Unable to establish call

[Pammetje]

Hi, I am unable to establish a call

I succesfully registered my SIP account at an external SIP Provider (not an internal Fritsbox) When I am trying to establish a call to my own mobile phone, it is not ringing last lines in the logfile ->

13:57:12.505 pjsua_core.c .TX 1847 bytes Request msg INVITE/cseq=14912 (tdta0x7f13e1254b88) to UDP 185.29.203.27:5060: INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP MY_INTERNET_EXIT_IPv4 ADDRESS:56256;rport;branch=z9hG4bKPjvb4djRSc3vEtdNuEolr2X6nLF4wqesPj Max-Forwards: 70 From: sip:[email protected];tag=AF2aSZ5KjHctQIWx99ybi.xjIZ-nBE6v To: sip:[email protected] Contact: <sip:MY_SIP_PHONENUMBER@<MY INTERNET EXIT IPv4 ADDRESS>:56256;ob>;+sip.ice Call-ID: gb5gqbnHqmbNRa6v56sX6.752E1CVNB6 CSeq: 14912 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 Content-Type: application/sdp Content-Length: 1198

v=0 o=- 3916472231 3916472231 IN IP4 IPv4_ADDRESS_HOME_ASSISTANT s=pjmedia b=AS:117 t=0 0 a=X-nat:0 m=audio 4008 RTP/AVP 96 97 98 99 3 0 8 9 100 120 121 122 123 c=IN IP4 IPv4_ADDRESS_HOME_ASSISTANT b=TIAS:96000 a=rtcp:4005 IN IPv4_ADDRESS_HOME_ASSISTANT a=sendrecv a=rtpmap:96 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:98 speex/32000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:100 opus/48000/2 a=fmtp:100 useinbandfec=1 a=rtpmap:120 telephone-event/16000 a=fmtp:120 0-16 a=rtpmap:121 telephone-event/8000 a=fmtp:121 0-16 a=rtpmap:122 telephone-event/32000 a=fmtp:122 0-16 a=rtpmap:123 telephone-event/48000 a=fmtp:123 0-16 a=ssrc:1675884403 cname:7a381cd37e4e4d80 a=ice-ufrag:4933c834 a=ice-pwd:7d683e3944eadf4d674d258d a=candidate:Hac111602 1 UDP 2130706431 IPv4_ADDRESS_HOME_ASSISTANT 4008 typ host a=candidate:Hac1e2001 1 UDP 2130706175 172.30.32.1 4008 typ host a=candidate:Hac1ee801 1 UDP 2130705919 172.30.232.1 4008 typ host a=candidate:Hac111602 2 UDP 2130706430 IPv4_ADDRESS_HOME_ASSISTANT 4005 typ host a=candidate:Hac1e2001 2 UDP 2130706174 172.30.32.1 4005 typ host a=candidate:Hac1ee801 2 UDP 2130705918 172.30.232.1 4005 typ host

[RRobby]

Good morning @Pammetje , did you solve your issue? I'm exactly in the same situation; no ring on outbound call to my mobile phone. Making a packet capture on router , I see that SIP INVITE go out DSL interface but no packets come back from SIP provider.

In order to exclude any issue related to firewall/NAT I made the same call using a SIP client on my pc ( works like a charms and obviously a see return packets from provider) . I compared the packet captures because I suppose that could be something related to SDP ( Session Description Protocol ); In NO WORKING invite ( ha-sip from Home Assistant ) there are a lot of parameters ( codecs / ICE candidates ) that could not be supported by provider.

Question to all ; Is this possible slim down the SDP content in order to provide very few necessary parameters to SIP provider ( codec ,dtmf , audio ports,contact and nothing else ? Thanks in advance for any suggestion.
Roberto

[Pammetje]

I was not able to solve the issue because my SIP provider uses other/incompatible codecs/settings (than a Fritzbox) My final solution was installing a local FreePBX VM, connect the local FreePBX VM to my SIP provider, and connect the HA plugin to the local FreePBX. I got this setup up & running for half a year now and its very stable. (with a local FreePBX you have full control on codecs and settings)