November 2020 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Issue: INCOMPATIBLE_DESTINATION / AUDIO RTP REPORTS ERROR [Missing local host]
by Maciej Bylica 30 Nov '20

30 Nov '20

Hi Will check that one and update you Thanks Maciej. wt., 24 lis 2020 o 19:42 Sonny Lafuente via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> napisał(a): > > > > ---------- Forwarded message ---------- > From: Sonny Lafuente <lafuente_sonny(a)yahoo.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Sun, 22 Nov 2020 21:21:02 +0800 > Subject: Re: [Freeswitch-users] Issue: INCOMPATIBLE_DESTINATION / AUDIO > RTP REPORTS ERROR [Missing local host] > Try to add this line in the vars.xml > > <action application="set" data="media_mix_inbound_outbound_codecs=true"/> > > > Regards, > Sonny > > On Nov 22, 2020, at 4:03 AM, Maciej Bylica <mbgatherer(a)gmail.com> wrote: > > > Sure, will try... > > Thanks > MAciej > > pt., 20 lis 2020 o 17:47 Brian West <brian(a)freeswitch.com> napisał(a): > >> set them to "autonat:x.x.x.x" then set local-network-acl >> >> >> On Fri, Nov 20, 2020 at 6:45 AM Maciej Bylica <mbgatherer(a)gmail.com> >> wrote: >> >>> We normally use "proxy_media=true" and this is the situation that >>> triggers the issue. >>> IPs provided are modified into 10.20.30.xx form, but we use public IPs, >>> so no NAT is involved, hence both parameters have default values like: >>> >>> <param name="ext-rtp-ip" value="auto-nat"/> >>> >>> <param name="ext-sip-ip" value="auto-nat"/> >>> >>> Thanks >>> Maciej. >>> >>> pt., 20 lis 2020 o 12:42 David Villasmil <david.villasmil.work(a)gmail.com> >>> napisał(a): >>> >>>> In order to really use bypass_media, the two endpoints MUST see each >>>> other, no NAT involved. >>>> Although I read somewhere from Brian that parameter does nothing (but I >>>> could be misremembering) >>>> In any case, what do you have in your sofia profile in ext-rtp-ip and >>>> ext-sip-ip? >>>> >>>> On Fri, 20 Nov 2020 at 10:52, Maciej Bylica <mbgatherer(a)gmail.com> >>>> wrote: >>>> >>>>> Hello, >>>>> >>>>> Sorry for delay, i had to catch up on a few things a bit. >>>>> Instead of 79.x.x.x. it should be 10.20.30.41 .... and yes this is >>>>> what i expect to have. >>>>> >>>>> But i still don't know why FS generates such an event. >>>>> As for workaround i use "bypass_media=true" instead of " >>>>> proxy_media=true", but the question is how to solve it. >>>>> >>>>> Thanks >>>>> Maciej >>>>> >>>>> >>>>> >>>>> sob., 14 lis 2020 o 09:21 Brian : <brians(a)iptel.co> napisał(a): >>>>> >>>>>> I dont know if youre sanitizing ips but the sdp from the 'terminator' >>>>>> has an ip that isnt in any of the other signalling. Is that what you >>>>>> expect? >>>>>> >>>>>> 79.x.x.x >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> On Friday, November 6, 2020, Maciej Bylica <mbgatherer(a)gmail.com> >>>>>> wrote: >>>>>> > Hi all, >>>>>> > I am working on 1.10.5 (today's build) with proxy_media=true >>>>>> configuration set in dialplan config. >>>>>> > Around 2-3% of total call attempts are CANCELed with >>>>>> "INCOMPATIBLE_DESTINATION" / "AUDIO RTP REPORTS ERROR: [Missing local >>>>>> host]" notification. >>>>>> > I assume that the problem is probably closely connected with codec >>>>>> negotiation, but in my case (proxy_media=true) FS does forward the packets >>>>>> onward and endpoints must agree on the same codec to accept the call. >>>>>> > I have compared all the SIP call flows incoming and outgoing >>>>>> (INVITES, 183) and found no clue. >>>>>> > >>>>>> > Some of the important log snippets: >>>>>> > - the call is CANCELed once FS receives 183 and gets first early >>>>>> media packets >>>>>> > >>>>>> > 8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 >>>>>> [INFO] switch_ivr_originate.c:3801 Sending early media >>>>>> > 8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 >>>>>> [DEBUG] switch_core_media.c:8729 PROXY AUDIO RTP [sofia/outside_1/ >>>>>> 4916222112233(a)10.20.30.20] 10.20.30.6:26502->10.20.30.6:26502 codec: >>>>>> 18 ms: 20 >>>>>> > 8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 >>>>>> [ERR] switch_core_media.c:9669 AUDIO RTP REPORTS ERROR: [Missing local host] >>>>>> > 8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 >>>>>> [NOTICE] switch_core_media.c:9670 Hangup sofia/outside_1/ >>>>>> 4916222112233(a)10.20.30.20 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] >>>>>> > a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 >>>>>> [DEBUG] switch_ivr_originate.c:3808 sofia/outside_1/ >>>>>> 4916222112233(a)10.20.30.20 Media Establishment Failed. >>>>>> > a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 >>>>>> [NOTICE] switch_ivr_originate.c:3810 Hangup >>>>>> sofia/outside_1/3144917111223344 [CS_CONSUME_MEDIA] >>>>>> [INCOMPATIBLE_DESTINATION] >>>>>> > 8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 >>>>>> [DEBUG] switch_ivr_originate.c:3995 Originate Resulted in Error Cause: 88 >>>>>> [INCOMPATIBLE_DESTINATION] >>>>>> > a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 >>>>>> [DEBUG] switch_channel.c:3565 (sofia/outside_1/3144917111223344) Callstate >>>>>> Change DOWN -> EARLY >>>>>> > a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 >>>>>> [DEBUG] sofia.c:7339 sofia/outside_1/3144917111223344 skip receive message >>>>>> [PROGRESS_EVENT] (channel is hungup already) >>>>>> > 8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 >>>>>> [INFO] mod_dptools.c:3631 Originate Failed. Cause: INCOMPATIBLE_DESTINATION >>>>>> > a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 >>>>>> [DEBUG] switch_core_media.c:12291 sofia/outside_1/3144917111223344 too late. >>>>>> > >>>>>> > - incoming (Originator -> FS) SIP INVITE has following SDP >>>>>> > v=0 >>>>>> > o=- 1604689110 1604689110 IN IP4 10.20.30.6 >>>>>> > s=- >>>>>> > c=IN IP4 10.20.30.6 >>>>>> > t=0 0 >>>>>> > m=audio 26502 RTP/AVP 18 0 8 101 >>>>>> > a=rtpmap:18 G729/8000 >>>>>> > a=fmtp:18 annexb=no >>>>>> > a=rtpmap:0 PCMU/8000 >>>>>> > a=rtpmap:8 PCMA/8000 >>>>>> > a=rtpmap:101 telephone-event/8000 >>>>>> > a=fmtp:101 0-15 >>>>>> > a=ptime:20 >>>>>> > a=sendrecv >>>>>> > a=silenceSupp:off - - - - >>>>>> > >>>>>> > - outgoing (FS -> Terminator) SDP of SIP INVITE looks like this >>>>>> (codec transparent) >>>>>> > v=0 >>>>>> > o=FreeSWITCH 4241000132 4241000133 IN IP4 10.20.30.20 >>>>>> > s=FreeSWITCH >>>>>> > c=IN IP4 10.20.30.20 >>>>>> > t=0 0 >>>>>> > m=audio 31034 RTP/AVP 18 0 8 101 >>>>>> > a=rtpmap:18 G729/8000 >>>>>> > a=fmtp:18 annexb=no >>>>>> > a=rtpmap:0 PCMU/8000 >>>>>> > a=rtpmap:8 PCMA/8000 >>>>>> > a=rtpmap:101 telephone-event/8000 >>>>>> > a=fmtp:101 0-15 >>>>>> > a=ptime:20 >>>>>> > a=silenceSupp:off - - - - >>>>>> > >>>>>> > - 183 that is received (Terminator -> FS) >>>>>> > v=0 >>>>>> > o=JOANO_SDP 24930105 0 IN IP4 10.20.30.40 >>>>>> > s=JOANO-SIP >>>>>> > c=IN IP4 79.133.196.167 >>>>>> > t=0 0 >>>>>> > m=audio 18200 RTP/AVP 8 101 >>>>>> > a=rtpmap:8 PCMA/8000 >>>>>> > a=rtpmap:101 telephone-event/8000 >>>>>> > a=fmtp:101 0-15 >>>>>> > a=silenceSupp:off - - - - >>>>>> > a=ptime:20 >>>>>> > >>>>>> > As a result FS tears down the call attempt by sending (FS -> >>>>>> Originator) >>>>>> > >>>>>> > SIP/2.0 488 Not Acceptable Here >>>>>> > Via: SIP/2.0/UDP 10.20.30.10:5060 >>>>>> ;received=10.20.30.10;rport=5060;branch=z9hG4bK0609c6993b9c-30-16a290 >>>>>> > Via: SIP/2.0/UDP 10.20.30.2:5060;branch=z9hG4bK5765.136934e2.0 >>>>>> > Max-Forwards: 66 >>>>>> > From: <sip:4916222112233@10.20.30.5:5061;user=phone>;tag=b6b8aec94s >>>>>> > To: <sip:3144917111223344@10.20.30.20;user=phone>;tag=DcUe4343Uvj8Q >>>>>> > Call-ID: 684d5998-d859dd12-ca30a685-551b1244 >>>>>> > CSeq: 1 INVITE >>>>>> > User-Agent: FreeSWITCH-mod_sofia/1.10.5-release~64bit >>>>>> > Accept: application/sdp >>>>>> > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >>>>>> REGISTER, REFER, NOTIFY >>>>>> > Supported: timer, path, replaces >>>>>> > Allow-Events: talk, hold, conference, refer >>>>>> > Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" >>>>>> > Content-Length: 0 >>>>>> > >>>>>> > and CANCEL (FS -> Terminator) >>>>>> > CANCEL sip:3144917111223344@10.20.30.30:5060 SIP/2.0 >>>>>> > Via: SIP/2.0/UDP 10.20.30.20:5080;rport;branch=z9hG4bKK01B9mcmpp5jS >>>>>> > Max-Forwards: 70 >>>>>> > From: <sip:4916222112233@10.20.30.20>;tag=g76r9mQeKQN0a >>>>>> > To: <sip:3144917111223344@10.20.30.30:5060> >>>>>> > Call-ID: e788fdfd-9b04-1239-6881-fc15b4264fc0 >>>>>> > CSeq: 27790763 CANCEL >>>>>> > Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" >>>>>> > Content-Length: 0 >>>>>> > >>>>>> > Could somebody help me to address the issue I am struggling with ? >>>>>> > >>>>>> > Detailed debugging logs and sip signalization might be found here: >>>>>> > https://pastebin.com/G0sjGiw8 - FS logs >>>>>> > https://pastebin.com/2Zt6r2mF - SIP logs >>>>>> > >>>>>> > Many thanks in advance, >>>>>> > Maciej >>>>>> > >>>>>> _________________________________________________________________________ >>>>>> >>>>>> 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(a)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(a)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(a)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(a)lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> https://freeswitch.com >>>> >>>> -- >>>> Regards, >>>> >>>> David Villasmil >>>> email: david.villasmil.work(a)gmail.com >>>> phone: +34669448337 >>>> >>>> _________________________________________________________________________ >>>> >>>> The FreeSWITCH project is sponsored by SignalWire >>>> https://signalwire.com >>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>>> services. >>>> Build your next product on our scalable cloud platform. >>>> >>>> Join our online community to chat in real time >>>> https://signalwire.community >>>> >>>> Professional FreeSWITCH Services >>>> sales(a)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(a)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(a)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(a)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(a)freeswitch.com >> >> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 >> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&e…> >> >> Email: brian(a)freeswitch.com >> >> Mobile: 918-424-9378 >> >> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> >> >> [image: https://www.facebook.com/signalwireinc?src=email] >> <https://www.facebook.com/freeswitch> [image: >> https://twitter.com/freeswitch] <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(a)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(a)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(a)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(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > > > > ---------- Forwarded message ---------- > From: Sonny Lafuente via FreeSWITCH-users < > freeswitch-users(a)lists.freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Tue, 24 Nov 2020 10:42:02 -0800 (PST) > Subject: Re: [Freeswitch-users] Issue: INCOMPATIBLE_DESTINATION / AUDIO > RTP REPORTS ERROR [Missing local host] > _________________________________________________________________________ > > 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(a)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(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com

3 4

SMPP incoming message error with deliver_sm sequence number = 0
by I. Adebayo 30 Nov '20

30 Nov '20

Hi all We are testing mod_smpp using a kannel smpp server. We have two issues: 1. When kannel sends a message with deliver_sm with sequence number = 0, the mod_smpp receives the message with all the parameters set to 0 and throws error ""mod_smpp_gateway.c:257 smpp: error decoding the receive buffer: .. invalid sequence number". Is there any way we can resolve this issue? 2. After mod_smpp establishes a bind to the server, it does not send enquire_link. It will always time out and rebind again. As smpp client, it should be sending enquire_link to keep the connection alive. Is there any way we can enable the enquire_link in the module configuration? Thanks for your help. Ismail <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> Virus-free. www.avast.com <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

1 0

macOS Wiki - Wow 10 years, already!
by mario_fs 27 Nov '20

27 Nov '20

For macOS users and anyone interested: Nov 22 2020 is the tenth anniversary of when I put up the first complete install guide for macOS on the wiki <https://freeswitch.org/confluence/display/FREESWITCH/macOS>, boy does that make me feel old. I realized this yesterday while doing my weekly FreeSwitch Development build/test for the macOS wiki. I know some have found it useful since I've seen it referenced on the mailing list, some info like emailing voicemail may be useful to Linux users. I also tried helping with the Linux stuff but that didn't last. Mario G

6 8

Timer D (32 seconds) being triggered after ACK in TCP INVITE Transaction
by João Vitor Arruda 26 Nov '20

26 Nov '20

Hello, I have a scenario where freeswitch bridges a call received with TCP on leg A to a destination with TCP on leg B When the destination rejects the call with 486 (Busy Here) that is sent back to leg A which receives the ACK as expected. So from the leg A perspective we have: INVITE 100 Trying 180 Ringing 486 Busy Here ACK After the ACK freeswitch seems to be starting the Timer D (32 seconds) even for that TCP transaction. This is what the log show when the ACK is received: tport.c:2771 tport_wakeup() tport_wakeup(0x561ffce1c3e0): events IN tport.c:2868 tport_recv_event() tport_recv_event(0x561ffce1c3e0) tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x561ffce1c3e0) msg 0x561ffcf2f700 from (tcp/192.168.100.100:34863) has 394 bytes, veclen = 1 tport.c:3023 tport_deliver() tport_deliver(0x561ffce1c3e0): msg 0x561ffcf2f700 (394 bytes) from tcp/192.168.100.100:34863/sip next=0 nta.c:2877 agent_recv_request() nta: received ACK sip:6111@172.16.100.1;transport=tcp SIP/2.0 (CSeq 1) nta.c:3017 agent_recv_request() nta: ACK (1) is going to INVITE (1) tport.c:2298 tport_set_secondary_timer() tport(0x561ffce1c3e0): reset timer 2020-11-26 17:37:16.622026 [DEBUG] switch_scheduler.c:144 Deleting task 292 switch_ivr_schedule_hangup (372ca1ff-4466-48c5-a5a0-a22612a217ed) then 32 seconds later nta.c:9106 outgoing_timer_dk() nta: timer D fired, terminate INVITE (28652323) tport.c:2298 tport_set_secondary_timer() tport(0x561ffcf30a80): reset timer nta.c:8804 outgoing_reclaim_queued() outgoing_reclaim_all(0, 0, 0x7f52bceb8958) nta.c:8927 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free tport.c:2298 tport_set_secondary_timer() tport(0x561ffce1c3e0): reset timer nta.c:5824 incoming_reclaim_queued() incoming_reclaim_all(0, 0, 0x7f52bceb8958) nta.c:7186 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout, 0/0 term, 1/1 free nta.c:1289 agent_timer() nta: timer not set Based on the RFC 3261 section-17.1.1.2 ( https://tools.ietf.org/html/rfc3261#section-17.1.1.2) it seems that Timer D should zero seconds for reliable transport (TCP): The ACK MUST be sent to the same address, port, and transport to which the original request was sent. The client transaction SHOULD start timer D when it enters the "Completed" state, with a value of at least 32 seconds for unreliable transports, and a value of zero seconds for reliable transports Timer D reflects the amount of time that the server transaction can remain in the "Completed" state when unreliable transports are used. Should this be considered a BUG?

1 0

sip Allow headers differents between 2 legs
by Mickael Hubert 26 Nov '20

26 Nov '20

Hi all, FS receives on its A-leg this SIP Allow header: Allow: OPTIONS, PUBLISH, INVITE, ACK, BYE, CANCEL But when it sends INVITE to B-leg, it adds others parameters: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY I saw that it's not possible to modify Allow header with FS. Is there a way to copy exactly the same Allow header between 2 legs ? thanks in advance ++

1 0

Help my son, answer a poll on Libra/CryptoCurrencies/Facebook
by Giovanni Maruzzelli 26 Nov '20

26 Nov '20

Hey FreeSWITCHers! Please take 50 seconds to check some radiobuttons on this anonymous poll: https://docs.google.com/forms/d/e/1FAIpQLSdPBoZTm_yCBRJB0lkQxqSa8eJrRAaM9pS… My son Rocco is writing a paper about Libra (FaceBook cryptocurrency) acceptance and privacy concerns. You'll be proud of yourself ! And I'll be very glad/grateful :)) -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

10 11

hangup from LUA script
by Gandalf Corvotempesta 25 Nov '20

25 Nov '20

Hi guys, i'm trying to make a very simple IVR with a lua script. I'm calling the lua script in this way: <action application="play_and_get_digits" data="6 12 10 7000 # say:'Please inser order number.' silence_stream://250 order_number \d+" /> <action application="lua" data="myivr.lua"/> And myivr.lua has something like this: ---------------------------------------------------------------- if session:getVariable("order_number") ~= "1234567" then session:speak("Error"); session:hangup("NO_ROUTE_DESTINATION") end session:speak("connecting to customer. Please wait"); session:setAutoHangup(false); session:execute("bridge", "{ignore_early_media=true,bridge_generate_comfort_noise=true,effective_caller_id=anonymous,hangup_after_bridge=true}sofia/gateway/mygateway/123456789"); -------------------------------------------------------------------- Now, the questions: 1) Biggest issue is that when I call session:hangup() , the call is still running, how can I force hangup ? 2) as this lua script is called after a "play_and_get_digits" (in dialplan), how can I replace the "Error" inside IF with something that will re-execute the dialplan from scratch asking for a new number ? In other words, if numbers are not "1234567", I have to ask for the numbers again. 3) how can I set the bridge to be anonymous on the remote phone ? I don't want the remote user to be able to see the caller number. It would be better if remote user will se MY number and not the origin number (it's a bridge: ORIGIN -> PBX -> REMOTE, the remote will see origin number. I want to be anonymous or PBX number) Tnx in advance.

2 2

SOFIA channel for Outgoing calls
by Vishal Kakkar 25 Nov '20

25 Nov '20

Hi all, Using a SIP trunk with 30 channels. Is there a way in sofia, to ensure all channels are used equally for *outgoing* *calls*? Like a roundrobin scheme to ensure the load is distributed across all 30 channels in a single SIP trunk. Thanks, Vishal.

3 3

mod_event_socket: command to play and collect DTMF
by Dmitry Mordovin 24 Nov '20

24 Nov '20

Hello I try to solve task: play a prompt and collect DTMF digits. Now I use Lua script and session:playAndGetDigits to did it. How can I do it thru mod_event_socket commands? I know uuid_broadcast command can play prompt, but I can’t find command to wait some DTMF digits. Any one can direct me to right simple solution? PS: one idea, it can be done if I can run same Lua script from mod_event_socket

4 3

Trickle ICE support
by Ramesh Kandasamy 24 Nov '20

24 Nov '20

Hi all, Does FS support the Trickle ICE feature yet? If yes, can someone please point me a link on the documentation for the same? Thanks Ramesh

1 0