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Genesys Cloud SIP integration with TLS fails when sofia global siptrace on
by Unai Rodriguez 29 Apr '22

29 Apr '22

Dear List, Genesys Cloud SIP trunk works fine when using TCP UDP. It also work when using TLS. It fails when using TLS and we have sofia global siptrace on. TCP dump shows connection reset with multiple retransmissions. This is turning into a big issue for us, is there a way to accelerate resolution? https://github.com/freeswitch/sofia-sip/issues/124 Thank you With best wishes, Unai Rodriguez

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gateway AUTH not working after update to 1.10.7 (from 1.10.6)
by Marcel Haldemann 27 Apr '22

27 Apr '22

Hi, i updated from FS 1.10.6 to 1.10.7 and after the update the gateway AUTH for outbound calls is not working anymore. * The Config files remain unchanged * The gateway is shown as REGED ins "sofia status" * The Dialstring used: sofia/gateway/sip.peoplefone.ch/+4199999(a)sips.peoplefone.ch The message is get is: 2022-04-27 08:55:46.177774 97.20% [WARNING] sofia_reg.c:2857 Cannot locate any authentication credentials to complete an authentication request for realm '"sip.peoplefone.ch"' 2022-04-27 08:55:46.177774 97.20% [NOTICE] sofia_reg.c:2881 Hangup sofia/carrier/%2B41447165008(a)sips.peoplefone.ch [CS_CONSUME_MEDIA] [MANDATORY_IE_MISSING] ... (see FS-Logs.txt) for more ... the gateway in the profile looks like this: <gateways> <gateway name="sip.peoplefone.ch"> <param name="username" value="4199999999"/> <param name="password" value="xxxxxxxxxxxxxxxxxxxxx"/> <param name="expire-seconds" value="180"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="retry-seconds" value="10"/> <param name="cid-type" value="none"/> </gateway> </gateways> ... see profile_carrier.txt for more ... Sofia Status: reeswitch(a)trtc.convercom.com> sofia status Name Type Data State ================================================================================================= carrier profile sip:mod_sofia@195.201.140.229:5060 RUNNING (0) carrier::sip.peoplefone.ch gateway sip:90783808802@sip.peoplefone.ch REGED ================================================================================================= 1 profile 0 aliases Why is this not working anymore with 1.10.7, but it was with 1.10.6 and how to make it work again?

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Serving configuration with mod_lua and sofia.conf
by Lloyd Aloysius 27 Apr '22

27 Apr '22

Hi All When I use the mod_cml_curl and sofia.conf . The following expanding properly. <param name="sip-ip" value="$${local_ip_v4}"/> <param name="sip-port" value="$${internal_sip_port}"/> But I use lua and sofia.conf, the following not working. Variables not calculated. <param name="sip-ip" value="$${local_ip_v4}"/> <param name="sip-port" value="$${internal_sip_port}"/> Why when I send the xml configuration using lua not working ($${local_ip_v4},$${internal_sip_port}) . But the same xml work for xml_curl Thanks Lloyd

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Call forward to another server/extension.
by HS 21 Apr '22

21 Apr '22

Dear all. I have an Opensips 3.0 + Fusion PBX setup. I would like calls received on Ext. 500 on Box A to be transferred to ext. 123 on Box B. I have successfully configured a trunk to Box B by setting up a Gateway and the status is REGED. I also added 123 in the Ext. field. In the dialplan expression I added 500. However, when I make the call it seems that I am transferred to ext. 500 on Box B and not Ext. 123. I am sure this has been discussed previously, but I can't seem to find much information. What am I doing incorrectly please?

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RE - Internal Interface suddenly freezes
by Kakiman Expert 11 Apr '22

11 Apr '22

hello Did you find any solution to this issue ? I am in production with 1.10.5 version and since a week, I have the same issue. The internal profile is out of service and registration is impossible. and we don't have any answer to "sofia status profile internal" , the prompt in fs_cli keeps waiting. the only way to solve is to restart freeswitch thanks regards

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user registration
by Kakiman Expert 08 Apr '22

08 Apr '22

Hello I am using freeSWITCH 10.06 and i am wondering about the difference between 2 commands : - show registration - sofia status profile internal reg both commands list the users registered, but sometimes, I am still seing users in show registrations whereas they are not present in sofia status. which is the command to have the right registration status ? what is the difference between these commands ? thanks regards

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Why doesn’t FreeSwitch offer PCMU/PCMA to the callee?
by Дилян Палаузов 08 Apr '22

08 Apr '22

Hello, I want to call from Linphone on Android to Gnome Calls. The SIP trace can be found at https://gitlab.gnome.org/GNOME/calls/-/issues/434 . Linphone announces OPUS. FreeSWITCH is supposed to do transcoding between codecs. It proposes however to Gnome Calls only OPUS as viable protocol. Gnome Calls rejects the INVITATION, as it does not speak OPUS. I the SIP profile I have <param name="inbound-codec-prefs" value="OPUS,G722,PCMU,PCMA,H264,VP8"/> <param name="outbound-codec-prefs" value="OPUS,G722,PCMU,PCMA,H264,VP8"/> While calling, the variable ep_codec_string has the value mod_opus.opus@48000h@20i@2c,CORE_PCM_MODULE.PCMU@8000h@20i@64000b,CORE_PCM_MODULE.PCMA@8000h @20i@64000b,mod_spandsp.G722@8000h@20i@64000b. I have enabled Late Negotiation. freeswitch@d> sofia status profile internal ======================================================================= ========================== Name internal Domain Name N/A Auto-NAT false DBName sofia_reg_internal Pres Hosts 192.168.0.199,192.168.0.199 Dialplan XML Context public Challenge Realm auto_from RTP-IP 192.168.0.199 SIP-IP 192.168.0.199 URL sip:mod_sofia@192.168.0.199:5060 BIND-URL sip:mod_sofia@192.168.0.199:5060;transport=udp,tcp WS-BIND-URL sip:mod_sofia@192.168.0.199:5066;transport=ws WSS-BIND-URL sips:mod_sofia@192.168.0.199:7443;transport=wss HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN OPUS,G722,PCMU,PCMA,H264,VP8 CODECS OUT OPUS,G722,PCMU,PCMA,H264,VP8 TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 MAX-RECV-RPS 1000 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT false CALLS-IN 12 FAILED-CALLS-IN 6 CALLS-OUT 6 FAILED-CALLS-OUT 6 REGISTRATIONS 4 FreeSwith offers only OPUS to Gnome Calls and the latter rejects the connection. The question is, why doesn't FreeSwitch offer transcoding, and the PCMA+PCMU codecs to the callee? I have one more question. I want to execute some actions unconditionally in a dialplan, in particular to record all calls. I do: <context name="default"> <extension name="recording" continue="true"> <condition break='never' > <anti-action application="set" data="record_sample_rate=32000"/> <anti-action application="record_session" data="$${recordings_dir}/B-${strftime(%Y-%m-%d-%H-%M- %S)}_${destination_number}_${caller_id_number}.wav"/> <action application="set" data="record_sample_rate=32000"/> <action application="record_session" data="$${recordings_dir}/A- ${strftime(%Y-%m-%d-%H-%M- %S)}_${destination_number}_${caller_id_number}.wav"/> </condition> </extension>  </context> It does work, but sometimes the action is executed, sometimes the anti- action. ( I hope I say the truth, since I made a lot of changes recently and these are the results I remember). Is there any way to execute a condition unconditionally, or rather to execute actions outside of a condition and extension? Here, to record all calls, without having an extension/condition? Thanks in advance for your ideas! Greetings Дилян

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Subject: Bridging two incoming channels in Bypass media mode
by Aditya Uppuluri 07 Apr '22

07 Apr '22

Yes. Both the clients are on the same LAN and I want to bridge both. But without FS being in the media path. Regards, Aditya Uppuluri

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Bridging two incoming channels in Bypass media mode
by Aditya Uppuluri 05 Apr '22

05 Apr '22

Hello Team. Is there a way to bridge two incoming calls in bypass media mode? I tried using uuid_bridge but it's expecting at least one channel to be in *answered* state. And if I answer one of the channels, then freeswitch is coming into the media path which I don't want. I have an external profile with inbound-bypass-media to true. Let me know if I can provide you with any further details. Thank you Regards, Aditya Uppuluri

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Re: [Freeswitch-users] Repository server licensing
by Bote Man 04 Apr '22

04 Apr '22

Do what I do: pull down the source code and compile it yourself. There is no need for the Personal Access Token, which only is needed to pull down the binary packages. John Boteler Bote Communications -----Original Message----- From: FreeSWITCH-users <freeswitch-users-bounces(a)lists.freeswitch.org> On Behalf Of Laura Smith via FreeSWITCH-users Sent: Monday, 4 April, 2022 06:36 To: freeswitch-users(a)lists.freeswitch.org Subject: [Freeswitch-users] Repository server licensing I find it somewhat odd that nobody seems to care much about the implied licensing that people are now forced to sign up to with the stupid Freeswitch repo changes. Specifically the registration action has an "By continuing you agree to our Legal Policies" commitment... _________________________________________________________________________ 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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