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Elrick B.

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Posts posted by Elrick B.

  1. Grandstream GRP2612 "No Response" dialing out after 69.1 or 69.1.1 Upgrade, works fine if back on 69.0.8

    in Grandstream Phones

    Posted

    On 11/12/2023 at 7:53 AM, Vodia PBX said:

    This problem was introduced with the video calling rework of the SDP handling. WebRTC is massively using SDP and with it comes a cleanup of the SDP parsing. Before 69, the PBX was tolerant (if not ignorant) with the media types, and RTP/AVP was seen as equivalent to RTP/SAVP. However it is not, and the tolerance could lead to static white noise. Now the PBX is more strict, and thus the trunks need explicitly be instructed to use RTP/SAVP. We are trying to figure out an automatic way to upgrade those trunks, but so far the manual change is the only way. The next build will at least have the templates for new trunks corrected. 

    Ok thanks

  2. Grandstream GRP2612 "No Response" dialing out after 69.1 or 69.1.1 Upgrade, works fine if back on 69.0.8

    in Grandstream Phones

    Posted

    On 11/2/2023 at 3:36 PM, Vodia PBX said:

    Well... it is ringing. At least something!

    However I would try to set the ice: savp to explicitly turn on RTP/SAVP. You can do that in 69.1.2 in the trunk by switching to the text mode and edit the line there (we need to add it to the HTML page as well). Then it should be doing SRTP for outbound calls. 

    Inbound calls are using SRTP properly?

    After changing that option in text mode. It fixed both issues (Calls ending, static when resuming call on hold). 

    Was that listed in the change logs? I didn't see it.

     

  3. Grandstream GRP2612 "No Response" dialing out after 69.1 or 69.1.1 Upgrade, works fine if back on 69.0.8

    in Grandstream Phones

    Posted

    On 11/2/2023 at 3:36 PM, Vodia PBX said:

    Well... it is ringing. At least something!

    However I would try to set the ice: savp to explicitly turn on RTP/SAVP. You can do that in 69.1.2 in the trunk by switching to the text mode and edit the line there (we need to add it to the HTML page as well). Then it should be doing SRTP for outbound calls. 

    Inbound calls are using SRTP properly?

    Sorry for taking so long to get back to you. Inbound calls are the same, it rings once quickly then ends the call. I will try your solution now.

  4. Grandstream GRP2612 "No Response" dialing out after 69.1 or 69.1.1 Upgrade, works fine if back on 69.0.8

    in Grandstream Phones

    Posted

    1 hour ago, Vodia PBX said:

    Oh so there are a few things with TLS on 69. You need to set an additional parameter to tell it to use TLS: 

    ice: savp

    Maybe you can attach the SIP INVITE and the response here (change the numbers to XXX) then this should become clear. 

    No longer have the pcap from the system as I uninstalled and reinstalled the PBX since then, but I do have Invite and Response from the Telnyx side:

INVITE sip:+1813xxxxxxx@sip.telnyx.com;user=phone SIP/2.0
Via: SIP/2.0/TLS 34.x.x.x:55514;branch=z9hG4bK-2efef9a6c828658f720b980265180c38;rport
From: <sip:+1656xxxxxxx@sxxxxx.xxxxxxcloud.com;user=phone>;tag=909344563
To: <sip:+1813xxxxxxx@sip.telnyx.com;user=phone>
Call-ID: 0gpsc8s@pbx
CSeq: 9594 INVITE
Max-Forwards: 70
Contact: <sip:xxxxxxxxx@34.x.x.x:55514;transport=tls>
Allow-Events: refer
Supported: 100rel, replaces, norefersub
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: Vodia-PBX/69.1.1
Remote-Party-ID: <sip:+1656xxxxxxx@sxxxxx.xxxxxxcloud.com;user=phone>
Proxy-Authorization: Digest realm="sip.telnyx.com",nonce="29c73b43-ae74-4f53-9f86-3cef85b7aabe",response="51404a08861a08388b0e023d2d6dfbcd",username="stxxxxxxx",uri="sip:+1813xxxxxxx@sip.telnyx.com;user=phone",qop=auth,nc=00000001,cnonce="373284d9",opaque="fbe7a55b-5951-449e-a422-bc203fdedf9f/10.13.37.24",algorithm=MD5
Content-Type: application/sdp
Content-Length: 372

v=0
o=- 2103907024 2103907024 IN IP4 34.x.x.x
s=-
c=IN IP4 34.x.x.x
t=0 0
m=audio 56512 RTP/AVP 0 8 109 9 101
c=IN IP4 34.x.x.x
a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=fmtp:101 0-15
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000/1
a=sendrecv

      

    SIP/2.0 180 Ringing
Via: SIP/2.0/TLS 34.x.x.x:55514;received=34.x.x.x;branch=z9hG4bK-2efef9a6c828658f720b980265180c38;rport=55514
Record-Route: <sip:10.255.0.1;transport=tcp;r2=on;lr;ftag=909344563>
Record-Route: <sip:192.76.120.10:5061;transport=tls;r2=on;lr;ftag=909344563>
From: <sip:+1656xxxxxxx@sxxxxx.xxxxxxcloud.com;user=phone>;tag=909344563
To: <sip:+1813xxxxxxx@sip.telnyx.com;user=phone>;tag=ep3yQgQ8mmKZa
Call-ID: 0gpsc8s@pbx
CSeq: 9594 INVITE
Contact: <sip:+1813xxxxxxx@10.13.37.24:5070;transport=tcp>
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY
Supported: path
Allow-Events: talk, hold, conference, refer
Content-Length: 0
X-Telnyx-Session-ID: 3b30dd96-74f7-11ee-bb75-02420a0de568
X-Telnyx-Leg-ID: 3b30d468-74f7-11ee-8f87-02420a0de568

     

  5. Grandstream GRP2612 "No Response" dialing out after 69.1 or 69.1.1 Upgrade, works fine if back on 69.0.8

    in Grandstream Phones

    Posted

    Not to stray too far from the OPs issue.

    When using our Yealink phones on the latest build with Telnyx on TLS, calls are not completing. It tries to ring then fail, and after checking the PCAP on the system and on Telnyx, it gave a "Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION", as if they are not able to negotiate codecs. I am guessing something happened between version 68.0.32 and 69.1.2 that doesn't play nice with Telnyx.

    I also have an issue when putting incoming calls on hold, once resumed its pure static. That issue is not present when placing outbound calls on hold though.

    Both issues are resolved if I roll back to v68.0.32.

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