Dan Cropp
2022-May-13 19:05 UTC
[asterisk-users] [External] [External] [External] Asterisk 18.12.0 question
Thank you Joshua!!! Not loading chan_sip module resolved the problem. Hope you have an awesome weekend. From: asterisk-users <asterisk-users-bounces at lists.digium.com> On Behalf Of Joshua C. Colp Sent: Friday, May 13, 2022 1:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [External] [asterisk-users] [External] [External] Asterisk 18.12.0 question On Fri, May 13, 2022 at 3:19 PM Dan Cropp <dan at amtelco.com<mailto:dan at amtelco.com>> wrote: Thanks Joshua. I didn’t describe that very well. When I first noticed the res_http_transport_websocket wasn’t loading on that box, I compared the modules folder on both boxes. My thought was I forgot some module that was required. I noticed I forgot to include these files, so I added them to the package. Rolled back the VM and re-installed. Didn’t make a difference whether they were present or not. /usr/lib/asterisk/modules/codec_g729a.* /usr/lib/asterisk/modules/codec_silk.* /usr/lib/asterisk/modules/codec_siren14.* /usr/lib/asterisk/modules/codec_siren7.* /usr/lib/asterisk/modules/format_ogg_opus.so Comparing the menuselect-tree between the two versions, only changes I see are func_evalexten res_aeap res_speech_aeap and four test_aeap_... added to the TEST_FRAMEWORK. Would it make sense for me to modify my bash script to disable those settings, compile, and try installing? Bash script configures the menuselect options and compiles asterisk. Seems like that would be a better apples to apples comparison. Eliminating the new features. You can. It would also make sense as a test to just not load chan_sip and see what happens. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com<http://www.sangoma.com> and www.asterisk.org<http://www.asterisk.org> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20220513/0a4fcd7f/attachment-0001.html>
Dan Cropp
2022-May-19 16:18 UTC
[asterisk-users] [External] [External] [External] Asterisk 18.12.0 question
After further testing, not sure this is chan_sip related. I can disable chan_sip.so from loading in modules.conf and that does solve the startup/loading for res_pjsip_transport_websocket. However, there is some issue with the wss transport. Seeing this in both 16.26.0 (not in 16.25.0) and 18.12.0 (not in 18.11.2). REGISTER message comes in, is accepted. However, when it goes to send the OPTIONS, it’s outputting the Unsupported transport. [05/19 10:11:41.992] VERBOSE[2456] res_pjsip_logger.c: <--- Received SIP request (907 bytes) from WSS:192.168.32.27:56443 ---> REGISTER sip:mybox.mydomain.com SIP/2.0 Via: SIP/2.0/WSS c2537bthsnvo.invalid;branch=z9hG4bK2816987 Max-Forwards: 69 To: <sip:1234 at mybox.mydomain.com> From: <sip:1234 at mybox.mydomain>;tag=24ipeon952 Call-ID: lshogr91tba8r5f335c1g5 CSeq: 2 REGISTER Authorization: Digest algorithm=MD5, username="1234", realm="asterisk", nonce="1652973101/72159fe10d9432b64a16fec84fc414e7", uri="sip:mybox.mydomain.com", response="f46f710af7db6e2e86ec2fabe38325e8", opaque="06a146a816d699e2", qop=auth, cnonce="meehpb38l93l", nc=00000001 Contact: <sip:e6hj0uh4 at c2537bthsnvo.invalid;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:032f947e-da85-4920-b944-86b52760937b>";expires=600 Expires: 600 Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO Supported: path,gruu,outbound User-Agent: JsSIP 3.3.6 Content-Length: 0 [05/19 10:11:41.993] VERBOSE[2456] res_pjsip_logger.c: <--- Transmitting SIP response (482 bytes) to WSS:192.168.32.27:56443 ---> SIP/2.0 200 OK Via: SIP/2.0/WSS c2537bthsnvo.invalid;rport=56443;received=192.168.32.27;branch=z9hG4bK2816987 Call-ID: lshogr91tba8r5f335c1g5 From: <sip:1234 at mybox.mydomain.com>;tag=24ipeon952 To: <sip:1234 at mybox.mydomain.com>;tag=z9hG4bK2816987 CSeq: 2 REGISTER Date: Thu, 19 May 2022 15:11:41 GMT Contact: <sip:e6hj0uh4 at 192.168.32.27:56443;transport=ws>;expires=599 Expires: 600 Server: Asterisk PBX 18.12.0 Content-Length: 0 [05/19 10:11:41.994] ERROR[2456] res_pjsip.c: Error 171060 'Unsupported transport (PJSIP_EUNSUPTRANSPORT)' sending OPTIONS request to endpoint 1234 Identical behavior happening with Asterisk 16.26.0, but not on Asterisk 16.25.0 Configuration files are same for between Asterisk versions. [transport3] type = transport bind = 0.0.0.0 protocol = wss allow_reload = no [1234] type = aor max_contacts = 1 remove_existing = yes qualify_frequency = 60 [1234] type = auth auth_type = userpass username = 1234 password = mypassword [1234] type = endpoint context = IS auth = 1234 aors = 1234 dtmf_mode = rfc4733 webrtc = yes disallow = all allow = ulaw transport = transport3 acl = acl5 Might this be because PJSIP 2.12 changes to the “WebRTC updates with AEC3 & AGC2” From: Dan Cropp Sent: Friday, May 13, 2022 2:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: RE: [External] [asterisk-users] [External] [External] Asterisk 18.12.0 question Thank you Joshua!!! Not loading chan_sip module resolved the problem. Hope you have an awesome weekend. From: asterisk-users <asterisk-users-bounces at lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com>> On Behalf Of Joshua C. Colp Sent: Friday, May 13, 2022 1:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com<mailto:asterisk-users at lists.digium.com>> Subject: Re: [External] [asterisk-users] [External] [External] Asterisk 18.12.0 question On Fri, May 13, 2022 at 3:19 PM Dan Cropp <dan at amtelco.com<mailto:dan at amtelco.com>> wrote: Thanks Joshua. I didn’t describe that very well. When I first noticed the res_http_transport_websocket wasn’t loading on that box, I compared the modules folder on both boxes. My thought was I forgot some module that was required. I noticed I forgot to include these files, so I added them to the package. Rolled back the VM and re-installed. Didn’t make a difference whether they were present or not. /usr/lib/asterisk/modules/codec_g729a.* /usr/lib/asterisk/modules/codec_silk.* /usr/lib/asterisk/modules/codec_siren14.* /usr/lib/asterisk/modules/codec_siren7.* /usr/lib/asterisk/modules/format_ogg_opus.so Comparing the menuselect-tree between the two versions, only changes I see are func_evalexten res_aeap res_speech_aeap and four test_aeap_... added to the TEST_FRAMEWORK. Would it make sense for me to modify my bash script to disable those settings, compile, and try installing? Bash script configures the menuselect options and compiles asterisk. Seems like that would be a better apples to apples comparison. Eliminating the new features. You can. It would also make sense as a test to just not load chan_sip and see what happens. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com<http://www.sangoma.com> and www.asterisk.org<http://www.asterisk.org> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20220519/811c4353/attachment.html>