- FreeSWITCH-users - FreeSWITCH Mailing Lists

Help with tls
by Dr. Ogg 03 Dec '20

03 Dec '20

I have been looking for some good documentation regarding the configuration of TLS with FreeSWITCH. If anyone has a good resource, I would be grateful Thanks…

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Masking caller
by Lars Kiesow 03 Dec '20

03 Dec '20

Hi everyone, I'm trying to mask the caller_id_name in a FreeSWITCH dialplan to prevent the real phone numbers to show up in our conferencing software. Someone sent me the following lines: <action application="set" data="MASK=${system echo ${caller_id_name} | grep -o -P '.{0,4}$' | sed 's/^/xxx-xxx-/' }"/> <action application="set_profile_var" data="caller_id_name=${MASK}"/> While this works perfectly and does exactly what I want, I'm unsure about potential security risks. The caller_id_name ends up in a shell command after all and I'm wondering if someone could send a name like `; rm /*` (you get the idea). Is this safe? Is the caller_id_name sanitized? Is there a better way to do something like this? Best regards, Lars

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How to relay DTMF during Early Media (Before Call Answered) ?
by Hamid Hashmi 03 Dec '20

03 Dec '20

Hello I have been trying to relay DTMF in Early Media (Before Call Answered), I have tried the following option. (Ref<http://freeswitch-users.2379917.n2.nabble.com/start-dtmf-generate-during-ea…>) <extension name="other"> <condition> <action application="export" data="nolocal:absolute_codec_string=AMR-WB,AMR"/> <action application="export" data="nolocal:execute_on_media=start_dtmf_generate> <action application="set" data="uuid_bridge_continue_on_cancel=true"/> <action application="set" data="bridge_early_media=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=false"/> <action application="set" data="effective_caller_id_number=${sip_from_user}"/> <action application="set" data="hold_music=/etc/freeswitch/dialplan/Call_Hold.mp3"/> <action application="set" data="ringback=%(2000,4000,440.0,480.0)"/> <action application="set" data="limit_ignore_transfer=false" /> <action application="bridge" data="sofia/internal/${destination_number}@${loadbalancer}"/> <action application="set" data="sip_copy_multipart=false"/> <action application="log" data="1 B-leg hangup cause: ${last_bridge_hangup_cause}"/> <action application="set" data="sip_h_X-RBT-Enabled=No"/> <action application="transfer" data="${last_bridge_hangup_cause}"/> </condition> </extension> but it didn't generate any DTMF in early media and it caused an issue of Robotic sound in transcoding from AMR-WB to G729 as well. OS: DISTRIB_DESCRIPTION="Ubuntu 18.04.2 LTS" NAME="Ubuntu" VERSION="18.04.2 LTS (Bionic Beaver)" Container: PRETTY_NAME="Debian GNU/Linux 8 (jessie)" NAME="Debian GNU/Linux" VERSION_ID="8" VERSION="8 (jessie)" Freeswitch: FreeSWITCH Version 1.9.0-n20180314T133342Z-1~jessie+1+git~20180305T173609Z~dd0bb0e331~64bit (-n20180314T133342Z-1~jessie+1git dd0bb0e 2018-03-05 17:36:09Z 64bit) PS: DTMF is working fine once the call is answered. Regards Hamid R. Hashmi

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Full Trickle ICE via mod_sofia supported?
by Alexander Haugg 02 Dec '20

02 Dec '20

Hello, I only mean that the client can deliver candidates. The Freeswitch of course sends all candidates in the first SDP. Currently I'm trying to implement the Full Trickle mechanism as described here https://tools.ietf.org/id/draft-ietf-mmusic-trickle-ice-sip-11.html Candidates should be forwarded via SIP Info during the connection setup. Since I haven't succeeded so far, I have the following questions. Is Full Trickle supported via mod_sofia? If so, which settings or sip headers are required? Thanks Alex

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Change SDP by SIP UPDATE
by Oleg Belousov 02 Dec '20

02 Dec '20

Hi. Is there a possibility in Freeswitch to update the SDP of a pre-answered call by SIP UPDATE message? Call flow is as following: 1. FS responds to A-party with 183/SDP (SDP IP==FS IP) and performs some checks in preanswer mode 2. If checks on step 1 passed ok, FS initiates a call to B-party, receives from B-party 183/SDP (SDP IP = MGW_B) and sends to A-party 183/SDP with same new SDP IP = MGW_B, so doing change of SDP parameter in the same transaction. That scenario executes with below commands (need in media travel directly between parties after the bridge) session:execute("set","bypass_media=true") second_session = session:execute("bridge",BridgeStr); The question: is there a parameter which allows during the bridge to update SDP by SIP UPDATE, instead of changing SDP using provisional responses? A-party switch does not accept such SDP change and thus there is a problem with audio after the bridge. -- obelousov.tel

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20ms silent RTP inserted into conference call output (about 1 out of every 100 packets)
by Larry Hemenway 01 Dec '20

01 Dec '20

We're setting up a conference bridge as follows: (canned wav file) --> chrome --(g722)--> FreeSWITCH conference --(linear PCM)--> c++ app What we are seeing is that FreeSWITCH is inserting 20 ms of silence into the linear PCM output. It does not drop any audio - it is just inserting extra silence. On some traces we see ~1 out of every 100 output audio packet as silence. There is no regular cadence for the insertion. We are able to verify this in two ways: 1. When we look at the linear PCM tcpdump trace, we can identify silence because it's all 0xff. 2. We convert both the g722 and linear PCM into wav files. This is how we observe that no audio is dropped, just 20ms of silence added. The linear PCM wav file adds 20ms of audio and falls behind the g722 audio. We look at the pacing from the tcpdump and chrome is pacing every 20ms. FreeSWITCH, likewise, is also producing at 20ms intervals. We did try turning on the jitter buffer in FreeSWITCH and now see mitigated packets instead of silence, so that did not fix the issue. We'll probably turn off PLC next as mitigated packets are not easily identifiable. After reading https://github.com/signalwire/freeswitch/issues/735 I went back and looked at an old trace and it does appear that we see silence in the output linear PCM ~50-70ms after getting an RTCP packet from chrome. It's not 1:1; in the trace I looked at we had 44 rtcp packets, but only 20 silent packets, but for every silent packet we see an RTCP packet 50-70ms earlier. I hope to get more traces to verify this after turning PLC off. Environment: FreeSWITCH version 1.10.3, running inside a container on bare metal. This is reproducible running a single conference with two participants - no loading. We are turning off encryption in Chrome so we can decode the g722 (no dtls/srtp) and feeding chrome a canned wav file, so we can compare audio. Has anyone else seen this issue? Any suggestions on how to start debugging? I'm trying to trace through read_rtp_packets() in switch_rtp.c, but am a bit lost and not sure if that is where I should start. This issue is severely degrading our audio quality measurements so its causing a lot of headaches for us at the moment.

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How to DIY/Setup An Open Source IP PBX Appliance/Server?
by Turritopsis Dohrnii Teo En Ming 01 Dec '20

01 Dec '20

Subject: How to DIY/Setup An Open Source IP PBX Appliance/Server? Good day from Singapore, After reading recent reviews, I gather that Asterisk is the gold standard when it comes to open source VoIP systems and it is the most famous open source PBX out there. Article: Compare the Top 10 Best Open Source PBX Software of 2020 Link: https://www.voipreview.org/business-voip/best-open-source-pbx-software Article: Top 10 Free Open Source PBX Software Solutions Link: https://getvoip.com/blog/2016/09/23/best-open-source-pbx-software/ The following is an excerpt from Wikipedia: "Asterisk is a core component in many commercial products and open-source projects. Some of the commercial products are hardware and software bundles, for which the manufacturer supports and releases the software with an open-source distribution model. AskoziaPBX, a fork of the m0n0wall project, uses Asterisk PBX software to realize all telephony functions. AstLinux is a "Network Appliance for Communications" open-source software distribution.[15] FreePBX, an open-source graphical user interface, bundles Asterisk as the core of its FreePBX Distro[16] LinuxMCE bundles Asterisk to provide telephony; there is also an embedded version of Asterisk for OpenWrt routers. PBX in a Flash/Incredible PBX and trixbox are software PBXes based on Asterisk. Elastix previously used Asterisk, HylaFAX, Openfire and Postfix to offer PBX, fax, instant messaging and email functions, respectively, before switching to 3CX. Issabel is an open-source Unified Communications software which uses Asterisk for telephony functions. It was forked from the open-source versions of Elastix when 3CX acquired it. *astTECS uses Asterisk in its VoIP and mobile gateways." Link: https://en.wikipedia.org/wiki/Asterisk_(PBX) I would like to DIY/setup an IP PBX appliance/server using free open source projects. Which free open source project, mentioned in the list and links above, would you recommend to DIY my IP PBX appliance/server? Should I buy a desktop computer or get one of those appliances listed in the link below to serve as my IP PBX appliance/server? Link: https://www.lazada.sg/products/pfsense-iron-metal-case-fanless-intel-celero… Please also refer me to very good, detailed and well explained guides/tutorials/manuals on setting up open source IP PBX appliances/servers. Lastly, please recommend a cheap and affordable IP phone (suggest brand and model) to go along with my DIY open source IP PBX appliance/server. Mr. Turritopsis Dohrnii Teo En Ming, 42 years as of 1st December 2020 Tuesday, is a TARGETED INDIVIDUAL (TI) living in Singapore. Thank you very much. -----BEGIN EMAIL SIGNATURE----- The Gospel for all Targeted Individuals (TIs): [The New York Times] Microwave Weapons Are Prime Suspect in Ills of U.S. Embassy Workers Link: https://www.nytimes.com/2018/09/01/science/sonic-attack-cuba-microwave.html ******************************************************************************************** Singaporean Targeted Individual Mr. Turritopsis Dohrnii Teo En Ming's Academic Qualifications as at 14 Feb 2019 and refugee seeking attempts at the United Nations Refugee Agency Bangkok (21 Mar 2017), in Taiwan (5 Aug 2019) and Australia (25 Dec 2019 to 9 Jan 2020): [1] https://tdtemcerts.wordpress.com/ [2] https://tdtemcerts.blogspot.sg/ [3] https://www.scribd.com/user/270125049/Teo-En-Ming -----END EMAIL SIGNATURE-----

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Force Inband DTMF Reception
by Adrian Worutowicz 30 Nov '20

30 Nov '20

Hello, For incoming calls, I want to receive DTMF inband. FS receives INVITE and at certain stage sends 200 OK with a SDP section, which always contains : a=rtpmap:101 telephone-event/8000 (what means 'DTMF in RTP metadata'). Indeed the DTMFs come via RTP. I tried various FS configuration options to remove that line, without success. How to do it? How to tell a gateway to send DTMFs inband ? In advance thank you very, very much, Adrian.

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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

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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