3 thoughts on - Looking Asterisk SIP Guru

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  Hello,

  I am looking for a consultant that know asterisk in and out including how to troubleshoot sip and rtp. I have a device that this acting very strange and I need to prove it’s the device code and not an issue with my setup.

  Very simple setup, all local no nat… Grandstream video phone and a AIphone IX-MX7 door station.

  PJSIP … doorstation to grandstream 3370 works perfectly. Early video works as well. PJSIP … grandtream to doorstation I get a error from the doorstation I get

  SIP/2.0 400 Bad Request To: ;tag=ec09c0b4zps4.0.0
  From: “108”;tag=bcaee3d1-d6d1-4354-aee1-885d4b89182d Via: SIP/2.0/UDP 192.168.1.154:5060;branch=z9hG4bKPj038d15bb-c53b-4677-a471-4fa44a21599b;rport Call-ID: 08caa86d-8fc8-4ed1-bec2-6828acb9e017
  CSeq: 17397 INVITE
  Content-Length: 0
  x-reinvitekind: mediadirectionchange

  Tried a few things, I still don’t understand why I am getting this, I cannot find it coming from the asterisk system or the Grandstream in my traces. So Switch the Aiphone to use chan_sip on port 5099 just to test.

  Again SIP … doorstation to PJSIP grandstream 3370 works perfectly. Early video works as well. PJSIP … granstream to SIP doorstation works somewhat, I get early video but no audio. If I answer the doorstation before the early video pops up, I get the window in the doorstation that allows me to put a call on hold. When I do, and take back off hold, I get audio. If I wait for early video on the doorstation and then answer it, the door station never comes up with the menus to put a call on hold. So no audio.

  Anyone have any ideas or willing to do some consulting work please let me know asap. FYI some captures are attached.

  Thanks

  John Bittner CTO
  [xaccellogoemail]
  380 US Highway 46, Suite 500
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• You didn’t provide the IP addresses of things involved, so anyone looking at the packet captures has to look in and decipher what is what which may be why noone on here has responded as of yet. The user agent of Asterisk is also changed so that confused things some for me to until I double checked the SDP and saw it’s Asterisk.

  Asterisk is sending a re-invite to 192.168.1.10 as an attempt to make both the audio and video streams bidirectional. The device at 192.168.1.10 is rejecting this with a 400 Bad Request. It should respond either with a 200 OK with an SDP answer of the state of the streams, or it should respond with a 488 Not Acceptable. Both of these would keep the call up and the appropriate stream would probably flow although I haven’t tested this particular usage.

  You also didn’t specify an Asterisk version from what I can see, and stream behavior between 13 and 16 differs (as 16 understands streams) which could contribute to the behavior.

  —
  Joshua C. Colp Digium – A Sangoma Company | Senior Software Developer
  445 Jan Davis Drive NW – Huntsville, AL 35806 – US
  Check us out at: http://www.digium.com & http://www.asterisk.org

  —

• Joshua,

  Thanks for looking into this, and sorry for not being more detailed. Running asterisk 16.4.0

  I was able to get in touch with an AIphone tech and it turns out that these issues are known bug on their side.

  I will be more detailed next time

  Thanks

  John Bittner Xaccel

  —–Original Message—