400 2022-02-28 14:00:56.583665 [INFO] switch_channel.c:522 RECV DTMF 5:2400 2022-02-28 14:00:57.063662 [INFO] switch_channel.c:522 RECV DTMF 5:2400 2022-02-28 14:00:57.123670 [INFO] switch_channel.c:522 RECV DTMF 8:2720 2022-02-28 14:00:57.323673 [ERR] switch_rtp.c:686 Discarded long-duration DTMF on Call-ID: 3b5d08d3-d898-40b6-ae00-5886713ca6dd, DTMF: 0x0 Duration: 54088 2022-02-28 14:00:57.603671 [INFO] switch_channel.c:522 RECV DTMF 9:2400 2022-02-28 14:00:57.824102 [INFO] switch_channel.c:522 RECV DTMF 8:2720 2022-02-28 14:00:57.903663 [ERR] switch_rtp.c:686 Discarded long-duration DTMF on Call-ID: 3b5d08d3-d898-40b6-ae00-5886713ca6dd, DTMF: 0x0 Duration: 57002 2022-02-28 14:00:58.183668 [INFO] switch_channel.c:522 RECV DTMF 6:2400 2022-02-28 14:00:58.363666 [ERR] switch_rtp.c:686 Discarded long-duration DTMF on Call-ID: 3b5d08d3-d898-40b6-ae00-5886713ca6dd, DTMF: 0x35 Duration: 57073 2022-02-28 14:00:58.503667 [INFO] switch_channel.c:522 RECV DTMF 9:2400 2022-02-28 14:00:58.643667 [INFO] switch_channel.c:522 RECV DTMF 1:2400 2022-02-28 14:00:58.963663 [INFO] switch_channel.c:522 RECV DTMF 6:2400 2022-02-28 14:00:59.283665 [INFO] switch_channel.c:522 RECV DTMF 1:2400 The first spurious character is 0x00 (which FS will drop anyway) but this could easily be a valid character (and would escape detection because its duration is not too long - it's long but not unreasonable). So, this is an imperfect "fix" as we occasionally see a spurious character with a legitimate value and a "sensible" duration - sufficiently often for users to complain. The last [ERR] entry shows a '5' being introduced but this is dropped because the duration is "crazy" (57073) As mentioned, this does not happen with secure media disabled - only with SRTP. So, I am wondering if the code correctly processes a digit then, sometimes (somehow) incorrectly attempts to decode part of an audio packet as though it were RFC2833. Has anyone else experienced this? Does anyone know what may be causing the problem and what the fix might be. I can provide console output and tcp dumps if required, along with any other details that may help (I'm also comfortable making code changes - to for extra logging/diagnostics etc). Thank you. Kind regards, Andy -------------- next part -------------- An HTML attachment was scrubbed... URL: From csadi at hotmail.com Mon Feb 28 20:34:09 2022 From: csadi at hotmail.com (Adiseshu Channasamudhram) Date: Mon, 28 Feb 2022 20:34:09 +0000 Subject: [Freeswitch-users] Copying From header Message-ID: Hello FS Team, How can I copy the From header in INVITE from incoming leg to the outgoing leg? Basically i have the carrier send the From header as below [not the clean way] From: ;isup-oli=00;tag=3418741140365741_c2b08.2.4.1638259348522.0_6724417_21166610 I want to make sure i send out the oli in the outgoing INVITE From header. Note that in the incoming From , the oli is outside the uri Thanks a lot in advance Regards Adises -------------- next part -------------- An HTML attachment was scrubbed... URL: