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: <sip:104 at 192.168.1.10>;tag=ec09c0b4zps4.0.0 From: "108"<sip:108 at 192.168.1.154>;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 Totowa, NJ 07512 Phone: 201.806.2602 x2405 Fax: 201.806.2604 Cell: 973.390.1090 www.xaccel.net<http://www.xaccel.net/> CONFIDENTIALITY NOTICE: This e-mail message, including any attachments, is for the sole use of the intended recipient(s) and may contain confidential and privileged information which should not be shared or forwarded. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail and destroy all copies of the e-mail. 0¿ ª0NEiJ@@LÀ¨ À¨ÄÄ}ª!SIP/2.0 400 Bad Request To: <sip:104 at 192.168.1.10>;tag=ec09c0b4zps4.0.0 From: "108"<sip:108 at 192.168.1.154>;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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190624/2727f871/attachment-0001.html> -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 4300 bytes Desc: image001.png URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190624/2727f871/attachment-0001.png> -------------- next part -------------- A non-text attachment was scrubbed... Name: capture-to-aiphonewithholdandwaitforpreviewvideo Type: application/octet-stream Size: 22285 bytes Desc: capture-to-aiphonewithholdandwaitforpreviewvideo URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190624/2727f871/attachment-0004.obj> -------------- next part -------------- A non-text attachment was scrubbed... Name: capture-to-aiphonewithhold Type: application/octet-stream Size: 38959 bytes Desc: capture-to-aiphonewithhold URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190624/2727f871/attachment-0005.obj> -------------- next part -------------- A non-text attachment was scrubbed... Name: capture-from-aiphone Type: application/octet-stream Size: 26704 bytes Desc: capture-from-aiphone URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190624/2727f871/attachment-0006.obj> -------------- next part -------------- A non-text attachment was scrubbed... Name: capture-to-aiphone Type: application/octet-stream Size: 13624 bytes Desc: capture-to-aiphone URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190624/2727f871/attachment-0007.obj>
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: <sip:104 at 192.168.1.10>;tag=ec09c0b4zps4.0.0 From: "108"<sip:108 at 192.168.1.154>;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 <image001.png> 380 US Highway 46, Suite 500 Totowa, NJ 07512 Phone: 201.806.2602 x2405 Fax: 201.806.2604 Cell: 973.390.1090 www.xaccel.net<http://www.xaccel.net/> CONFIDENTIALITY NOTICE: This e-mail message, including any attachments, is for the sole use of the intended recipient(s) and may contain confidential and privileged information which should not be shared or forwarded. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail and destroy all copies of the e-mail. 0¿ ª0NEiJ@@LÀ¨ À¨ÄÄ}ª!SIP/2.0 400 Bad Request To: <sip:104 at 192.168.1.10>;tag=ec09c0b4zps4.0.0 From: "108"<sip:108 at 192.168.1.154>;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 <capture-to-aiphonewithholdandwaitforpreviewvideo> <capture-to-aiphonewithhold> <capture-from-aiphone> <capture-to-aiphone> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190625/0c468e61/attachment.html>
On Thu, Jun 27, 2019, at 11:28 AM, John T. Bittner wrote:> > 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 getYou 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: www.digium.com & 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----- From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua C. Colp Sent: Thursday, June 27, 2019 10:41 AM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Looking Asterisk SIP Guru On Thu, Jun 27, 2019, at 11:28 AM, John T. Bittner wrote:> > 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 > getYou 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: www.digium.com & www.asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- BEGIN-ANTISPAM-VOTING-LINKS ------------------------------------------------------ Teach Canit xAntispam if this mail is spam: Spam: http://mx1.xantispam.net/canit/b.php?c=s&i=020tOGuhK&m=13b91acb500d&rlm=xaccel-net Not spam: http://mx1.xantispam.net/canit/b.php?c=n&i=020tOGuhK&m=13b91acb500d&rlm=xaccel-net Forget vote: http://mx1.xantispam.net/canit/b.php?c=f&i=020tOGuhK&m=13b91acb500d&rlm=xaccel-net ------------------------------------------------------ END-ANTISPAM-VOTING-LINKS