Unable to create channel of type 'PJSIP' (cause 58 - Bearer capability not available) Everyone is busy/congested at this time (1:0/0/1)

Alan_Demerda · December 13, 2017, 9:44pm

Hi Guys,

i have installed Asterisk 15.3 with Pjsip 2.6 on a cloud server. I am testing using https://www.doubango.org/sipml5 …trying to call web2 from web1. The call is not established and I am getting this error:

“Unable to create channel of type ‘PJSIP’ (cause 58 - Bearer capability not available) Everyone is busy/congested at this time (1:0/0/1)”

Here are the full pjsip log:

<— Received SIP request (1009 bytes) from UDP:188.165.231.30:10060 —>
REGISTER sip:pbxbox.net SIP/2.0
Via: SIP/2.0/UDP 188.165.231.30:10060;branch=z9hG4bKOyOmuMhUh9iUcmbhnKQt7Fcc5THkLGH4;rport
From: "web1"sip:web1@pbxbox.net;tag=ugvWaP0hJ1y2wJMQsEVM
To: "web1"sip:web1@pbxbox.net
Contact: "web1"sip:web1@188.165.231.30:10060;rtcweb-breaker=no;transport=udp;ws-src-ip=41.33.108.226;ws-src-port=13509;ws-src-proto=wss;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: e3a258e4-2c6c-b4b6-199e-2979dc88267d
CSeq: 19201 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“web1”,realm=“asterisk”,nonce=“1513200633/b53d1f615995919cfa167f10a36a7332”,uri=“sip:pbxbox.net”,response=“bbf8ee8eb53cc3c926a565de6c619f3a”,algorithm=md5,cnonce=“25999dc8f8663e4d47d2b1f9566a0722”,opaque=“15dad41a4f5570c0”,qop=auth,nc=00000002
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom
Via: SIP/2.0/TCP 41.33.108.226:13509;rport;branch=z9hG4bKOyOmuMhUh9iUcmbhnKQt7Fcc5THkLGH4;ws-hacked=WSS

<— Transmitting SIP response (678 bytes) to UDP:188.165.231.30:10060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 188.165.231.30:10060;rport=10060;received=188.165.231.30;branch=z9hG4bKOyOmuMhUh9iUcmbhnKQt7Fcc5THkLGH4
Via: SIP/2.0/TCP 41.33.108.226:13509;rport;branch=z9hG4bKOyOmuMhUh9iUcmbhnKQt7Fcc5THkLGH4;ws-hacked=WSS
Call-ID: e3a258e4-2c6c-b4b6-199e-2979dc88267d
From: “web1” sip:web1@pbxbox.net;tag=ugvWaP0hJ1y2wJMQsEVM
To: “web1” sip:web1@pbxbox.net;tag=z9hG4bKOyOmuMhUh9iUcmbhnKQt7Fcc5THkLGH4
CSeq: 19201 REGISTER
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1513200734/4c2b64a1dfec2175e6f41843c88e1af6”,opaque=“4101b0d54b113011”,stale=true,algorithm=md5,qop="auth"
Server: Asterisk PBX 15.1.3
Content-Length: 0

<— Received SIP request (1009 bytes) from UDP:188.165.231.30:10060 —>
REGISTER sip:pbxbox.net SIP/2.0
Via: SIP/2.0/UDP 188.165.231.30:10060;branch=z9hG4bKoVhSRqpoPoEFbWrSCWTItgAON6TdZ0aD;rport
From: "web1"sip:web1@pbxbox.net;tag=ugvWaP0hJ1y2wJMQsEVM
To: "web1"sip:web1@pbxbox.net
Contact: "web1"sip:web1@188.165.231.30:10060;rtcweb-breaker=no;transport=udp;ws-src-ip=41.33.108.226;ws-src-port=13509;ws-src-proto=wss;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: e3a258e4-2c6c-b4b6-199e-2979dc88267d
CSeq: 19202 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“web1”,realm=“asterisk”,nonce=“1513200734/4c2b64a1dfec2175e6f41843c88e1af6”,uri=“sip:pbxbox.net”,response=“89d8c7b3256ca38b838ed46d243a24fe”,algorithm=md5,cnonce=“27685cf93b1224fe9e151d981d970ce5”,opaque=“4101b0d54b113011”,qop=auth,nc=00000001
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom
Via: SIP/2.0/TCP 41.33.108.226:13509;rport;branch=z9hG4bKoVhSRqpoPoEFbWrSCWTItgAON6TdZ0aD;ws-hacked=WSS

-- Added contact 'sip:web1@188.165.231.30:10060;transport=udp;rtcweb-breaker=no;ws-src-ip=41.33.108.226;ws-src-port=13509;ws-src-proto=wss' to AOR 'web1' with expiration of 200 seconds
-- Removed contact 'sip:web1@188.165.231.30:10060;transport=udp;rtcweb-breaker=no;ws-src-ip=41.35.175.27;ws-src-port=57405;ws-src-proto=wss' from AOR 'web1' due to remove_existing

<— Transmitting SIP response (692 bytes) to UDP:188.165.231.30:10060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 188.165.231.30:10060;rport=10060;received=188.165.231.30;branch=z9hG4bKoVhSRqpoPoEFbWrSCWTItgAON6TdZ0aD
Via: SIP/2.0/TCP 41.33.108.226:13509;rport;branch=z9hG4bKoVhSRqpoPoEFbWrSCWTItgAON6TdZ0aD;ws-hacked=WSS
Call-ID: e3a258e4-2c6c-b4b6-199e-2979dc88267d
From: “web1” sip:web1@pbxbox.net;tag=ugvWaP0hJ1y2wJMQsEVM
To: “web1” sip:web1@pbxbox.net;tag=z9hG4bKoVhSRqpoPoEFbWrSCWTItgAON6TdZ0aD
CSeq: 19202 REGISTER
Date: Wed, 13 Dec 2017 21:32:14 GMT
Contact: sip:web1@188.165.231.30:10060;transport=udp;rtcweb-breaker=no;ws-src-ip=41.33.108.226;ws-src-port=13509;ws-src-proto=wss;expires=199
Server: Asterisk PBX 15.1.3
Content-Length: 0

== Contact web1/sip:web1@188.165.231.30:10060;transport=udp;rtcweb-breaker=no;ws-src-ip=41.33.108.226;ws-src-port=13509;ws-src-proto=wss has been created
== Contact web1/sip:web1@188.165.231.30:10060;transport=udp;rtcweb-breaker=no;ws-src-ip=41.35.175.27;ws-src-port=57405;ws-src-proto=wss has been deleted
– Contact web1/sip:web1@188.165.231.30:10060;transport=udp;rtcweb-breaker=no;ws-src-ip=41.33.108.226;ws-src-port=13509;ws-src-proto=wss is now Unknown. RTT: 0.000 msec
<— Received SIP request (3367 bytes) from UDP:188.165.231.30:10060 —>
INVITE sip:web1@pbxbox.net SIP/2.0
Via: SIP/2.0/UDP 188.165.231.30:10060;branch=z9hG4bK76l7flGtVogAZoLNsmWUg6zs1mIR7wpT;rport
From: "web2"sip:web2@pbxbox.net;tag=VqH9t01oTl5y9BXvBc3M
To: sip:web1@pbxbox.net
Contact: "web2"sip:web2@188.165.231.30:10060;rtcweb-breaker=no;click2call=no;transport=udp;ws-src-ip=41.35.175.27;ws-src-port=52467;ws-src-proto=wss;+g.oma.sip-im;language="en,fr"
Call-ID: f45b457f-fa04-b587-4424-b5ae9fc6e15c
CSeq: 28015 INVITE
Content-Type: application/sdp
Content-Length: 2630
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom
Via: SIP/2.0/TCP 41.35.175.27:52467;rport;branch=z9hG4bK76l7flGtVogAZoLNsmWUg6zs1mIR7wpT;ws-hacked=WSS

v=0
o=- 2367316182999042600 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS VNq9ACBHIOK0YcfWUlAAGaUahtigoIX0Ijme
m=audio 51850 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 41.35.175.27
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:3543857585 1 udp 2122260223 192.168.4.1 51846 typ host generation 0 network-id 1
a=candidate:4157144802 1 udp 2122194687 192.168.188.1 51847 typ host generation 0 network-id 2
a=candidate:1928035339 1 udp 2122134271 fd58:2af7:13b5:ae00:15f:53c0:51a1:a42c 51848 typ host generation 0 network-id 4 network-cost 10
a=candidate:3331753358 1 udp 2122068735 fd58:2af7:13b5:ae00:21ff:17b9:b8f0:5546 51849 typ host generation 0 network-id 5 network-cost 10
a=candidate:2795255774 1 udp 2121998079 192.168.1.7 51850 typ host generation 0 network-id 3 network-cost 10
a=candidate:2646148417 1 tcp 1518280447 192.168.4.1 9 typ host tcptype active generation 0 network-id 1
a=candidate:3108700690 1 tcp 1518214911 192.168.188.1 9 typ host tcptype active generation 0 network-id 2
a=candidate:1013481723 1 tcp 1518154495 fd58:2af7:13b5:ae00:15f:53c0:51a1:a42c 9 typ host tcptype active generation 0 network-id 4 network-cost 10
a=candidate:2283108734 1 tcp 1518088959 fd58:2af7:13b5:ae00:21ff:17b9:b8f0:5546 9 typ host tcptype active generation 0 network-id 5 network-cost 10
a=candidate:3894397742 1 tcp 1518018303 192.168.1.7 9 typ host tcptype active generation 0 network-id 3 network-cost 10
a=candidate:264484875 1 udp 1685790463 41.35.175.27 51850 typ srflx raddr 192.168.1.7 rport 51850 generation 0 network-id 3 network-cost 10
a=ice-ufrag:Shii
a=ice-pwd:NJge11Kp+Ex1dZURkAmckb4o
a=ice-options:trickle
a=fingerprint:sha-256 E8:02:69:26:EB:15:16:1A:66:9A:6C:D4:FB:70:B3:12:56:DD:68:1E:5F:0F:63:A1:B1:13:FC:6B:4A:FE:D5:DF
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:3567731579 cname:yQrVyNYvvbbKQeH1
a=ssrc:3567731579 msid:VNq9ACBHIOK0YcfWUlAAGaUahtigoIX0Ijme bc13c095-9841-4fb9-b414-3650907ec1e6
a=ssrc:3567731579 mslabel:VNq9ACBHIOK0YcfWUlAAGaUahtigoIX0Ijme
a=ssrc:3567731579 label:bc13c095-9841-4fb9-b414-3650907ec1e6

<— Transmitting SIP response (657 bytes) to UDP:188.165.231.30:10060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 188.165.231.30:10060;rport=10060;received=188.165.231.30;branch=z9hG4bK76l7flGtVogAZoLNsmWUg6zs1mIR7wpT
Via: SIP/2.0/TCP 41.35.175.27:52467;rport;branch=z9hG4bK76l7flGtVogAZoLNsmWUg6zs1mIR7wpT;ws-hacked=WSS
Call-ID: f45b457f-fa04-b587-4424-b5ae9fc6e15c
From: “web2” sip:web2@pbxbox.net;tag=VqH9t01oTl5y9BXvBc3M
To: sip:web1@pbxbox.net;tag=z9hG4bK76l7flGtVogAZoLNsmWUg6zs1mIR7wpT
CSeq: 28015 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1513200736/4b9851d85d4f34b09e7a2098d775f3e4”,opaque=“1497b18d3735e860”,algorithm=md5,qop="auth"
Server: Asterisk PBX 15.1.3
Content-Length: 0

<— Received SIP request (371 bytes) from UDP:188.165.231.30:10060 —>
ACK sip:web1@pbxbox.net SIP/2.0
Via: SIP/2.0/UDP 188.165.231.30:10060;branch=z9hG4bK76l7flGtVogAZoLNsmWUg6zs1mIR7wpT;rport
From: "web2"sip:web2@pbxbox.net;tag=VqH9t01oTl5y9BXvBc3M
To: sip:web1@pbxbox.net;tag=z9hG4bK76l7flGtVogAZoLNsmWUg6zs1mIR7wpT
Call-ID: f45b457f-fa04-b587-4424-b5ae9fc6e15c
CSeq: 28015 ACK
Content-Length: 0
Max-Forwards: 70

<— Received SIP request (475 bytes) from UDP:188.165.231.30:10060 —>
ACK sip:web1@pbxbox.net SIP/2.0
Via: SIP/2.0/UDP 188.165.231.30:10060;branch=z9hG4bK76l7flGtVogAZoLNsmWUg6zs1mIR7wpT;rport
From: "web2"sip:web2@pbxbox.net;tag=VqH9t01oTl5y9BXvBc3M
To: sip:web1@pbxbox.net;tag=z9hG4bK76l7flGtVogAZoLNsmWUg6zs1mIR7wpT
Call-ID: f45b457f-fa04-b587-4424-b5ae9fc6e15c
CSeq: 28015 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP 41.35.175.27:52467;rport;branch=z9hG4bK76l7flGtVogAZoLNsmWUg6zs1mIR7wpT;ws-hacked=WSS

<— Received SIP request (3652 bytes) from UDP:188.165.231.30:10060 —>
INVITE sip:web1@pbxbox.net SIP/2.0
Via: SIP/2.0/UDP 188.165.231.30:10060;branch=z9hG4bKgUNRnu97GiHqppTJJSetdH5QFNyISbGZ;rport
From: "web2"sip:web2@pbxbox.net;tag=VqH9t01oTl5y9BXvBc3M
To: sip:web1@pbxbox.net
Contact: "web2"sip:web2@188.165.231.30:10060;rtcweb-breaker=no;click2call=no;transport=udp;ws-src-ip=41.35.175.27;ws-src-port=52467;ws-src-proto=wss;+g.oma.sip-im;language="en,fr"
Call-ID: f45b457f-fa04-b587-4424-b5ae9fc6e15c
CSeq: 28016 INVITE
Content-Type: application/sdp
Content-Length: 2630
Max-Forwards: 70
Authorization: Digest username=“web2”,realm=“asterisk”,nonce=“1513200736/4b9851d85d4f34b09e7a2098d775f3e4”,uri=“sip:web1@pbxbox.net”,response=“eb1763982c7e6022bacb9f52fd58d281”,algorithm=md5,cnonce=“701756cbb68639f1e78278ffb1c018c4”,opaque=“1497b18d3735e860”,qop=auth,nc=00000001
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom
Via: SIP/2.0/TCP 41.35.175.27:52467;rport;branch=z9hG4bKgUNRnu97GiHqppTJJSetdH5QFNyISbGZ;ws-hacked=WSS

v=0
o=- 2367316182999042600 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS VNq9ACBHIOK0YcfWUlAAGaUahtigoIX0Ijme
m=audio 51850 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 41.35.175.27
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:3543857585 1 udp 2122260223 192.168.4.1 51846 typ host generation 0 network-id 1
a=candidate:4157144802 1 udp 2122194687 192.168.188.1 51847 typ host generation 0 network-id 2
a=candidate:1928035339 1 udp 2122134271 fd58:2af7:13b5:ae00:15f:53c0:51a1:a42c 51848 typ host generation 0 network-id 4 network-cost 10
a=candidate:3331753358 1 udp 2122068735 fd58:2af7:13b5:ae00:21ff:17b9:b8f0:5546 51849 typ host generation 0 network-id 5 network-cost 10
a=candidate:2795255774 1 udp 2121998079 192.168.1.7 51850 typ host generation 0 network-id 3 network-cost 10
a=candidate:2646148417 1 tcp 1518280447 192.168.4.1 9 typ host tcptype active generation 0 network-id 1
a=candidate:3108700690 1 tcp 1518214911 192.168.188.1 9 typ host tcptype active generation 0 network-id 2
a=candidate:1013481723 1 tcp 1518154495 fd58:2af7:13b5:ae00:15f:53c0:51a1:a42c 9 typ host tcptype active generation 0 network-id 4 network-cost 10
a=candidate:2283108734 1 tcp 1518088959 fd58:2af7:13b5:ae00:21ff:17b9:b8f0:5546 9 typ host tcptype active generation 0 network-id 5 network-cost 10
a=candidate:3894397742 1 tcp 1518018303 192.168.1.7 9 typ host tcptype active generation 0 network-id 3 network-cost 10
a=candidate:264484875 1 udp 1685790463 41.35.175.27 51850 typ srflx raddr 192.168.1.7 rport 51850 generation 0 network-id 3 network-cost 10
a=ice-ufrag:Shii
a=ice-pwd:NJge11Kp+Ex1dZURkAmckb4o
a=ice-options:trickle
a=fingerprint:sha-256 E8:02:69:26:EB:15:16:1A:66:9A:6C:D4:FB:70:B3:12:56:DD:68:1E:5F:0F:63:A1:B1:13:FC:6B:4A:FE:D5:DF
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:3567731579 cname:yQrVyNYvvbbKQeH1
a=ssrc:3567731579 msid:VNq9ACBHIOK0YcfWUlAAGaUahtigoIX0Ijme bc13c095-9841-4fb9-b414-3650907ec1e6
a=ssrc:3567731579 mslabel:VNq9ACBHIOK0YcfWUlAAGaUahtigoIX0Ijme
a=ssrc:3567731579 label:bc13c095-9841-4fb9-b414-3650907ec1e6

== Setting global variable ‘SIPDOMAIN’ to ‘pbxbox.net’
<— Transmitting SIP response (460 bytes) to UDP:188.165.231.30:10060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 188.165.231.30:10060;rport=10060;received=188.165.231.30;branch=z9hG4bKgUNRnu97GiHqppTJJSetdH5QFNyISbGZ
Via: SIP/2.0/TCP 41.35.175.27:52467;rport;branch=z9hG4bKgUNRnu97GiHqppTJJSetdH5QFNyISbGZ;ws-hacked=WSS
Call-ID: f45b457f-fa04-b587-4424-b5ae9fc6e15c
From: “web2” sip:web2@pbxbox.net;tag=VqH9t01oTl5y9BXvBc3M
To: sip:web1@pbxbox.net
CSeq: 28016 INVITE
Server: Asterisk PBX 15.1.3
Content-Length: 0

== DTLS ECDH initialized (automatic), faster PFS enabled
– Executing [web1@main:1] NoOp(“PJSIP/web2-00000006”, “Call from webrtc client to: web1”) in new stack
– Executing [web1@main:2] Set(“PJSIP/web2-00000006”, “_SIPSRTP_CRYPTO=enable”) in new stack
– Executing [web1@main:3] Set(“PJSIP/web2-00000006”, “CHANNEL(secure_bridge_signaling)=1”) in new stack
– Executing [web1@main:4] Set(“PJSIP/web2-00000006”, “CHANNEL(secure_bridge_media)=1”) in new stack
– Executing [web1@main:5] Dial(“PJSIP/web2-00000006”, “PJSIP/web1”) in new stack
[Dec 13 15:32:16] WARNING[13873][C-00000004]: channel.c:6141 request_channel: Setting security requirements failed
[Dec 13 15:32:16] WARNING[13873][C-00000004]: app_dial.c:2510 dial_exec_full: Unable to create channel of type ‘PJSIP’ (cause 58 - Bearer capability not available)
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [web1@main:6] Hangup(“PJSIP/web2-00000006”, “”) in new stack
== Spawn extension (main, web1, 6) exited non-zero on ‘PJSIP/web2-00000006’
<— Transmitting SIP response (538 bytes) to UDP:188.165.231.30:10060 —>
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 188.165.231.30:10060;rport=10060;received=188.165.231.30;branch=z9hG4bKgUNRnu97GiHqppTJJSetdH5QFNyISbGZ
Via: SIP/2.0/TCP 41.35.175.27:52467;rport;branch=z9hG4bKgUNRnu97GiHqppTJJSetdH5QFNyISbGZ;ws-hacked=WSS
Call-ID: f45b457f-fa04-b587-4424-b5ae9fc6e15c
From: “web2” sip:web2@pbxbox.net;tag=VqH9t01oTl5y9BXvBc3M
To: sip:web1@pbxbox.net;tag=ffd76243-b7d7-4187-9688-9677f94e2bcb
CSeq: 28016 INVITE
Server: Asterisk PBX 15.1.3
Reason: Q.850;cause=58
Content-Length: 0

<— Received SIP request (368 bytes) from UDP:188.165.231.30:10060 —>
ACK sip:web1@pbxbox.net SIP/2.0
Via: SIP/2.0/UDP 188.165.231.30:10060;branch=z9hG4bKgUNRnu97GiHqppTJJSetdH5QFNyISbGZ;rport
From: "web2"sip:web2@pbxbox.net;tag=VqH9t01oTl5y9BXvBc3M
To: sip:web1@pbxbox.net;tag=ffd76243-b7d7-4187-9688-9677f94e2bcb
Call-ID: f45b457f-fa04-b587-4424-b5ae9fc6e15c
CSeq: 28016 ACK
Content-Length: 0
Max-Forwards: 70

<— Received SIP request (472 bytes) from UDP:188.165.231.30:10060 —>
ACK sip:web1@pbxbox.net SIP/2.0
Via: SIP/2.0/UDP 188.165.231.30:10060;branch=z9hG4bKgUNRnu97GiHqppTJJSetdH5QFNyISbGZ;rport
From: "web2"sip:web2@pbxbox.net;tag=VqH9t01oTl5y9BXvBc3M
To: sip:web1@pbxbox.net;tag=ffd76243-b7d7-4187-9688-9677f94e2bcb
Call-ID: f45b457f-fa04-b587-4424-b5ae9fc6e15c
CSeq: 28016 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP 41.35.175.27:52467;rport;branch=z9hG4bKgUNRnu97GiHqppTJJSetdH5QFNyISbGZ;ws-hacked=WSS

<— Received SIP request (475 bytes) from UDP:188.165.231.30:10060 —>
ACK sip:web1@pbxbox.net SIP/2.0
Via: SIP/2.0/UDP 188.165.231.30:10060;branch=z9hG4bK76l7flGtVogAZoLNsmWUg6zs1mIR7wpT;rport
From: "web2"sip:web2@pbxbox.net;tag=VqH9t01oTl5y9BXvBc3M
To: sip:web1@pbxbox.net;tag=z9hG4bK76l7flGtVogAZoLNsmWUg6zs1mIR7wpT
Call-ID: f45b457f-fa04-b587-4424-b5ae9fc6e15c
CSeq: 28015 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP 41.35.175.27:52467;rport;branch=z9hG4bK76l7flGtVogAZoLNsmWUg6zs1mIR7wpT;ws-hacked=WSS

<— Received SIP request (472 bytes) from UDP:188.165.231.30:10060 —>
ACK sip:web1@pbxbox.net SIP/2.0
Via: SIP/2.0/UDP 188.165.231.30:10060;branch=z9hG4bKgUNRnu97GiHqppTJJSetdH5QFNyISbGZ;rport
From: "web2"sip:web2@pbxbox.net;tag=VqH9t01oTl5y9BXvBc3M
To: sip:web1@pbxbox.net;tag=ffd76243-b7d7-4187-9688-9677f94e2bcb
Call-ID: f45b457f-fa04-b587-4424-b5ae9fc6e15c
CSeq: 28016 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP 41.35.175.27:52467;rport;branch=z9hG4bKgUNRnu97GiHqppTJJSetdH5QFNyISbGZ;ws-hacked=WSS

<— Received SIP request (475 bytes) from UDP:188.165.231.30:10060 —>
ACK sip:web1@pbxbox.net SIP/2.0
Via: SIP/2.0/UDP 188.165.231.30:10060;branch=z9hG4bK76l7flGtVogAZoLNsmWUg6zs1mIR7wpT;rport
From: "web2"sip:web2@pbxbox.net;tag=VqH9t01oTl5y9BXvBc3M
To: sip:web1@pbxbox.net;tag=z9hG4bK76l7flGtVogAZoLNsmWUg6zs1mIR7wpT
Call-ID: f45b457f-fa04-b587-4424-b5ae9fc6e15c
CSeq: 28015 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP 41.35.175.27:52467;rport;branch=z9hG4bK76l7flGtVogAZoLNsmWUg6zs1mIR7wpT;ws-hacked=WSS

<— Received SIP request (472 bytes) from UDP:188.165.231.30:10060 —>
ACK sip:web1@pbxbox.net SIP/2.0
Via: SIP/2.0/UDP 188.165.231.30:10060;branch=z9hG4bKgUNRnu97GiHqppTJJSetdH5QFNyISbGZ;rport
From: "web2"sip:web2@pbxbox.net;tag=VqH9t01oTl5y9BXvBc3M
To: sip:web1@pbxbox.net;tag=ffd76243-b7d7-4187-9688-9677f94e2bcb
Call-ID: f45b457f-fa04-b587-4424-b5ae9fc6e15c
CSeq: 28016 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP 41.35.175.27:52467;rport;branch=z9hG4bKgUNRnu97GiHqppTJJSetdH5QFNyISbGZ;ws-hacked=WSS

<— Received SIP request (475 bytes) from UDP:188.165.231.30:10060 —>
ACK sip:web1@pbxbox.net SIP/2.0
Via: SIP/2.0/UDP 188.165.231.30:10060;branch=z9hG4bK76l7flGtVogAZoLNsmWUg6zs1mIR7wpT;rport
From: "web2"sip:web2@pbxbox.net;tag=VqH9t01oTl5y9BXvBc3M
To: sip:web1@pbxbox.net;tag=z9hG4bK76l7flGtVogAZoLNsmWUg6zs1mIR7wpT
Call-ID: f45b457f-fa04-b587-4424-b5ae9fc6e15c
CSeq: 28015 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP 41.35.175.27:52467;rport;branch=z9hG4bK76l7flGtVogAZoLNsmWUg6zs1mIR7wpT;ws-hacked=WSS

<— Received SIP request (472 bytes) from UDP:188.165.231.30:10060 —>
ACK sip:web1@pbxbox.net SIP/2.0
Via: SIP/2.0/UDP 188.165.231.30:10060;branch=z9hG4bKgUNRnu97GiHqppTJJSetdH5QFNyISbGZ;rport
From: "web2"sip:web2@pbxbox.net;tag=VqH9t01oTl5y9BXvBc3M
To: sip:web1@pbxbox.net;tag=ffd76243-b7d7-4187-9688-9677f94e2bcb
Call-ID: f45b457f-fa04-b587-4424-b5ae9fc6e15c
CSeq: 28016 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP 41.35.175.27:52467;rport;branch=z9hG4bKgUNRnu97GiHqppTJJSetdH5QFNyISbGZ;ws-hacked=WSS

I am using wss on port 8089. Can anyone help?
Thanks
Alan

rmudgett · December 15, 2017, 9:21pm

There’s the problem. chan_pjsip does not support the CHANNEL(secure_bridge_xxx) method of setting up a secure connection. The endpoint and transport configuration setup secure connections for pjsip channels.

Alan_Demerda · December 15, 2017, 10:20pm

Thanks for your reply.

I am using this config in extensions.conf:

; Peers
exten => _web.,1,NoOp(Call from webrtc client to: ${EXTEN})
exten => _web.,n,Set(_SIPSRTP_CRYPTO=enable)
exten => _web.,n,Set(CHANNEL(secure_bridge_signaling)=1)
exten => _web.,n,Set(CHANNEL(secure_bridge_media)=1)
exten => _web.,n,Dial(PJSIP/${EXTEN})
;exten => _web.,n,Confbridge(${EXTEN})
exten => _web.,n,HangUp()

What should i do instead?

Thanks
Alan

rmudgett · December 15, 2017, 11:08pm

Remove these lines. Setting those values is not supported by chan_pjsip and is preventing the pjsip channel from being created. The pjsip endpoint just needs to be configured to do encryption.

Alan_Demerda · December 15, 2017, 11:11pm

here is what i got:

v1*CLI>
== DTLS ECDH initialized (automatic), faster PFS enabled
== DTLS ECDH initialized (automatic), faster PFS enabled
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
> 0x7f0f98018af0 – Strict RTP learning after remote address set to: 41.35.175.27:53338
– Executing [web2@main:1] NoOp(“SIP/web1-00000000”, “Call from webrtc client to: web2”) in new stack
– Executing [web2@main:2] Set(“SIP/web1-00000000”, “_SIPSRTP_CRYPTO=enable”) in new stack
– Executing [web2@main:3] Dial(“SIP/web1-00000000”, “PJSIP/web2”) in new stack
[Dec 15 17:10:16] ERROR[26344]: chan_pjsip.c:2413 request: Unable to create PJSIP channel with empty endpoint name
[Dec 15 17:10:16] WARNING[26412][C-00000001]: app_dial.c:2510 dial_exec_full: Unable to create channel of type ‘PJSIP’ (cause 6 - Channel unacceptable)
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [web2@main:4] Hangup(“SIP/web1-00000000”, “”) in new stack
== Spawn extension (main, web2, 4) exited non-zero on ‘SIP/web1-00000000’

rmudgett · December 16, 2017, 12:19am

This message is wrong and misleading. After looking at the code, you don’t have an endpoint named web2 even configured in pjsip.conf.

You also appear to be trying to use chan_sip and chan_pjsip at the same time. It would be better to get things working with one channel technology or the other. Using both at the same time makes the dialplan complicated because the two technologies do things differently. As a result, your dialplan has to be aware of which channel technology it is dealing with.

Alan_Demerda · December 16, 2017, 9:39am

Sorry but i am confused now, should i put the config of web1 and web2 either in sip.conf or pjsip.conf? The existing situation is i am putting in both as below:

sip.conf:

`[web1]
type=friend
host=dynamic
secret=web1
encryption=yes
avpf=yes
icesupport=yes
context=main
directmedia=no
transport=wss
force_avp=yes
dtlsenable=yes
dtlsverify=no
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlsprivatekey=/etc/asterisk/keys/asterisk.key
dtlssetup=actpass
rtcp_mux=yes
videosupport=yes

[web2]
type=friend
host=dynamic
secret=web2
encryption=yes
avpf=yes
icesupport=yes
context=main
directmedia=no
transport=wss
force_avp=yes
dtlsenable=yes
dtlsverify=no
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlsprivatekey=/etc/asterisk/keys/asterisk.key
dtlssetup=actpass
rtcp_mux=yes
videosupport=yes`

and this is pjsip.conf:

[web1]
type=aor
max_contacts=1

[web1]
type=auth
username=web1
password=web1

[web1]
type=endpoint
aors=web1
auth=web1
host=dynamic
use_avpf=yes
media_encryption=dtls
dtls_ca_file=/etc/asterisk/keys/ca.crt
dtls_cert_file=/etc/asterisk/keys/asterisk.pem
dtls_verify=fingerprint
dtls_setup=actpass
ice_support=yes
media_use_received_transport=yes
rtcp_mux=yes
context=main
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=h264

[web2]
type=aor
max_contacts=1

[web2]
type=auth
username=web2
password=web2

[web2]
type=endpoint
aors=web2
auth=web2
host=dynamic
use_avpf=yes
media_encryption=dtls
dtls_ca_file=/etc/asterisk/keys/ca.crt
dtls_cert_file=/etc/asterisk/keys/asterisk.pem
dtls_verify=fingerprint
dtls_setup=actpass
ice_support=yes
media_use_received_transport=yes
rtcp_mux=yes
context=main
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=h264

Please tell me what to remove or edit.

jcolp · December 16, 2017, 11:40am

Which one do you want to use going forward? Take the configuration out of the other one.

Alan_Demerda · December 16, 2017, 1:36pm

I went forward with sip.conf and removed the config from pjsip.conf, then i re-edited extensions.conf accordingly and everything went fine.

Thanks a lot for your help.

Appreciated!
Alan

Alan_Demerda · December 17, 2017, 10:26am

Sorry for coming again but when i do the pjsip and remove the config from sip.conf it cannot recieve the calls on pjsip channel and i get this error

v1*CLI>
== WebSocket connection from ‘41.33.108.226:54271’ closed
== WebSocket connection from ‘41.33.108.226:28070’ for protocol ‘sip’ accepted using version ‘13’
[Dec 17 04:19:08] NOTICE[30570]: chan_sip.c:28684 handle_request_register: Registration from ‘"web2"sip:web2@x1.pbxbox.net’ failed for ‘41.33.108.226:28070’ - Wrong password

I am using the same password as before so why i cannot connect with pjsip channel?

david551 · December 17, 2017, 10:43am

You seem to be assuming that SIP and PJSIP are different protocols; they are not. You can use SIP or PJSIP to connect to a remote SIP system whether or not that system is implemented using the PJSIP code.

What rarely makes sense is having both chan_sip and chan_pjsip implementations of of SIP installed on Asterisk.

(It is possible that you remote system requires SIP features only implemented by PJSIP, but then hte normal approach would be to use chan_pjsip for all your SIP peers.

jcolp · December 17, 2017, 11:43am

Due to the load order it appears as though chan_sip is loading before chan_pjsip and as a result it is getting the WebRTC traffic. If you aren’t going to use chan_sip then you can add “noload => chan_sip.so” to your modules.conf and then PJSIP will be used.

Alan_Demerda · December 17, 2017, 1:03pm

ok that worked out, the call is setup but no audio at all, on the other hand if web1 and web2 call the conference number 911 then the voice is good and clear. here is my extensions.conf:

; Conference calls
exten => _9X.,1,NoOp(Call from to Conference ID: ${EXTEN})
exten => _9X.,n,Set(_SIPSRTP_CRYPTO=enable)
exten => _9X.,n,Confbridge(${EXTEN})
exten => _9X.,n,HangUp()

; Peers
exten => _web.,1,NoOp(Call from webrtc client to: ${EXTEN})
exten => _web.,n,Set(_SIPSRTP_CRYPTO=enable)
exten => _web.,n,Dial(PJSIP/${EXTEN})
exten => _web.,n,HangUp()

here is the log from the console:

v1*CLI> 
  == Setting global variable 'SIPDOMAIN' to 'vd.pbxbox.net'
  == DTLS ECDH initialized (automatic), faster PFS enabled
    -- Executing [web2@main:1] NoOp("PJSIP/web1-00000000", "Call from webrtc client to: web2") in new stack
    -- Executing [web2@main:2] Set("PJSIP/web1-00000000", "_SIPSRTP_CRYPTO=enable") in new stack
    -- Executing [web2@main:3] Dial("PJSIP/web1-00000000", "PJSIP/web2") in new stack
    -- Called PJSIP/web2
  == DTLS ECDH initialized (automatic), faster PFS enabled
    -- PJSIP/web2-00000001 is ringing
       > 0x2e458b0 -- Strict RTP learning after remote address set to: 41.33.108.226:7522
    -- PJSIP/web2-00000001 answered PJSIP/web1-00000000
       > 0x2e39200 -- Strict RTP learning after remote address set to: 41.33.108.226:38568
    -- Channel PJSIP/web2-00000001 joined 'simple_bridge' basic-bridge <bc0ba62e-b734-4d23-84f9-36ab15e41c7f>
    -- Channel PJSIP/web1-00000000 joined 'simple_bridge' basic-bridge <bc0ba62e-b734-4d23-84f9-36ab15e41c7f>
       > 0x2e39200 -- Strict RTP learning after ICE completion
       > 0x2e39200 -- Strict RTP switching source address to 193.182.144.174:57833
       > 0x2e39200 -- Strict RTP learning complete - Locking on source address 193.182.144.174:57833
       > 0x2e458b0 -- Strict RTP learning after ICE completion
v1*CLI>

Alan_Demerda · December 21, 2017, 7:38am

My question in brief: why simple bridge is not working while ConBridge is working fine?

jcolp · December 21, 2017, 10:53am

You’ll need to do further debugging to see why. A simple bridge merely forwards media it receives. It doesn’t generate any on its own. A confbridge, however, is always feeding a constant stream. Those are the difference. Have you checked “rtp set debug on” to see if Asterisk is getting and forwarding media in the simple bridge case? Have you looked at the WebRTC internals in the browser to see if it is showing sending and receiving?

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