From brian at linuxpenguins.xyz Mon Aug 1 10:47:19 2022 From: brian at linuxpenguins.xyz (Brian May) Date: Mon, 01 Aug 2022 20:47:19 +1000 Subject: [Freeswitch-users] NOT_REGISTERED error Message-ID: <87bkt4xmtk.fsf@canidae.wired.pri> Hello, I seem to be having random issues that come and go. So will deal with them one at a time. First problem, the incoming call from one extension to another randomly gets blocked. But other times it works. Logs when calling 1005 from 1004: === cut === EXECUTE [depth=0] sofia/internal-ipv6/1004 at sip.microcomaustralia.com.au bridge(user/1005 at sip.microcomaustralia.com.au) 2022-08-01 20:06:45.066758 98.33% [DEBUG] switch_channel.c:1269 sofia/internal-ipv6/1004 at sip.microcomaustralia.com.au EXPORTING[export_vars] [RFC2822_DATE]=[Mon, 01 Aug 2022 20:06:45 +1000] to event 2022-08-01 20:06:45.066758 98.33% [DEBUG] switch_channel.c:1269 sofia/internal-ipv6/1004 at sip.microcomaustralia.com.au EXPORTING[export_vars] [dialed_extension]=[1005] to event 2022-08-01 20:06:45.066758 98.33% [DEBUG] switch_ivr_originate.c:2281 Parsing global variables 2022-08-01 20:06:45.066758 98.33% [DEBUG] switch_channel.c:1269 sofia/internal-ipv6/1004 at sip.microcomaustralia.com.au EXPORTING[export_vars] [RFC2822_DATE]=[Mon, 01 Aug 2022 20:06:45 +1000] to event 2022-08-01 20:06:45.066758 98.33% [DEBUG] switch_channel.c:1269 sofia/internal-ipv6/1004 at sip.microcomaustralia.com.au EXPORTING[export_vars] [dialed_extension]=[1005] to event 2022-08-01 20:06:45.066758 98.33% [DEBUG] switch_ivr_originate.c:2281 Parsing global variables 2022-08-01 20:06:45.066758 98.33% [NOTICE] switch_channel.c:1123 New Channel sofia/internal-ipv6/1005 at 59.167.180.193:50138 [43494707-9a11-45a6-956e-c6096b48f60b] 2022-08-01 20:06:45.066758 98.33% [DEBUG] mod_sofia.c:5121 (sofia/internal-ipv6/1005 at 59.167.180.193:50138) State Change CS_NEW -> CS_INIT 2022-08-01 20:06:45.066758 98.33% [DEBUG] switch_core_state_machine.c:581 (sofia/internal-ipv6/1005 at 59.167.180.193:50138) Running State Change CS_INIT (Cur 2 Tot 34) 2022-08-01 20:06:45.066758 98.33% [NOTICE] switch_ivr_originate.c:3039 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] === cut === But the user is registered: === cut === freeswitch at huey> show registrations reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname,metadata 1000,sip.microcomaustralia.com.au,3136333931393535353535353131-bybjp0g3gomy,sofia/internal/sip:1000 at 192.168.2.97:45452;line=10p5s5g9,1659350487,192.168.2.97,45452,udp,huey, 1004,sip.microcomaustralia.com.au,5rG1JkQ51w4XzyjV7ET8ng..,sofia/internal-ipv6/sip:1004@[2001:44b8:4112:8a05:859:d448:94ef:d733]:35862;rinstance=e736892a33822d9d;transport=tcp,1659348746,2001:44b8:4112:8a05:859:d448:94ef:d733,36207,tcp,huey, 1005,sip.microcomaustralia.com.au,kJaeILXv1Nta2tQEo8AV7A..,sofia/internal-ipv6/sip:1005 at 59.167.180.193:50138;transport=TCP;rinstance=e58807bf09569293,1659349053,2001:44b8:4112:8a02:e88:d16b:f1f6:c2ed,34495,tcp,huey, 3 total. freeswitch at huey> sofia status profile internal-ipv6 reg Registrations: ================================================================================================= Call-ID: 5rG1JkQ51w4XzyjV7ET8ng.. User: 1004 at sip.microcomaustralia.com.au Contact: "" Agent: Zoiper v2.10.17.3-mod Status: Registered(TCP)(unknown) EXP(2022-08-01 20:12:26) EXPSECS(314) Ping-Status: Reachable Ping-Time: 0.00 Host: huey IP: 2001:44b8:4112:8a05:859:d448:94ef:d733 Port: 36207 Auth-User: 1004 Auth-Realm: sip.microcomaustralia.com.au MWI-Account: 1004 at sip.microcomaustralia.com.au Call-ID: kJaeILXv1Nta2tQEo8AV7A.. User: 1005 at sip.microcomaustralia.com.au Contact: "" Agent: Z 5.5.13 v2.10.18.3 Status: Registered(TCP)(unknown) EXP(2022-08-01 20:17:33) EXPSECS(621) Ping-Status: Reachable Ping-Time: 0.00 Host: huey IP: 2001:44b8:4112:8a02:e88:d16b:f1f6:c2ed Port: 34495 Auth-User: 1005 Auth-Realm: sip.microcomaustralia.com.au MWI-Account: 1005 at sip.microcomaustralia.com.au Total items returned: 2 ================================================================================================= freeswitch at huey> sofia status profile internal reg Registrations: ================================================================================================= Call-ID: 3136333931393535353535353131-bybjp0g3gomy User: 1000 at sip.microcomaustralia.com.au Contact: "Extension" Agent: snomD785/10.1.84.15 Status: Registered(UDP)(unknown) EXP(2022-08-01 21:11:27) EXPSECS(3632) Ping-Status: Reachable Ping-Time: 0.00 Host: huey IP: 192.168.2.97 Port: 45452 Auth-User: 1000 Auth-Realm: sip.microcomaustralia.com.au MWI-Account: 1000 at sip.microcomaustralia.com.au Total items returned: 1 ================================================================================================= === cut === While there are some things I don't understand here, fact is I believe sofia/internal-ipv6/1005 should be registered.... Curiously calls in the other direction - 1005 to 1004 work fine at the moment. I thought maybe the IPv6 profile was causing problems, so I deleted it, but get similar problems: === cut === EXECUTE [depth=0] sofia/internal/1004 at 59.167.180.194 bridge(user/1005 at sip.microcomaustralia.com.au) 2022-08-01 20:32:37.711523 98.03% [DEBUG] switch_channel.c:1269 sofia/internal/1004 at 59.167.180.194 EXPORTING[export_vars] [RFC2822_DATE]=[Mon, 01 Aug 2022 20:32:37 +1000] to event 2022-08-01 20:32:37.711523 98.03% [DEBUG] switch_channel.c:1269 sofia/internal/1004 at 59.167.180.194 EXPORTING[export_vars] [dialed_extension]=[1005] to event 2022-08-01 20:32:37.711523 98.03% [DEBUG] switch_ivr_originate.c:2281 Parsing global variables 2022-08-01 20:32:37.711523 98.03% [DEBUG] switch_channel.c:1269 sofia/internal/1004 at 59.167.180.194 EXPORTING[export_vars] [RFC2822_DATE]=[Mon, 01 Aug 2022 20:32:37 +1000] to event 2022-08-01 20:32:37.711523 98.03% [DEBUG] switch_channel.c:1269 sofia/internal/1004 at 59.167.180.194 EXPORTING[export_vars] [dialed_extension]=[1005] to event 2022-08-01 20:32:37.711523 98.03% [DEBUG] switch_ivr_originate.c:2281 Parsing global variables 2022-08-01 20:32:37.711523 98.03% [NOTICE] switch_channel.c:1123 New Channel sofia/internal/1005 at 59.167.180.193:50138 [e2d0ccc0-26eb-4b99-9aff-162a182435dd] 2022-08-01 20:32:37.711523 98.03% [DEBUG] mod_sofia.c:5121 (sofia/internal/1005 at 59.167.180.193:50138) State Change CS_NEW -> CS_INIT 2022-08-01 20:32:37.711523 98.03% [DEBUG] switch_core_state_machine.c:581 (sofia/internal/1005 at 59.167.180.193:50138) Running State Change CS_INIT (Cur 2 Tot 6) 2022-08-01 20:32:37.711523 98.03% [NOTICE] switch_ivr_originate.c:3039 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2022-08-01 20:32:37.711523 98.03% [DEBUG] switch_core_state_machine.c:624 (sofia/internal/1005 at 59.167.180.193:50138) State INIT 2022-08-01 20:32:37.711523 98.03% [DEBUG] mod_sofia.c:97 sofia/internal/1005 at 59.167.180.193:50138 SOFIA INIT 2022-08-01 20:32:37.711523 98.03% [INFO] sofia_glue.c:1651 sofia/internal/1005 at 59.167.180.193:50138 sending invite call-id: (null) === cut === === cut === freeswitch at huey> sofia status profile internal reg Registrations: ================================================================================================= Call-ID: 3136333931393535353535353131-bybjp0g3gomy User: 1000 at sip.microcomaustralia.com.au Contact: "Extension" Agent: snomD785/10.1.84.15 Status: Registered(UDP)(unknown) EXP(2022-08-01 21:11:27) EXPSECS(2322) Ping-Status: Reachable Ping-Time: 0.00 Host: huey IP: 192.168.2.97 Port: 45452 Auth-User: 1000 Auth-Realm: sip.microcomaustralia.com.au MWI-Account: 1000 at sip.microcomaustralia.com.au Call-ID: JStfFaD7O6YhbD5glF1O2w.. User: 1004 at sip.microcomaustralia.com.au Contact: "" Agent: Zoiper v2.10.17.3-mod Status: Registered(TCP)(unknown) EXP(2022-08-01 20:42:37) EXPSECS(592) Ping-Status: Reachable Ping-Time: 0.00 Host: huey IP: 192.168.5.69 Port: 55579 Auth-User: 1004 Auth-Realm: 59.167.180.194 MWI-Account: 1004 at sip.microcomaustralia.com.au Call-ID: pffe0efiQEXsqtVIjbxNoA.. User: 1005 at sip.microcomaustralia.com.au Contact: "" Agent: Z 5.5.13 v2.10.18.3 Status: Registered(TCP)(unknown) EXP(2022-08-01 20:43:24) EXPSECS(639) Ping-Status: Reachable Ping-Time: 0.00 Host: huey IP: 192.168.2.216 Port: 51245 Auth-User: 1005 Auth-Realm: sip.microcomaustralia.com.au MWI-Account: 1005 at sip.microcomaustralia.com.au Call-ID: CkKLFKmXc User: 1001 at sip.microcomaustralia.com.au Contact: "home" Agent: unknown Status: Registered(TCP)(unknown) EXP(2022-08-01 20:48:31) EXPSECS(946) Ping-Status: Reachable Ping-Time: 0.00 Host: huey IP: 192.168.5.213 Port: 53628 Auth-User: 1001 Auth-Realm: sip.microcomaustralia.com.au MWI-Account: 1001 at sip.microcomaustralia.com.au Total items returned: 4 ================================================================================================= freeswitch at huey> show registrations reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname,metadata 1000,sip.microcomaustralia.com.au,3136333931393535353535353131-bybjp0g3gomy,sofia/internal/sip:1000 at 192.168.2.97:45452;line=10p5s5g9,1659352287,192.168.2.97,45452,udp,huey, 1004,sip.microcomaustralia.com.au,JStfFaD7O6YhbD5glF1O2w..,sofia/internal/sip:1004 at 192.168.5.69:35862;rinstance=424e027f47c09f90;transport=tcp,1659350557,192.168.5.69,55579,tcp,huey, 1005,sip.microcomaustralia.com.au,pffe0efiQEXsqtVIjbxNoA..,sofia/internal/sip:1005 at 59.167.180.193:50138;transport=TCP;rinstance=dafd96f89850cb02,1659350604,192.168.2.216,51245,tcp,huey, 1001,sip.microcomaustralia.com.au,CkKLFKmXc,sofia/internal/sip:28256742 at 192.168.5.213:53628;transport=tcp;ob,1659350911,192.168.5.213,53628,tcp,huey, 4 total. === cut === As a result I put IPv6 back again. 1001 is an experimental client which I don't entirely trust, but don't think that is causing any issues for calls where it is not involved. Just quick summary of the some of the other issues: * Occasionally the calls just freeze. Nothing shown in freeswitch logs. No sign of any packets being dropped. Just client (zoiper) freezes and does nothing. I suspect this might be a client issue, maybe nothing to do with freeswitch. * (the real problem I am trying to investigate) outgoing PSTN calls work fine. But incoming calls from our VOIP provider - there is a 10+ second delay on all incoming audio, and I can't see any reason for this. i.e. I mean I say something on remote end, wait 10 seconds, then it is received at local end. * Turning on echo cancellation on a Linux based host running zoiper seemed to cut out audio after about 1 second. So I disabled that option, seems to have fixed that problem. Any ideas? -- Brian May https://linuxpenguins.xyz/brian/ From brian at linuxpenguins.xyz Mon Aug 1 23:20:31 2022 From: brian at linuxpenguins.xyz (Brian May) Date: Tue, 02 Aug 2022 09:20:31 +1000 Subject: [Freeswitch-users] NOT_REGISTERED error In-Reply-To: <87bkt4xmtk.fsf@canidae.wired.pri> References: <87bkt4xmtk.fsf@canidae.wired.pri> Message-ID: <878ro7y2io.fsf@canidae.wired.pri> OK, not sure I understand, but I think I resolved some issues: * The 1005 client had the STUN server settings enabled. But this is entirely within my network, without any NAT. So this setting not appropriate. Had I known better, I would have realised that the Contact field shown in the registrations was showing my external IP address, not the internal IP address which is required. The USER_NOT_REGISTERED message is a red hearing, I still get that when it is working. Weird. * Perhaps, just to make things more confusing the above only applies to IPv4, my IPv6 address was correct. Although my logs show that the 1005 was being registered to the IPv6 profile using its IPv4 address. Huh? Something to do with the client I guess (zoiper). This explains why calls worked one way, but not the other way. * For some reason setting zrtp_secure_media=false kills incoming connection handling. As in answering calls fails. Despite the fact all clients appear to have ZRTP disabled. Huh? I don't understand... * Now that I have worked out these unrelated issues, to investigate the 10 second audio problem again. Brian May writes: > Just quick summary of the some of the other issues: > > * Occasionally the calls just freeze. Nothing shown in freeswitch logs. > No sign of any packets being dropped. Just client (zoiper) freezes and > does nothing. I suspect this might be a client issue, maybe nothing to > do with freeswitch. > > * (the real problem I am trying to investigate) outgoing PSTN calls work > fine. But incoming calls from our VOIP provider - there is a 10+ > second delay on all incoming audio, and I can't see any reason for > this. i.e. I mean I say something on remote end, wait 10 seconds, then > it is received at local end. > > * Turning on echo cancellation on a Linux based host running zoiper > seemed to cut out audio after about 1 second. So I disabled that > option, seems to have fixed that problem. > > Any ideas? > -- > Brian May > https://linuxpenguins.xyz/brian/ -- Brian May https://linuxpenguins.xyz/brian/ From brian at linuxpenguins.xyz Tue Aug 2 06:48:55 2022 From: brian at linuxpenguins.xyz (Brian May) Date: Tue, 02 Aug 2022 16:48:55 +1000 Subject: [Freeswitch-users] NOT_REGISTERED error In-Reply-To: <878ro7y2io.fsf@canidae.wired.pri> References: <87bkt4xmtk.fsf@canidae.wired.pri> <878ro7y2io.fsf@canidae.wired.pri> Message-ID: <875yjbxhrc.fsf@canidae.wired.pri> I was wrong, calls still sometimes not getting through to local clients without any reason I can see. -- Brian May https://linuxpenguins.xyz/brian/ From brian at linuxpenguins.xyz Wed Aug 3 03:00:43 2022 From: brian at linuxpenguins.xyz (Brian May) Date: Wed, 03 Aug 2022 13:00:43 +1000 Subject: [Freeswitch-users] codec negotiation error Message-ID: <871qtyxc84.fsf@canidae.wired.pri> Hello, I receiving an incoming call, with the G.729 PCMA codec. I try to bridge this call to my VOIP provider. This generates the following SDP packet: === cut === v=0 o=FreeSWITCH 175841631 175841632 IN IP4 59.167.180.194 s=FreeSWITCH c=IN IP4 59.167.180.194 t=0 0 m=audio 29188 RTP/AVP 0 101 8 3 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 === cut === I think this is bad, as PCMA isn't listed in the codecs available. How do I debug this? I am assuming this might be the reason why I am not getting incoming audio (does this make sense?) Regards -- Brian May https://linuxpenguins.xyz/brian/ From brian at linuxpenguins.xyz Wed Aug 3 03:37:34 2022 From: brian at linuxpenguins.xyz (Brian May) Date: Wed, 03 Aug 2022 13:37:34 +1000 Subject: [Freeswitch-users] codec negotiation error In-Reply-To: <871qtyxc84.fsf@canidae.wired.pri> References: <871qtyxc84.fsf@canidae.wired.pri> Message-ID: <87y1w6vvy9.fsf@canidae.wired.pri> Brian May writes: > I am assuming this might be the reason why I am not getting incoming > audio (does this make sense?) Never mind, I found https://freeswitch.org/confluence/display/FREESWITCH/verbose_sdp Turned it on, works as expected, but still not getting incoming audio. Arrgh! So far everything I have looked at appears to be OK, no idea why I am having audio problems. -- Brian May https://linuxpenguins.xyz/brian/ From brian at linuxpenguins.xyz Wed Aug 3 03:41:07 2022 From: brian at linuxpenguins.xyz (Brian May) Date: Wed, 03 Aug 2022 13:41:07 +1000 Subject: [Freeswitch-users] codec negotiation error In-Reply-To: <87y1w6vvy9.fsf@canidae.wired.pri> References: <871qtyxc84.fsf@canidae.wired.pri> <87y1w6vvy9.fsf@canidae.wired.pri> Message-ID: <87v8ravvsc.fsf@canidae.wired.pri> Brian May writes: > Never mind, I found > https://freeswitch.org/confluence/display/FREESWITCH/verbose_sdp Plus https://freeswitch-users.freeswitch.narkive.com/jyFwqoeR/incompatible-destination-did-fs-not-offer-a-codec -- Brian May https://linuxpenguins.xyz/brian/ From piotr at dataandsignal.com Thu Aug 4 17:06:15 2022 From: piotr at dataandsignal.com (Piotr Gregor) Date: Thu, 4 Aug 2022 18:06:15 +0100 Subject: [Freeswitch-users] Need help to setup stir shaken in Freeswitch In-Reply-To: References: Message-ID: Hi Pete, Sure, set the "sip_stir_shaken_attest" variable on an outbound call. See: https://github.com/signalwire/freeswitch/blob/master/src/mod/endpoints/mod_sofia/sofia_glue.c#L1134 https://github.com/signalwire/freeswitch/blob/master/src/mod/endpoints/mod_sofia/mod_sofia.c#L6508 cheers, Piotr Gregor Software Engineer M: (+44) 07483 866 525 L: (+44) 01256 597 470 www: dataandsignal.com On Sat, May 21, 2022 at 2:29 AM Pete Kay wrote: > Hi > > I see that we can sign calls using AS features in freeswitch. > > Is there any way to assign different attestation levels ( A/B/C ) on a > call by call basis? > > I see there is a stir_shaken_attest variable, but is there a way to > set the different attestation level in the dial plan? > > Thanks, > Pete > > On Wed, May 12, 2021 at 12:42 AM Dragos Oancea > wrote: > > > > In FS master branch, > src/mod/endpoints/mod_sofia/test/conf/freeswitch.xml , you can see some > example settings. > > > > > > > > On Tue, May 11, 2021 at 10:23 PM Pete Kay wrote: > >> > >> Hi > >> > >> Does anyone know how to configure Freeswitch for stir shaken? > >> > >> Could you give me some info or doc how to do it? > >> > >> Thanks, > >> Pete > >> > _________________________________________________________________________ > >> > >> 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 piotr at dataandsignal.com Thu Aug 4 17:39:14 2022 From: piotr at dataandsignal.com (Piotr Gregor) Date: Thu, 4 Aug 2022 18:39:14 +0100 Subject: [Freeswitch-users] Spurious DTMF characters when using RFC2833 with SRTP In-Reply-To: References: Message-ID: Hi Andrew, It's reproducible - please then - attach or send me directly a pcap (with a=crypto line if it's SRTP) - describe call setup - describe configuration (variables set on channel) - anything else if significant and I can have a look. regards, Piotr Gregor Software Engineer M: (+44) 07483 866 525 L: (+44) 01256 597 470 www: dataandsignal.com On Tue, Apr 12, 2022 at 4:34 PM Dragos Oancea wrote: > You could open a github issue. > > On Fri, Apr 8, 2022 at 1:40 PM Andy Newlands > wrote: > >> Further to my earlier quest, and after a lot of pain, I believe I have >> figured out the cause of the issue. It lies with what appears to be a bug >> do_flush() in switch_rtp.c - this would certainly manifest as seemingly >> random/intermittent behaviour in respect of DTMF handling/processing. >> >> I have amended the code and I am currently testing my fix. >> >> Kind regards, >> >> Andy >> >> >> On Mon, 28 Feb 2022 at 14:12, Andy Newlands >> wrote: >> >>> 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 >>> >> _________________________________________________________________________ >> >> 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 piotr at dataandsignal.com Sat Aug 6 15:03:21 2022 From: piotr at dataandsignal.com (Piotr Gregor) Date: Sat, 6 Aug 2022 16:03:21 +0100 Subject: [Freeswitch-users] codec negotiation error In-Reply-To: <87v8ravvsc.fsf@canidae.wired.pri> References: <871qtyxc84.fsf@canidae.wired.pri> <87y1w6vvy9.fsf@canidae.wired.pri> <87v8ravvsc.fsf@canidae.wired.pri> Message-ID: > > m=audio 29188 RTP/AVP 0 101 8 3 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > === cut === > I think this is bad, as PCMA isn't listed in the codecs available. Oops. You know what 8 in the m=audio line stands for Brian? Yes, it is for PCMA. How do I debug this? You can start from checking if these packets are sent at all, and if you get those packets at all. Use wireshark or tcpdump. Do the same on both machines, you will see if and where RTP is sent/recved. If all RTP is sent but it cannot reach your FreeSWITCH, then check ports are allowed in the firewall. regards, Piotr Gregor Software Engineer M: (+44) 07483 866 525 L: (+44) 01256 597 470 www: dataandsignal.com On Wed, Aug 3, 2022 at 10:15 PM Brian May wrote: > Brian May writes: > > > Never mind, I found > > https://freeswitch.org/confluence/display/FREESWITCH/verbose_sdp > > Plus > > https://freeswitch-users.freeswitch.narkive.com/jyFwqoeR/incompatible-destination-did-fs-not-offer-a-codec > -- > Brian May > https://linuxpenguins.xyz/brian/ > > _________________________________________________________________________ > > 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 brian at linuxpenguins.xyz Mon Aug 8 04:32:55 2022 From: brian at linuxpenguins.xyz (Brian May) Date: Mon, 08 Aug 2022 14:32:55 +1000 Subject: [Freeswitch-users] codec negotiation error In-Reply-To: References: <871qtyxc84.fsf@canidae.wired.pri> <87y1w6vvy9.fsf@canidae.wired.pri> <87v8ravvsc.fsf@canidae.wired.pri> Message-ID: <87edxrwe14.fsf@canidae.wired.pri> Piotr Gregor writes: > You can start from checking if these packets are sent at all, and if you > get those packets at all. > Use wireshark or tcpdump. > Do the same on both machines, you will see if and where RTP is sent/recved. > If all RTP is sent but it cannot reach your FreeSWITCH, then check ports > are allowed in the firewall. But how do I know if the remote server which I don't have access to is actually sending the RTP? I don't appear to see anything. This was a big unknown. Is it sending? Is it sending to the correct address? etc. I think I have it working now. Although somewhat confused, because I tried this same setup earlier - or I thought I had - and it didn't work. My theory: * I could see outgoing RTP packets but not incoming RTP packets. * Outboard RTP packets were being blocked by my firewall (EdgeRouter), because its stateful connection tracking doesn't seem to be working anymore. * Remote server decided "seeing as I am not receiving any RTP packets I am not going to bother sending any either". Actually this bit seems a bit suspicious. * By whitelisting 1024-65535 UDP outgoing (incoming was already whitelisted), it works. Which isn't great solution, but maybe the best I can do for now. Result: It looks like incoming packets is the problem, but in actual fact outgoing packets is the problem. Also curious, when I run tcpdump on the EdgeRouter, it doesn't show any of these UDP packets, even when there must be some because I have two way audio communications. So the one really good tool I have to debug firewall issues is actually very misleading. There are some points in this theory that don't entirely add up (such as the bandwidth graphs for the Internet port on the Internet side of the firewall seemed to increase when blocked voice session in progress), but I think the important part is that I do have this working again. -- Brian May https://linuxpenguins.xyz/brian/ From brian at linuxpenguins.xyz Mon Aug 8 11:42:25 2022 From: brian at linuxpenguins.xyz (Brian May) Date: Mon, 08 Aug 2022 21:42:25 +1000 Subject: [Freeswitch-users] codec negotiation error In-Reply-To: <87edxrwe14.fsf@canidae.wired.pri> References: <871qtyxc84.fsf@canidae.wired.pri> <87y1w6vvy9.fsf@canidae.wired.pri> <87v8ravvsc.fsf@canidae.wired.pri> <87edxrwe14.fsf@canidae.wired.pri> Message-ID: <87bksvvu5a.fsf@canidae.wired.pri> Brian May writes: > But how do I know if the remote server which I don't have access to is > actually sending the RTP? I don't appear to see anything. This was a big > unknown. Is it sending? Is it sending to the correct address? etc. Arrggh. I thought it was fixed, but now the problem has come back again. I can see the outgoing audio packets, but I cannot see any incoming audio packets. Oh, OK need to use: Fair enough, the end phone is behind NAT. But now while incoming audio is working, outgoing audio is not working. Arrghhh! tcpdump seems to show 3 streams: * stream in from phone * stream out to phone * stream in from provider But no stream out to provider. How do I debug this? -- Brian May https://linuxpenguins.xyz/brian/ From brian at freeswitch.com Mon Aug 8 12:13:25 2022 From: brian at freeswitch.com (Brian West) Date: Mon, 8 Aug 2022 07:13:25 -0500 Subject: [Freeswitch-users] codec negotiation error In-Reply-To: <87bksvvu5a.fsf@canidae.wired.pri> References: <871qtyxc84.fsf@canidae.wired.pri> <87y1w6vvy9.fsf@canidae.wired.pri> <87v8ravvsc.fsf@canidae.wired.pri> <87edxrwe14.fsf@canidae.wired.pri> <87bksvvu5a.fsf@canidae.wired.pri> Message-ID: Proxy media there to allow things thru that FreeSWITCH doesn't know about the codecs, it's no longer useful and shouldn't be used as it isn't a fix for anything really. /b On Mon, Aug 8, 2022 at 6:58 AM Brian May wrote: > Brian May writes: > > > But how do I know if the remote server which I don't have access to is > > actually sending the RTP? I don't appear to see anything. This was a big > > unknown. Is it sending? Is it sending to the correct address? etc. > > Arrggh. I thought it was fixed, but now the problem has come back again. > > I can see the outgoing audio packets, but I cannot see any incoming > audio packets. > > Oh, OK need to use: > > > > Fair enough, the end phone is behind NAT. > > But now while incoming audio is working, outgoing audio is not working. > Arrghhh! > > tcpdump seems to show 3 streams: > > * stream in from phone > * stream out to phone > * stream in from provider > > But no stream out to provider. > > How do I debug this? > -- > Brian May > https://linuxpenguins.xyz/brian/ > > _________________________________________________________________________ > > 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 -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at linuxpenguins.xyz Mon Aug 8 12:27:51 2022 From: brian at linuxpenguins.xyz (Brian May) Date: Mon, 08 Aug 2022 22:27:51 +1000 Subject: [Freeswitch-users] codec negotiation error In-Reply-To: <87bksvvu5a.fsf@canidae.wired.pri> References: <871qtyxc84.fsf@canidae.wired.pri> <87y1w6vvy9.fsf@canidae.wired.pri> <87v8ravvsc.fsf@canidae.wired.pri> <87edxrwe14.fsf@canidae.wired.pri> <87bksvvu5a.fsf@canidae.wired.pri> Message-ID: <878rnyx6m0.fsf@canidae.wired.pri> > Actually, I think I was getting confused here. I don't want that. But simply: But that is the default anyway. If I use bypass_media=false, then I get no inbound audio on inbound calls. As in I don't see any packets. And the IP address we provide in the SDP packet is correct. And I have firewall rules that explicitly allow the traffic. If I use proxy_media=true, then I get no outbound audio on inbound calls. Which is weird that inboard audio works. https://freeswitch.org/confluence/display/FREESWITCH/Proxy+Media -- Brian May https://linuxpenguins.xyz/brian/ From brian at freeswitch.com Mon Aug 8 17:13:02 2022 From: brian at freeswitch.com (Brian West) Date: Mon, 8 Aug 2022 12:13:02 -0500 Subject: [Freeswitch-users] codec negotiation error In-Reply-To: <878rnyx6m0.fsf@canidae.wired.pri> References: <871qtyxc84.fsf@canidae.wired.pri> <87y1w6vvy9.fsf@canidae.wired.pri> <87v8ravvsc.fsf@canidae.wired.pri> <87edxrwe14.fsf@canidae.wired.pri> <87bksvvu5a.fsf@canidae.wired.pri> <878rnyx6m0.fsf@canidae.wired.pri> Message-ID: you probably have a nat issue, what's the topology like? What is ext-*-ip set to, and what is local-network-acl set to? On Mon, Aug 8, 2022 at 8:03 AM Brian May wrote: > > > > Actually, I think I was getting confused here. I don't want that. But > simply: > > > > But that is the default anyway. > > If I use bypass_media=false, then I get no inbound audio on inbound > calls. As in I don't see any packets. And the IP address we provide in > the SDP packet is correct. And I have firewall rules that explicitly > allow the traffic. > > If I use proxy_media=true, then I get no outbound audio on inbound > calls. Which is weird that inboard audio works. > > https://freeswitch.org/confluence/display/FREESWITCH/Proxy+Media > -- > Brian May > https://linuxpenguins.xyz/brian/ > > _________________________________________________________________________ > > 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 -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at linuxpenguins.xyz Mon Aug 8 21:47:43 2022 From: brian at linuxpenguins.xyz (Brian May) Date: Tue, 09 Aug 2022 07:47:43 +1000 Subject: [Freeswitch-users] codec negotiation error In-Reply-To: References: <871qtyxc84.fsf@canidae.wired.pri> <87y1w6vvy9.fsf@canidae.wired.pri> <87v8ravvsc.fsf@canidae.wired.pri> <87edxrwe14.fsf@canidae.wired.pri> <87bksvvu5a.fsf@canidae.wired.pri> <878rnyx6m0.fsf@canidae.wired.pri> Message-ID: <875yj2wgow.fsf@canidae.wired.pri> Brian West writes: > you probably have a nat issue, what's the topology like? What is ext-*-ip > set to, and what is local-network-acl set to? provider (internet) <--firewall--> Freeswitch (external IP) <--firewall--> Phone (internal IP) Freeswitch is outside the NAT, and looking at the packets, it is sending the correct IP addresses to the provider. All firewalls are whitelisted to send all UDP 1024-65535 through. As far as freeswitch is concerned there is no NAT. Unless I try to get the phone to talk directly to the provider. Which is something I am not even attempting to do right now. local-network-acl is set to localnet.auto If there was a codec negotiation issue, what would the symptoms be? I would expect dropped phone calls, not dropped audio. So I don't think that is what I am experiencing here. My gut feeling is that the remote provider is sending audio (wish I could prove it) which means traffic is getting blocked somewhere - most likely my firewall. So will concentrate my efforts here. Starting of by disabling SIP support in my firewall. https://community.ui.com/questions/VoIP-SIP-Phone-through-NAT-on-EdgeMax/0632ce0e-9b8d-4bf0-907f-ac55485e32e6 -- Brian May https://linuxpenguins.xyz/brian/ From brian at freeswitch.com Mon Aug 8 21:54:43 2022 From: brian at freeswitch.com (Brian West) Date: Mon, 8 Aug 2022 16:54:43 -0500 Subject: [Freeswitch-users] codec negotiation error In-Reply-To: <875yj2wgow.fsf@canidae.wired.pri> References: <871qtyxc84.fsf@canidae.wired.pri> <87y1w6vvy9.fsf@canidae.wired.pri> <87v8ravvsc.fsf@canidae.wired.pri> <87edxrwe14.fsf@canidae.wired.pri> <87bksvvu5a.fsf@canidae.wired.pri> <878rnyx6m0.fsf@canidae.wired.pri> <875yj2wgow.fsf@canidae.wired.pri> Message-ID: Focus on the NAT issue between the phone and FreeSWITCH. /b On Mon, Aug 8, 2022 at 4:47 PM Brian May wrote: > Brian West writes: > > > you probably have a nat issue, what's the topology like? What is ext-*-ip > > set to, and what is local-network-acl set to? > > provider (internet) <--firewall--> Freeswitch (external IP) <--firewall--> > Phone (internal IP) > > Freeswitch is outside the NAT, and looking at the packets, it is sending > the correct IP addresses to the provider. > > All firewalls are whitelisted to send all UDP 1024-65535 through. > > As far as freeswitch is concerned there is no NAT. Unless I try to get > the phone to talk directly to the provider. Which is something I am not > even attempting to do right now. > > local-network-acl is set to localnet.auto > > If there was a codec negotiation issue, what would the symptoms be? I > would expect dropped phone calls, not dropped audio. So I don't think > that is what I am experiencing here. > > My gut feeling is that the remote provider is sending audio (wish I > could prove it) which means traffic is getting blocked somewhere - most > likely my firewall. So will concentrate my efforts here. Starting of by > disabling SIP support in my firewall. > > https://community.ui.com/questions/VoIP-SIP-Phone-through-NAT-on-EdgeMax/0632ce0e-9b8d-4bf0-907f-ac55485e32e6 > -- > Brian May > https://linuxpenguins.xyz/brian/ > -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at linuxpenguins.xyz Tue Aug 9 00:16:47 2022 From: brian at linuxpenguins.xyz (Brian May) Date: Tue, 09 Aug 2022 10:16:47 +1000 Subject: [Freeswitch-users] codec negotiation error In-Reply-To: References: <871qtyxc84.fsf@canidae.wired.pri> <87y1w6vvy9.fsf@canidae.wired.pri> <87v8ravvsc.fsf@canidae.wired.pri> <87edxrwe14.fsf@canidae.wired.pri> <87bksvvu5a.fsf@canidae.wired.pri> <878rnyx6m0.fsf@canidae.wired.pri> <875yj2wgow.fsf@canidae.wired.pri> Message-ID: <8735e6w9sg.fsf@canidae.wired.pri> Brian West writes: > Focus on the NAT issue between the phone and FreeSWITCH. All of that seems to be working... Here is a partial tcpdump with packets to/from the server: 09:49:29.224492 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172 09:49:29.243749 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172 09:49:29.263882 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172 09:49:29.283501 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172 09:49:29.303796 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172 09:49:29.321400 IP 59.167.180.194.19289 > 103.140.134.33.21005: UDP, length 112 09:49:29.323725 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172 09:49:29.343927 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172 09:49:29.363231 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172 09:49:29.383833 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172 09:49:29.403074 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172 09:49:29.423905 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172 09:49:29.443366 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172 09:49:29.463924 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172 09:49:29.483771 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172 09:49:29.503919 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172 09:49:29.523919 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172 09:49:29.543937 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172 09:49:29.548972 IP 103.140.134.33.21005 > 59.167.180.194.19289: UDP, length 84 09:49:29.563806 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172 09:49:29.583645 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172 09:49:29.604072 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172 09:49:29.623863 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172 09:49:29.643842 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172 09:49:29.663620 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172 09:49:29.683939 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172 As you can see outgoing audio is getting sent. And it is confirmed to be received, I can hear it. But I simply am not getting the audio packets in. Periodically I do get an RTCP packet in however. As in the 84 byte packet above. This gets all the way through to my freeswitch server. This RTCP is a sender report, and seems to indicate that the sender is sending me packets, that the sender is using my correct IP address, and no firewall is blocking any packet. So how is it possible I am receiving RTCP sender reports, but I am not receiving the data packets? -- Brian May https://linuxpenguins.xyz/brian/ From brian at linuxpenguins.xyz Tue Aug 9 02:22:05 2022 From: brian at linuxpenguins.xyz (Brian May) Date: Tue, 09 Aug 2022 12:22:05 +1000 Subject: [Freeswitch-users] codec negotiation error In-Reply-To: <8735e6w9sg.fsf@canidae.wired.pri> References: <871qtyxc84.fsf@canidae.wired.pri> <87y1w6vvy9.fsf@canidae.wired.pri> <87v8ravvsc.fsf@canidae.wired.pri> <87edxrwe14.fsf@canidae.wired.pri> <87bksvvu5a.fsf@canidae.wired.pri> <878rnyx6m0.fsf@canidae.wired.pri> <875yj2wgow.fsf@canidae.wired.pri> <8735e6w9sg.fsf@canidae.wired.pri> Message-ID: <87wnbiupf6.fsf@canidae.wired.pri> OK, so some facts: * Similar problems for 2 providers. * If incoming call is bridged, it fails. * If incoming call is redirected to 9196 echo test it works. * The call to 9196 is answered immediately. But the time take to answer the call does not appear to be significant. I created tcpdumps of both cases. * In both cases, simple INVITE, TRYING, OK, ACK sequence. * In the good case, the provider starts sending data immediately after the ACK. My reasoning is: * The server must be doing something different in order for the behaviour to change of the packets it is sending. * This means the server must know which case is being tested. * This follows that we must be communicating something different to the server, depending on which case is being tested. I have opened two instances of wireshark, and comparing the two traces, I see only insignificant differences: * port numbers are different. * In the not-working case we send a Session Progress message while the phone is ringing, I think this is just a consequence of taking more time to answer the call. * In the working case, in the OK message freeswitch sends a "Accept: application/sdp" header. In the non-working case we omit the header. I find it hard to believe this could be of any consequence. All other details, codecs, IP address details, etc are identical. Oh, almost missed a difference; we send to the provider in the OK message (bad vs good calls): -P-Asserted-Identity: "Outbound Call" +P-Asserted-Identity: "61390369013" What is this header? Could the fact I am sending the wrong value be significant? My provider doesn't know anything about my 1005 extension. I am somewhat doubtful actually. -- Brian May https://linuxpenguins.xyz/brian/ From bkk at ednt.de Tue Aug 9 10:44:55 2022 From: bkk at ednt.de (Bernd Krueger-Knauber) Date: Tue, 9 Aug 2022 12:44:55 +0200 Subject: [Freeswitch-users] Freeswitch 1.10.7 no BYE to proxy In-Reply-To: References: Message-ID: <000da162-d917-696f-9497-102a5787f63d@ednt.de> Hi, we need a proxy (kamailio) in front of freeswitch. In general it works, but ... I get no BYE if the client direct connected to freeswitch hangs up. The INVITE from the proxy contains the correct Record-Route entry: INVITE sip:yyy at xxx.xxx.xx;transport=TCP SIP/2.0 Record-Route: Via: SIP/2.0/TCP proxy.address.xxx.xxx;branch=z9hG4bK4fcd.b5f0a67d3fa655e0f3b123a8e95ff5b5.0;i=2 Via: SIP/2.0/TCP phone.address.xxx.xxx:53558;received=phone.address.xxx.xxx;rport=53558;branch=z9hG4bKPj30dd14b14974bcb42dc24d2f650f;alias Any ideas? From gilles at sauvaire.com Tue Aug 9 12:30:08 2022 From: gilles at sauvaire.com (Gilles SAUVAIRE) Date: Tue, 9 Aug 2022 14:30:08 +0200 Subject: [Freeswitch-users] launch an external program in dialplan Message-ID: Hi there, I need inside a dialplan, to launch an external program, and to take the return of this program to use it in the dialplan. I think a LUA script can do it? If anyone has an example script, I'm interested. Or is another technique better? Launching an external program at each call can be very resource-intensive, maybe make a request on a specific web service port, or something else? the idea is to add the user's name in the from. (but this name changes depending on the external context, it must be calculated, I cannot make a simple database query) someone has already done it I guess I'm not the first to have this need... Thank you all... Gilles -------------- next part -------------- An HTML attachment was scrubbed... URL: From gregor at infomedia.si Tue Aug 9 12:40:28 2022 From: gregor at infomedia.si (Gregor Nanger) Date: Tue, 9 Aug 2022 14:40:28 +0200 Subject: [Freeswitch-users] launch an external program in dialplan In-Reply-To: References: Message-ID: I think Httapi would be useful in your scenario. Especially if you need to use the response from an external URL and use it in dialplan. On Tue, 9 Aug 2022 at 14:31, Gilles SAUVAIRE wrote: > Hi there, > > > > I need inside a dialplan, to launch an external program, and to take the > return of this program to use it in the dialplan. > > > > I think a LUA script can do it? > > > > If anyone has an example script, I'm interested. > > Or is another technique better? > > Launching an external program at each call can be very resource-intensive, > maybe make a request on a specific web service port, or something else? > > > > the idea is to add the user's name in the from. > > (but this name changes depending on the external context, it must be > calculated, I cannot make a simple database query) > > > > someone has already done it I guess I'm not the first to have this need... > > > > Thank you all... > > > > Gilles > > > > > > > _________________________________________________________________________ > > 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 -- Gregor Nanger *CTO* t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia • www.infomedia.si -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Tue Aug 9 12:47:26 2022 From: brian at freeswitch.com (Brian West) Date: Tue, 9 Aug 2022 07:47:26 -0500 Subject: [Freeswitch-users] codec negotiation error In-Reply-To: <87wnbiupf6.fsf@canidae.wired.pri> References: <871qtyxc84.fsf@canidae.wired.pri> <87y1w6vvy9.fsf@canidae.wired.pri> <87v8ravvsc.fsf@canidae.wired.pri> <87edxrwe14.fsf@canidae.wired.pri> <87bksvvu5a.fsf@canidae.wired.pri> <878rnyx6m0.fsf@canidae.wired.pri> <875yj2wgow.fsf@canidae.wired.pri> <8735e6w9sg.fsf@canidae.wired.pri> <87wnbiupf6.fsf@canidae.wired.pri> Message-ID: Sounds like you need to setup outbound caller ID, do you have a full sip trace? On Mon, Aug 8, 2022 at 9:22 PM Brian May wrote: > OK, so some facts: > > * Similar problems for 2 providers. > * If incoming call is bridged, it fails. > * If incoming call is redirected to 9196 echo test it works. > * The call to 9196 is answered immediately. But the time take to answer > the call does not appear to be significant. > > I created tcpdumps of both cases. > > * In both cases, simple INVITE, TRYING, OK, ACK sequence. > * In the good case, the provider starts sending data immediately after the > ACK. > > My reasoning is: > > * The server must be doing something different in order for the > behaviour to change of the packets it is sending. > > * This means the server must know which case is being tested. > > * This follows that we must be communicating something different to the > server, depending on which case is being tested. > > I have opened two instances of wireshark, and comparing the two traces, > I see only insignificant differences: > > * port numbers are different. > > * In the not-working case we send a Session Progress message while the > phone is ringing, I think this is just a consequence of taking more > time to answer the call. > > * In the working case, in the OK message freeswitch sends a "Accept: > application/sdp" header. > In the non-working case we omit the header. > I find it hard to believe this could be of any consequence. > > All other details, codecs, IP address details, etc are identical. > > > Oh, almost missed a difference; we send to the provider in the OK > message (bad vs good calls): > > -P-Asserted-Identity: "Outbound Call" > +P-Asserted-Identity: "61390369013" > > What is this header? Could the fact I am sending the wrong value be > significant? My provider doesn't know anything about my 1005 extension. > > I am somewhat doubtful actually. > > -- > Brian May > https://linuxpenguins.xyz/brian/ > -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Tue Aug 9 13:24:32 2022 From: brian at freeswitch.com (Brian West) Date: Tue, 9 Aug 2022 08:24:32 -0500 Subject: [Freeswitch-users] Freeswitch 1.10.7 no BYE to proxy In-Reply-To: <000da162-d917-696f-9497-102a5787f63d@ednt.de> References: <000da162-d917-696f-9497-102a5787f63d@ednt.de> Message-ID: Pay close attention to the record route and path on the original request. Do you have a full trace? On Tue, Aug 9, 2022 at 6:02 AM Bernd Krueger-Knauber wrote: > Hi, > > we need a proxy (kamailio) in front of freeswitch. > In general it works, but ... > I get no BYE if the client direct connected to freeswitch hangs up. > > The INVITE from the proxy contains the correct Record-Route entry: > > INVITE sip:yyy at xxx.xxx.xx;transport=TCP SIP/2.0 > Record-Route: > Via: SIP/2.0/TCP > > proxy.address.xxx.xxx;branch=z9hG4bK4fcd.b5f0a67d3fa655e0f3b123a8e95ff5b5.0;i=2 > Via: SIP/2.0/TCP > > phone.address.xxx.xxx:53558;received=phone.address.xxx.xxx;rport=53558;branch=z9hG4bKPj30dd14b14974bcb42dc24d2f650f;alias > > Any ideas? > > > > _________________________________________________________________________ > > 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 -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From jasonh at thinksimplicity.com Tue Aug 9 13:35:00 2022 From: jasonh at thinksimplicity.com (=?UTF-8?Q?Jason_Holden?=) Date: Tue, 9 Aug 2022 13:35:00 +0000 Subject: [Freeswitch-users] problem with Grandstream FXO gateway References: Message-ID: <0100018282d0d62d-45b44d36-c4b1-4e84-8aa5-494c57976628-000000@email.amazonses.com> All, Any suggestions on the following? We have a Grandstream FXO gateway where a call is sent to and I receive a 180 rinigng followed by a 200ok message but the line plays back an audio busy message. Any ideas how to force a fail over / continueation in the dial plan in this situation?     Jason Holden   Phone: 1-866-836-9198 X405 Direct: 7868009949 www.thinksimplicity.com     -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 1004 bytes Desc: not available URL: From avi at avimarcus.net Tue Aug 9 15:34:05 2022 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 9 Aug 2022 15:34:05 +0000 Subject: [Freeswitch-users] launch an external program in dialplan In-Reply-To: References: Message-ID: <01000182833ddb9e-d5dd7b38-38c8-4e66-a3cb-bd141fa50339-000000@email.amazonses.com> You can make a curl call or a system call directly in the dial plan. You can even get the results in a channel variable for further processing. On Tue, Aug 9, 2022, 3:30 PM Gilles SAUVAIRE wrote: > Hi there, > > > > I need inside a dialplan, to launch an external program, and to take the > return of this program to use it in the dialplan. > > > > I think a LUA script can do it? > > > > If anyone has an example script, I'm interested. > > Or is another technique better? > > Launching an external program at each call can be very resource-intensive, > maybe make a request on a specific web service port, or something else? > > > > the idea is to add the user's name in the from. > > (but this name changes depending on the external context, it must be > calculated, I cannot make a simple database query) > > > > someone has already done it I guess I'm not the first to have this need... > > > > Thank you all... > > > > Gilles > > > > > > > _________________________________________________________________________ > > 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 bkk at ednt.de Tue Aug 9 15:48:08 2022 From: bkk at ednt.de (Bernd Krueger-Knauber) Date: Tue, 9 Aug 2022 17:48:08 +0200 Subject: [Freeswitch-users] Freeswitch 1.10.7 no BYE to proxy In-Reply-To: References: <000da162-d917-696f-9497-102a5787f63d@ednt.de> Message-ID: <86c6dd85-f2c9-76f0-e78d-4540ed0cb633@ednt.de> Hi Brian, thank you for your fast response. It is not a FreeSwitch problem! I solved it in the meanwhile. It was a firewall rule problem. FreeSwitch was not able to reach the proxy server. Strange fault, since I was able to make calls. Problem is solved. Thank you! Am 09.08.2022 um 15:24 schrieb Brian West: > Pay close attention to the record route and path on the original > request.  Do you have a full trace? > > On Tue, Aug 9, 2022 at 6:02 AM Bernd Krueger-Knauber wrote: > > Hi, > > we need a proxy (kamailio) in front of freeswitch. > In general it works, but ... > I get no BYE if the client direct connected to freeswitch hangs up. > > The INVITE from the proxy contains the correct Record-Route entry: > > INVITE sip:yyy at xxx.xxx.xx;transport=TCP SIP/2.0 > Record-Route: > Via: SIP/2.0/TCP > proxy.address.xxx.xxx;branch=z9hG4bK4fcd.b5f0a67d3fa655e0f3b123a8e95ff5b5.0;i=2 > Via: SIP/2.0/TCP > phone.address.xxx.xxx:53558;received=phone.address.xxx.xxx;rport=53558;branch=z9hG4bKPj30dd14b14974bcb42dc24d2f650f;alias > > Any ideas? > > > > _________________________________________________________________________ > > 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 > > > > -- > > Brian West | Co-founder and Developer > > Need Commercial support? email sales at freeswitch.com > > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > > > Email: brian at freeswitch.com > > Mobile: 918-424-9378 > > Website: https://www.FreeSWITCH.com > > https://www.facebook.com/signalwireinc?src=email > https://twitter.com/freeswitch > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Tue Aug 9 20:46:45 2022 From: brian at freeswitch.com (Brian West) Date: Tue, 9 Aug 2022 15:46:45 -0500 Subject: [Freeswitch-users] Freeswitch 1.10.7 no BYE to proxy In-Reply-To: <86c6dd85-f2c9-76f0-e78d-4540ed0cb633@ednt.de> References: <000da162-d917-696f-9497-102a5787f63d@ednt.de> <86c6dd85-f2c9-76f0-e78d-4540ed0cb633@ednt.de> Message-ID: Glad to hear it! On Tue, Aug 9, 2022 at 10:48 AM Bernd Krueger-Knauber wrote: > Hi Brian, > > thank you for your fast response. > It is not a FreeSwitch problem! > > I solved it in the meanwhile. > It was a firewall rule problem. > FreeSwitch was not able to reach the proxy server. > Strange fault, since I was able to make calls. > > Problem is solved. > > Thank you! > > > Am 09.08.2022 um 15:24 schrieb Brian West: > > Pay close attention to the record route and path on the original request. > Do you have a full trace? > > On Tue, Aug 9, 2022 at 6:02 AM Bernd Krueger-Knauber wrote: > >> Hi, >> >> we need a proxy (kamailio) in front of freeswitch. >> In general it works, but ... >> I get no BYE if the client direct connected to freeswitch hangs up. >> >> The INVITE from the proxy contains the correct Record-Route entry: >> >> INVITE sip:yyy at xxx.xxx.xx;transport=TCP SIP/2.0 >> Record-Route: >> Via: SIP/2.0/TCP >> >> proxy.address.xxx.xxx;branch=z9hG4bK4fcd.b5f0a67d3fa655e0f3b123a8e95ff5b5.0;i=2 >> Via: SIP/2.0/TCP >> >> phone.address.xxx.xxx:53558;received=phone.address.xxx.xxx;rport=53558;branch=z9hG4bKPj30dd14b14974bcb42dc24d2f650f;alias >> >> Any ideas? >> >> >> >> _________________________________________________________________________ >> >> 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 > > > > -- > > Brian West | Co-founder and Developer > > Need Commercial support? email sales at freeswitch.com > > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > > > Email: brian at freeswitch.com > > Mobile: 918-424-9378 > > Website: https://www.FreeSWITCH.com > > [image: https://www.facebook.com/signalwireinc?src=email] > [image: > https://twitter.com/freeswitch] > > > -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Tue Aug 9 20:47:21 2022 From: brian at freeswitch.com (Brian West) Date: Tue, 9 Aug 2022 15:47:21 -0500 Subject: [Freeswitch-users] launch an external program in dialplan In-Reply-To: <01000182833ddb9e-d5dd7b38-38c8-4e66-a3cb-bd141fa50339-000000@email.amazonses.com> References: <01000182833ddb9e-d5dd7b38-38c8-4e66-a3cb-bd141fa50339-000000@email.amazonses.com> Message-ID: you can also use the 'system' app. On Tue, Aug 9, 2022 at 11:08 AM Avi Marcus wrote: > You can make a curl call or a system call directly in the dial plan. > > You can even get the results in a channel variable for further processing. > > On Tue, Aug 9, 2022, 3:30 PM Gilles SAUVAIRE wrote: > >> Hi there, >> >> >> >> I need inside a dialplan, to launch an external program, and to take the >> return of this program to use it in the dialplan. >> >> >> >> I think a LUA script can do it? >> >> >> >> If anyone has an example script, I'm interested. >> >> Or is another technique better? >> >> Launching an external program at each call can be very >> resource-intensive, maybe make a request on a specific web service port, or >> something else? >> >> >> >> the idea is to add the user's name in the from. >> >> (but this name changes depending on the external context, it must be >> calculated, I cannot make a simple database query) >> >> >> >> someone has already done it I guess I'm not the first to have this need... >> >> >> >> Thank you all... >> >> >> >> Gilles >> >> >> >> >> >> >> _________________________________________________________________________ >> >> 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 -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at linuxpenguins.xyz Tue Aug 9 22:04:07 2022 From: brian at linuxpenguins.xyz (Brian May) Date: Wed, 10 Aug 2022 08:04:07 +1000 Subject: [Freeswitch-users] codec negotiation error In-Reply-To: References: <871qtyxc84.fsf@canidae.wired.pri> <87y1w6vvy9.fsf@canidae.wired.pri> <87v8ravvsc.fsf@canidae.wired.pri> <87edxrwe14.fsf@canidae.wired.pri> <87bksvvu5a.fsf@canidae.wired.pri> <878rnyx6m0.fsf@canidae.wired.pri> <875yj2wgow.fsf@canidae.wired.pri> <8735e6w9sg.fsf@canidae.wired.pri> <87wnbiupf6.fsf@canidae.wired.pri> Message-ID: <87r11pul9k.fsf@canidae.wired.pri> Brian West writes: > Sounds like you need to setup outbound caller ID, do you have a full sip > trace? Have a look at both traces here: https://gist.github.com/brianmay/2ec0f404b901daf5a9e763aac0989cfe I really thing the key to this problem must be somewhere here. Although to my eyes everything looks OK. Note there are two traces, one from my mobile to the audio repeat service (9196) which worked, and one from my mobile to an extension (1005) which didn't work. Personally I find it hard to believe that sending the wrong ID in response to the final OK message could cause loss of audio. This isn't the outbound caller ID that was transmitted by the caller (my provider) to the callee (freeswitch) on the invite, this is the callee (freeswitch) telling my provider who they just called. But if you think this could cause problems, and call tell me how to change it (everything I see if for the outbound caller id), then I can try to change it. -- Brian May https://linuxpenguins.xyz/brian/ From brian at freeswitch.com Tue Aug 9 22:27:05 2022 From: brian at freeswitch.com (Brian West) Date: Tue, 9 Aug 2022 17:27:05 -0500 Subject: [Freeswitch-users] codec negotiation error In-Reply-To: <87r11pul9k.fsf@canidae.wired.pri> References: <871qtyxc84.fsf@canidae.wired.pri> <87y1w6vvy9.fsf@canidae.wired.pri> <87v8ravvsc.fsf@canidae.wired.pri> <87edxrwe14.fsf@canidae.wired.pri> <87bksvvu5a.fsf@canidae.wired.pri> <878rnyx6m0.fsf@canidae.wired.pri> <875yj2wgow.fsf@canidae.wired.pri> <8735e6w9sg.fsf@canidae.wired.pri> <87wnbiupf6.fsf@canidae.wired.pri> <87r11pul9k.fsf@canidae.wired.pri> Message-ID: Good Ole CrazySwitch there, Do you happen to have logs from it's side? Also try setting extension-in-contact on the gateway. On Tue, Aug 9, 2022 at 5:04 PM Brian May wrote: > Brian West writes: > > > Sounds like you need to setup outbound caller ID, do you have a full sip > > trace? > > Have a look at both traces here: > > https://gist.github.com/brianmay/2ec0f404b901daf5a9e763aac0989cfe > > I really thing the key to this problem must be somewhere here. Although > to my eyes everything looks OK. > > Note there are two traces, one from my mobile to the audio repeat > service (9196) which worked, and one from my mobile to an extension > (1005) which didn't work. > > Personally I find it hard to believe that sending the wrong ID in > response to the final OK message could cause loss of audio. > > This isn't the outbound caller ID that was transmitted by the caller (my > provider) to the callee (freeswitch) on the invite, this is the callee > (freeswitch) telling my provider who they just called. > > But if you think this could cause problems, and call tell me how to > change it (everything I see if for the outbound caller id), then I can > try to change it. > -- > Brian May > https://linuxpenguins.xyz/brian/ > -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at linuxpenguins.xyz Tue Aug 9 22:51:44 2022 From: brian at linuxpenguins.xyz (Brian May) Date: Wed, 10 Aug 2022 08:51:44 +1000 Subject: [Freeswitch-users] codec negotiation error In-Reply-To: References: <871qtyxc84.fsf@canidae.wired.pri> <87y1w6vvy9.fsf@canidae.wired.pri> <87v8ravvsc.fsf@canidae.wired.pri> <87edxrwe14.fsf@canidae.wired.pri> <87bksvvu5a.fsf@canidae.wired.pri> <878rnyx6m0.fsf@canidae.wired.pri> <875yj2wgow.fsf@canidae.wired.pri> <8735e6w9sg.fsf@canidae.wired.pri> <87wnbiupf6.fsf@canidae.wired.pri> <87r11pul9k.fsf@canidae.wired.pri> Message-ID: <87o7wtuj27.fsf@canidae.wired.pri> Brian West writes: > Good Ole CrazySwitch there, Do you happen to have logs from it's side? No :-( They said that they would check if they are sending outgoing audio, but haven't got back to me yet. My gut feeling though, is that there is nothing blocking any UDP packets anywhere between them and me, the fact I receive RTCP status packets from them and audio packets from them for the echo test I think proves this. I also tried sending test UDP packets to myself. So if I am not receiving audio it is because they are not sending it. > Also try setting extension-in-contact on the gateway. I don't observe any change after doing that. -- Brian May https://linuxpenguins.xyz/brian/ From gilles at sauvaire.com Wed Aug 10 05:48:44 2022 From: gilles at sauvaire.com (Gilles SAUVAIRE) Date: Wed, 10 Aug 2022 07:48:44 +0200 Subject: [Freeswitch-users] launch an external program in dialplan In-Reply-To: <01000182833ddb9e-d5dd7b38-38c8-4e66-a3cb-bd141fa50339-000000@email.amazonses.com> References: <01000182833ddb9e-d5dd7b38-38c8-4e66-a3cb-bd141fa50339-000000@email.amazonses.com> Message-ID: Thanks. You ave a exemple ? With get the results in a channel variable for further processing. ? Thanks De : FreeSWITCH-users De la part de Avi Marcus Envoyé : mardi 9 août 2022 17:34 À : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] launch an external program in dialplan You can make a curl call or a system call directly in the dial plan. You can even get the results in a channel variable for further processing. On Tue, Aug 9, 2022, 3:30 PM Gilles SAUVAIRE > wrote: Hi there, I need inside a dialplan, to launch an external program, and to take the return of this program to use it in the dialplan. I think a LUA script can do it? If anyone has an example script, I'm interested. Or is another technique better? Launching an external program at each call can be very resource-intensive, maybe make a request on a specific web service port, or something else? the idea is to add the user's name in the from. (but this name changes depending on the external context, it must be calculated, I cannot make a simple database query) someone has already done it I guess I'm not the first to have this need... Thank you all... Gilles _________________________________________________________________________ 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 krice at freeswitch.org Wed Aug 10 06:37:38 2022 From: krice at freeswitch.org (krice at freeswitch.org) Date: Wed, 10 Aug 2022 01:37:38 -0500 Subject: [Freeswitch-users] launch an external program in dialplan In-Reply-To: References: <01000182833ddb9e-d5dd7b38-38c8-4e66-a3cb-bd141fa50339-000000@email.amazonses.com> Message-ID: <064401d8ac83$afa06f90$0ee14eb0$@freeswitch.org> You could always just write your own dialplan module… theres several examples in the code base of the using that C api From: FreeSWITCH-users On Behalf Of Gilles SAUVAIRE Sent: Wednesday, August 10, 2022 12:49 AM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] launch an external program in dialplan Importance: High Thanks. You ave a exemple ? With get the results in a channel variable for further processing. ? Thanks De : FreeSWITCH-users > De la part de Avi Marcus Envoyé : mardi 9 août 2022 17:34 À : FreeSWITCH Users Help > Objet : Re: [Freeswitch-users] launch an external program in dialplan You can make a curl call or a system call directly in the dial plan. You can even get the results in a channel variable for further processing. On Tue, Aug 9, 2022, 3:30 PM Gilles SAUVAIRE > wrote: Hi there, I need inside a dialplan, to launch an external program, and to take the return of this program to use it in the dialplan. I think a LUA script can do it? If anyone has an example script, I'm interested. Or is another technique better? Launching an external program at each call can be very resource-intensive, maybe make a request on a specific web service port, or something else? the idea is to add the user's name in the from. (but this name changes depending on the external context, it must be calculated, I cannot make a simple database query) someone has already done it I guess I'm not the first to have this need... Thank you all... Gilles _________________________________________________________________________ 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 brian at linuxpenguins.xyz Wed Aug 10 06:56:24 2022 From: brian at linuxpenguins.xyz (Brian May) Date: Wed, 10 Aug 2022 16:56:24 +1000 Subject: [Freeswitch-users] codec negotiation error In-Reply-To: <87o7wtuj27.fsf@canidae.wired.pri> References: <871qtyxc84.fsf@canidae.wired.pri> <87y1w6vvy9.fsf@canidae.wired.pri> <87v8ravvsc.fsf@canidae.wired.pri> <87edxrwe14.fsf@canidae.wired.pri> <87bksvvu5a.fsf@canidae.wired.pri> <878rnyx6m0.fsf@canidae.wired.pri> <875yj2wgow.fsf@canidae.wired.pri> <8735e6w9sg.fsf@canidae.wired.pri> <87wnbiupf6.fsf@canidae.wired.pri> <87r11pul9k.fsf@canidae.wired.pri> <87o7wtuj27.fsf@canidae.wired.pri> Message-ID: <87fsi4vb6v.fsf@canidae.wired.pri> Arggh! After wasting hours on this, I *think* I finally found the problem: My VDSL modem! Vigor130. It has a setting "Data Filter" which was enabled. I don't need any sort of firewall, as this is plugged straight into my EdgeRouter which does that. So now disabled both "Data Filter" and "Call Filter". It looks like it should block "TCP/UDP, Port: from 137~139 to 53". So it shouldn't affect VOIP in anyway. So I didn't worry about testing it. Turned this setting off, and I can consistently make incoming VOIP calls. Fingers crossed. (well once I tested it correctly, put on the correct headset, etc) Also "Accept large incoming fragmented UDP or ICMP packets (used in some games and streaming)" is on and "Enable Strict Security Firewall" doesn't seem to make any difference, so I left that on also. I really don't understand why non-bridged calls worked fine. But don't particularly care now. Now I suspect that the SIP connection tracking will work fine in my firewall, so my gradually ease back into that. Although I have a suspicion this will not work correctly with TLS connections. -- Brian May https://linuxpenguins.xyz/brian/ From matthew at brightfire.net Mon Aug 15 19:49:38 2022 From: matthew at brightfire.net (Matthew Grooms) Date: Mon, 15 Aug 2022 14:49:38 -0500 Subject: [Freeswitch-users] freeswitch segfault in mod_cdr_pg_csv Message-ID: <221c5a58-bd94-4c61-0381-390c7fa67787@brightfire.net> Hey Everyone, I've been seeing freeswitch crash a bit in production recently and finally had a chance to look into the issue. Appears to be coming from mod_cdr_pg_csv ... $ /usr/local/freeswitch/bin/fs_cli -x "version" FreeSWITCH Version 1.10.7-release~64bit ( 64bit) 2022-08-15 12:00:02.033897 64.60% [CRIT] mod_cdr_pg_csv.c:274 INSERT command failed: connection not open Program terminated with signal SIGABRT, Aborted. #0  0x00007f0ec45cfa9f in raise () from /lib64/libc.so.6 [Current thread is 1 (Thread 0x7f0db4adc700 (LWP 1203217))] Missing separate debuginfos, use: ... (gdb) bt #0  0x00007f0ec45cfa9f in raise () from /lib64/libc.so.6 #1  0x00007f0ec45a2e05 in abort () from /lib64/libc.so.6 #2  0x00007f0ec4612037 in __libc_message () from /lib64/libc.so.6 #3  0x00007f0ec461919c in malloc_printerr () from /lib64/libc.so.6 #4  0x00007f0ec461af40 in _int_free () from /lib64/libc.so.6 #5  0x00007f0ec7248936 in freePGconn () from /lib64/libpq.so.5 #6  0x00007f0eba7ec8d0 in insert_cdr (     values=0x7f0dd41cd4a0 "'10.16.84.32','SubjectWell','+16193300768','+16199188365','default','2022-08-15 11:59:32',null,'2022-08-15 12:00:02',30,0,'NO_ANSWER','6fd81787-c290-430d-b67b-ebfafe9f29d5',null,null,'PCMU','PCMU','se"...)     at mod_cdr_pg_csv.c:289 #7  my_on_reporting (session=0x7f0df80b1668) at mod_cdr_pg_csv.c:385 #8  0x00007f0ec75221cd in switch_core_session_reporting_state (session=session at entry=0x7f0df80b1668) at src/switch_core_state_machine.c:932 #9  0x00007f0ec7522b68 in switch_core_session_run (session=0x7f0df80b1668) at src/switch_core_state_machine.c:606 #10 0x00007f0ec751d1be in switch_core_session_thread (thread=, obj=0x7f0df80b1668) at src/switch_core_session.c:1736 #11 0x00007f0ec751891b in switch_core_session_thread_pool_worker (thread=0x7f0dc8759bd0, obj=) at src/switch_core_session.c:1800 #12 0x00007f0ec786658c in dummy_worker (opaque=0x7f0dc8759bd0) at threadproc/unix/thread.c:151 #13 0x00007f0ec50cb1cf in start_thread () from /lib64/libpthread.so.0 #14 0x00007f0ec45badd3 in clone () from /lib64/libc.so.6 (gdb) frame 6 #6  0x00007f0eba7ec8d0 in insert_cdr (     values=0x7f0dd41cd4a0 "'10.16.84.32','SubjectWell','+16193300768','+16199188365','default','2022-08-15 11:59:32',null,'2022-08-15 12:00:02',30,0,'NO_ANSWER','6fd81787-c290-430d-b67b-ebfafe9f29d5',null,null,'PCMU','PCMU','se"...)     at mod_cdr_pg_csv.c:289 289             PQfinish(globals.db_connection); I'm going to try this in production unless someone else has a better idea on how to fix this ... --- mod_cdr_pg_csv.c.orig       2022-08-15 14:44:56.028404623 -0500 +++ mod_cdr_pg_csv.c    2022-08-15 14:45:55.122482570 -0500 @@ -286,7 +286,9 @@    error: -       PQfinish(globals.db_connection); +       if (PQstatus(globals.db_connection) == CONNECTION_OK) { +               PQfinish(globals.db_connection); +       }         globals.db_online = 0;         switch_mutex_unlock(globals.db_mutex); Thanks, -Matthew From dodu at hotmail.co.uk Thu Aug 18 13:53:07 2022 From: dodu at hotmail.co.uk (Josh H) Date: Thu, 18 Aug 2022 13:53:07 +0000 Subject: [Freeswitch-users] VP8 conference FIR Signal not always sent when frames are dropped Message-ID: Hi, I am using Verto to connect to FreeSwitch over webRTC in a conference call. This works fine most of the time however when using an older device that does not support hardware acceleration for the video encoding, it appears that some frames are dropped. The CPU is pegged at 100% utilisation, with 60-80% of that being from the chrome process. This is understandable when using older hardware however when this occurs FreeSwitch fails to send FIR packets to get a new keyframe, leading to the ‘video mute’ image being shown until the next scheduled keyframe (which with VP8 is, by default, around 3000 frames, equivalent to 100 seconds at 30fps). The client seems to be unaware of any issues as it continues to encode frames and send them over the network, leading me to believe it is an issue with FreeSwitch. My current solution is to enable the ‘kf-max-dist’ parameter in ‘vpx.conf.xml’ however this leads to occasional stuttering, I assume this is because it must encode a full frame every 10 seconds. The other solution I have tested is swapping to the H.264 codec (the only other codec supported by webRTC) which makes the issue less noticeable however the quality of the stream drops significantly when no FIR is requested. H.264 isn’t an option for me as some of my target clients are Android devices which do not support H.264 over webRTC. Given that sometimes the session works as expected with FIR packets being sent when needed, leading to a small video stutter before it recovers, I am unsure how to fix it or what logs etc to provide. Any pointers would be appreciated 😊 Thanks Josh -------------- next part -------------- An HTML attachment was scrubbed... URL: From jasonh at thinksimplicity.com Tue Aug 23 16:56:35 2022 From: jasonh at thinksimplicity.com (=?UTF-8?Q?Jason_Holden?=) Date: Tue, 23 Aug 2022 16:56:35 +0000 Subject: [Freeswitch-users] freeswitch 1.8 and 1.10 Valet parking BLF problem References: Message-ID: <01000182cba26b12-aa11051d-6e28-4358-8e8e-e6c59a0a9c30-000000@email.amazonses.com> All, Trying to monitor parking slots for valet park on Freeswitch. Using Polycoms or Voice Operator Pannal regular extension BLF works fine but I do not see the BLF status for the parking slots if a call is on hold their. Does anyone have any recommendations? Not finding much online about this.     Jason Holden   Phone: 1-866-836-9198 X405 Direct: 7868009949 www.thinksimplicity.com     -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 1004 bytes Desc: not available URL: From brian at freeswitch.com Tue Aug 23 20:29:59 2022 From: brian at freeswitch.com (Brian West) Date: Tue, 23 Aug 2022 15:29:59 -0500 Subject: [Freeswitch-users] Welcome! Message-ID: FreeSWITCHers, We've fixed the reverse DNS for lists.freeswitch.org, our RIR has removed our delegation multiple times, I think we've finally fixed the process that triggered it, please let me know if you experience any issues with this mailing list moving forward and again we apologize for the inconvenience. Thanks, Brian -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From tahir at ictinnovations.com Wed Aug 24 14:46:14 2022 From: tahir at ictinnovations.com (Tahir Almas Dhesi) Date: Wed, 24 Aug 2022 19:46:14 +0500 Subject: [Freeswitch-users] Welcome! In-Reply-To: References: Message-ID: Appreciate your efforts Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT On Wed, Aug 24, 2022 at 1:30 AM Brian West wrote: > FreeSWITCHers, > > We've fixed the reverse DNS for lists.freeswitch.org, our RIR has removed > our delegation multiple times, I think we've finally fixed the process that > triggered it, please let me know if you experience any issues with this > mailing list moving forward and again we apologize for the inconvenience. > > Thanks, > Brian > > > -- > > Brian West | Co-founder and Developer > > Need Commercial support? email sales at freeswitch.com > > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > > > Email: brian at freeswitch.com > > Mobile: 918-424-9378 > > Website: https://www.FreeSWITCH.com > > [image: https://www.facebook.com/signalwireinc?src=email] > [image: > https://twitter.com/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 brian at freeswitch.com Wed Aug 24 15:40:07 2022 From: brian at freeswitch.com (Brian West) Date: Wed, 24 Aug 2022 10:40:07 -0500 Subject: [Freeswitch-users] freeswitch 1.8 and 1.10 Valet parking BLF problem In-Reply-To: <01000182cba26b12-aa11051d-6e28-4358-8e8e-e6c59a0a9c30-000000@email.amazonses.com> References: <01000182cba26b12-aa11051d-6e28-4358-8e8e-e6c59a0a9c30-000000@email.amazonses.com> Message-ID: Did you subscribe to park+3500 and setup your presence map? ? /b On Tue, Aug 23, 2022 at 12:03 PM Jason Holden wrote: > All, > > Trying to monitor parking slots for valet park on Freeswitch. > > Using Polycoms or Voice Operator Pannal regular extension BLF works fine > but I do not see the BLF status for the parking slots if a call is on hold > their. > > Does anyone have any recommendations? Not finding much online about this. > > > > > > Jason Holden > > > > Phone: 1-866-836-9198 X405 > > Direct: 7868009949 > > www.thinksimplicity.com > > > > [image: Logo 2019 Email Final 250] > > > _________________________________________________________________________ > > 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 > -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 1004 bytes Desc: not available URL: From jasonh at thinksimplicity.com Wed Aug 24 16:36:31 2022 From: jasonh at thinksimplicity.com (=?UTF-8?Q?Jason_Holden?=) Date: Wed, 24 Aug 2022 16:36:31 +0000 Subject: [Freeswitch-users] freeswitch 1.8 and 1.10 Valet parking BLF problem In-Reply-To: References: <01000182cba26b12-aa11051d-6e28-4358-8e8e-e6c59a0a9c30-000000@email.amazonses.com> Message-ID: <01000182d0b666da-0ead6dfb-7510-4b10-87fa-323a0b2c1ad7-000000@email.amazonses.com> I assume I can have multiple exten regex lines?     Jason Holden   Phone: 1-866-836-9198 X405 Direct: 7868009949 www.thinksimplicity.com     From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, August 24, 2022 11:40 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] freeswitch 1.8 and 1.10 Valet parking BLF problem   Did you subscribe to park+3500 and setup your presence map?                       ?   /b   On Tue, Aug 23, 2022 at 12:03 PM Jason Holden > wrote: All, Trying to monitor parking slots for valet park on Freeswitch. Using Polycoms or Voice Operator Pannal regular extension BLF works fine but I do not see the BLF status for the parking slots if a call is on hold their. Does anyone have any recommendations? Not finding much online about this.     Jason Holden   Phone: 1-866-836-9198 X405 Direct: 7868009949 www.thinksimplicity.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   --   Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.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: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 1004 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.jpg Type: image/jpeg Size: 1004 bytes Desc: not available URL: From brian at freeswitch.com Wed Aug 24 17:38:55 2022 From: brian at freeswitch.com (Brian West) Date: Wed, 24 Aug 2022 12:38:55 -0500 Subject: [Freeswitch-users] freeswitch 1.8 and 1.10 Valet parking BLF problem In-Reply-To: <01000182d0b666da-0ead6dfb-7510-4b10-87fa-323a0b2c1ad7-000000@email.amazonses.com> References: <01000182cba26b12-aa11051d-6e28-4358-8e8e-e6c59a0a9c30-000000@email.amazonses.com> <01000182d0b666da-0ead6dfb-7510-4b10-87fa-323a0b2c1ad7-000000@email.amazonses.com> Message-ID: Yes On Wed, Aug 24, 2022 at 12:05 PM Jason Holden wrote: > I assume I can have multiple exten regex lines? > > > > > > Jason Holden > > > > Phone: 1-866-836-9198 X405 > > Direct: 7868009949 > > www.thinksimplicity.com > > > > [image: Logo 2019 Email Final 250] > > > > *From:* FreeSWITCH-users [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West > *Sent:* Wednesday, August 24, 2022 11:40 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] freeswitch 1.8 and 1.10 Valet parking > BLF problem > > > > Did you subscribe to park+3500 and setup your presence map? > > > > > > > > > > > > > > > > > > > > ? > > > > /b > > > > On Tue, Aug 23, 2022 at 12:03 PM Jason Holden > wrote: > > All, > > Trying to monitor parking slots for valet park on Freeswitch. > > Using Polycoms or Voice Operator Pannal regular extension BLF works fine > but I do not see the BLF status for the parking slots if a call is on hold > their. > > Does anyone have any recommendations? Not finding much online about this. > > > > > > Jason Holden > > > > Phone: 1-866-836-9198 X405 > > Direct: 7868009949 > > www.thinksimplicity.com > > > > [image: Logo 2019 Email Final 250] > > > > _________________________________________________________________________ > > 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 > > > > > -- > > > > Brian West | Co-founder and Developer > > Need Commercial support? email sales at freeswitch.com > > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > > > Email: brian at freeswitch.com > > Mobile: 918-424-9378 > > Website: https://www.FreeSWITCH.com > > [image: https://www.facebook.com/signalwireinc?src=email] > [image: > https://twitter.com/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 > -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 1004 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.jpg Type: image/jpeg Size: 1004 bytes Desc: not available URL: From jolexpert at gmail.com Wed Aug 24 09:25:38 2022 From: jolexpert at gmail.com (Kakiman Expert) Date: Wed, 24 Aug 2022 11:25:38 +0200 Subject: [Freeswitch-users] Fwd: websocket disconnection issue - error 1006 In-Reply-To: References: Message-ID: Hello I am working on FreeSWITCH 1.10.6 ans sip.js 0.16, and very often, I have a deconnection of the WSS session. The error message on javascript console on Chrome is "error 1006 - websocket unexpectly closed" This issue is very random but very often, and it breaks the incoming call. Do you experiment such issue ? Do you know how to solve it ? thanks Franck -------------- next part -------------- An HTML attachment was scrubbed... URL: From jolexpert at gmail.com Wed Aug 24 19:49:20 2022 From: jolexpert at gmail.com (Kakiman Expert) Date: Wed, 24 Aug 2022 21:49:20 +0200 Subject: [Freeswitch-users] Websocket closed unexpectly Message-ID: Hello I am working on FreeSWITCH 1.10.6 ans sip.js 0.16, and very often, I have a deconnection of the WSS session. The error message on javascript console on Chrome is "error 1006 - websocket unexpectly closed" This issue is very random but very often, and it breaks the incoming call. Do you experiment such issue ? Do you know how to solve it ? thanks Franck -------------- next part -------------- An HTML attachment was scrubbed... URL: From dodu at hotmail.co.uk Fri Aug 26 09:10:03 2022 From: dodu at hotmail.co.uk (Josh H) Date: Fri, 26 Aug 2022 09:10:03 +0000 Subject: [Freeswitch-users] VP8 conference FIR Signal not always sent when frames are dropped Message-ID: Hi, I am using Verto to connect to FreeSwitch over webRTC in a conference call. This works fine most of the time however when using an older device that does not support hardware acceleration for the video encoding, it appears that some frames are dropped. The CPU is pegged at 100% utilisation, with 60-80% of that being from the chrome process. This is understandable when using older hardware however when this occurs FreeSwitch fails to send FIR packets to get a new keyframe, leading to the ‘video mute’ image being shown until the next scheduled keyframe (which with VP8 is, by default, around 3000 frames, equivalent to 100 seconds at 30fps). The client seems to be unaware of any issues as it continues to encode frames and send them over the network, leading me to believe it is an issue with FreeSwitch. My current solution is to enable the ‘kf-max-dist’ parameter in ‘vpx.conf.xml’ however this leads to occasional stuttering, I assume this is because it must encode a full frame every 10 seconds. The other solution I have tested is swapping to the H.264 codec (the only other codec supported by webRTC) which makes the issue less noticeable however the quality of the stream drops significantly when no FIR is requested. H.264 isn’t an option for me as some of my target clients are Android devices which do not support H.264 over webRTC. Given that sometimes the session works as expected with FIR packets being sent when needed, leading to a small video stutter before it recovers, I am unsure how to fix it or what logs etc to provide. Any pointers would be appreciated 😊 Thanks Josh -------------- next part -------------- An HTML attachment was scrubbed... URL: