The detailed logs with pjsip set logger on
:
[Jan 21 20:26:46] ERROR[459]: res_pjsip_session.c:937 handle_incoming_sdp: 1002: Couldn't negotiate stream 0:audio-0:audio:sendrecv (nothing)
-- Executing [0612345678@work:1] Dial("PJSIP/1002-00000048", "PJSIP/0612345678@trunk-sfr") in new stack
-- Called PJSIP/0612345678@trunk-sfr
[Jan 21 20:26:47] WARNING[459]: res_pjsip_outbound_authenticator_digest.c:507 digest_create_request_with_auth: Endpoint: 'trunk-sfr': No auth objects matching realm(s) '' from challenge found.
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'PJSIP/1002-00000048' status is 'CHANUNAVAIL'
router3*CLI> pjsip set logger on
PJSIP Logging enabled
<--- Received SIP request (1165 bytes) from UDP:192.168.3.173:5060 --->
INVITE sip:0612345678@domain.local:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.173:5060;branch=z9hG4bK.ZUxlhrt0v;rport
From: <sip:1001@domain.local>;tag=kHQwvy0k5
To: sip:0612345678@domain.local
CSeq: 20 INVITE
Call-ID: 4TLbNc67YF
Max-Forwards: 70
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 495
Contact: <sip:1001@192.168.3.173;transport=udp>;expires=3599;+sip.instance="<urn:uuid:af16bccc-f148-00fe-91eb-78c65b9d3d31>"
User-Agent: Linphone Desktop/ (Ubuntu 22.04.1 LTS, Qt 5.15.3) LinphoneCore/4.4.21
v=0
o=1002 3348 599 IN IP4 192.168.3.173
s=Talk
c=IN IP4 192.168.3.173
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVPF 96 97 98 0 8 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr
<--- Transmitting SIP response (474 bytes) to UDP:192.168.3.173:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.3.173:5060;rport=5060;received=192.168.3.173;branch=z9hG4bK.ZUxlhrt0v
Call-ID: 4TLbNc67YF
From: <sip:1001@domain.local>;tag=kHQwvy0k5
To: <sip:0612345678@domain.local>;tag=z9hG4bK.ZUxlhrt0v
CSeq: 20 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1674332930/df807965872c3191c1685c3a733f96d8",opaque="5c8b7332194275dc",algorithm=MD5,qop="auth"
Server: Asterisk PBX 18.15.0
Content-Length: 0
<--- Received SIP request (418 bytes) from UDP:192.168.3.173:5060 --->
ACK sip:0612345678@domain.local:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.173:5060;branch=z9hG4bK.ZUxlhrt0v;rport
Call-ID: 4TLbNc67YF
From: <sip:1001@domain.local>;tag=kHQwvy0k5
To: <sip:0612345678@domain.local>;tag=z9hG4bK.ZUxlhrt0v
Contact: <sip:1001@192.168.3.173;transport=udp>;expires=3599;+sip.instance="<urn:uuid:af16bccc-f148-00fe-91eb-78c65b9d3d31>"
Max-Forwards: 70
CSeq: 20 ACK
<--- Received SIP request (1461 bytes) from UDP:192.168.3.173:5060 --->
INVITE sip:0612345678@domain.local:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.173:5060;branch=z9hG4bK.aknG4Xhac;rport
From: <sip:1001@domain.local>;tag=kHQwvy0k5
To: sip:0612345678@domain.local
CSeq: 21 INVITE
Call-ID: 4TLbNc67YF
Max-Forwards: 70
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 495
Contact: <sip:1001@192.168.3.173;transport=udp>;expires=3599;+sip.instance="<urn:uuid:af16bccc-f148-00fe-91eb-78c65b9d3d31>"
User-Agent: Linphone Desktop/ (Ubuntu 22.04.1 LTS, Qt 5.15.3) LinphoneCore/4.4.21
Authorization: Digest realm="asterisk", nonce="1674332930/df807965872c3191c1685c3a733f96d8", algorithm=MD5, opaque="5c8b7332194275dc", username="1002", uri="sip:0612345678@domain.local:5060", response="40e82d2720974dc417430f3f582b52b2", cnonce="V3qiXuvH7f2skqiR", nc=00000001, qop=auth
v=0
o=1002 3348 599 IN IP4 192.168.3.173
s=Talk
c=IN IP4 192.168.3.173
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVPF 96 97 98 0 8 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr
<--- Transmitting SIP response (300 bytes) to UDP:192.168.3.173:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.3.173:5060;rport=5060;received=192.168.3.173;branch=z9hG4bK.aknG4Xhac
Call-ID: 4TLbNc67YF
From: <sip:1001@domain.local>;tag=kHQwvy0k5
To: <sip:0612345678@domain.local>
CSeq: 21 INVITE
Server: Asterisk PBX 18.15.0
Content-Length: 0
[Jan 21 20:28:50] ERROR[470]: res_pjsip_session.c:937 handle_incoming_sdp: 1002: Couldn't negotiate stream 0:audio-0:audio:sendrecv (nothing)
<--- Transmitting SIP response (350 bytes) to UDP:192.168.3.173:5060 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 192.168.3.173:5060;rport=5060;received=192.168.3.173;branch=z9hG4bK.aknG4Xhac
Call-ID: 4TLbNc67YF
From: <sip:1001@domain.local>;tag=kHQwvy0k5
To: <sip:0612345678@domain.local>;tag=6X0ZD4BQOU.RQH4OCMm6GSb70gsekUi.
CSeq: 21 INVITE
Server: Asterisk PBX 18.15.0
Content-Length: 0
<--- Received SIP request (433 bytes) from UDP:192.168.3.173:5060 --->
ACK sip:0612345678@domain.local:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.173:5060;branch=z9hG4bK.aknG4Xhac;rport
Call-ID: 4TLbNc67YF
From: <sip:1001@domain.local>;tag=kHQwvy0k5
To: <sip:0612345678@domain.local>;tag=6X0ZD4BQOU.RQH4OCMm6GSb70gsekUi.
Contact: <sip:1001@192.168.3.173;transport=udp>;expires=3599;+sip.instance="<urn:uuid:af16bccc-f148-00fe-91eb-78c65b9d3d31>"
Max-Forwards: 70
CSeq: 21 ACK
<--- Received SIP request (1164 bytes) from UDP:192.168.3.173:5060 --->
INVITE sip:0612345678@domain.local:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.173:5060;branch=z9hG4bK.LTKWgc1dQ;rport
From: <sip:1001@domain.local>;tag=tGlzd585N
To: sip:0612345678@domain.local
CSeq: 20 INVITE
Call-ID: w~ARmS6~3G
Max-Forwards: 70
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 494
Contact: <sip:1001@192.168.3.173;transport=udp>;expires=3599;+sip.instance="<urn:uuid:af16bccc-f148-00fe-91eb-78c65b9d3d31>"
User-Agent: Linphone Desktop/ (Ubuntu 22.04.1 LTS, Qt 5.15.3) LinphoneCore/4.4.21
v=0
o=1002 3348 600 IN IP4 192.168.3.173
s=Talk
c=IN IP4 192.168.3.173
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 96 97 98 0 8 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr
<--- Transmitting SIP response (474 bytes) to UDP:192.168.3.173:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.3.173:5060;rport=5060;received=192.168.3.173;branch=z9hG4bK.LTKWgc1dQ
Call-ID: w~ARmS6~3G
From: <sip:1001@domain.local>;tag=tGlzd585N
To: <sip:0612345678@domain.local>;tag=z9hG4bK.LTKWgc1dQ
CSeq: 20 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1674332930/df807965872c3191c1685c3a733f96d8",opaque="0687bf9f261c9f3d",algorithm=MD5,qop="auth"
Server: Asterisk PBX 18.15.0
Content-Length: 0
<--- Received SIP request (418 bytes) from UDP:192.168.3.173:5060 --->
ACK sip:0612345678@domain.local:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.173:5060;branch=z9hG4bK.LTKWgc1dQ;rport
Call-ID: w~ARmS6~3G
From: <sip:1001@domain.local>;tag=tGlzd585N
To: <sip:0612345678@domain.local>;tag=z9hG4bK.LTKWgc1dQ
Contact: <sip:1001@192.168.3.173;transport=udp>;expires=3599;+sip.instance="<urn:uuid:af16bccc-f148-00fe-91eb-78c65b9d3d31>"
Max-Forwards: 70
CSeq: 20 ACK
<--- Received SIP request (1460 bytes) from UDP:192.168.3.173:5060 --->
INVITE sip:0612345678@domain.local:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.173:5060;branch=z9hG4bK.uUBBaJGQ1;rport
From: <sip:1001@domain.local>;tag=tGlzd585N
To: sip:0612345678@domain.local
CSeq: 21 INVITE
Call-ID: w~ARmS6~3G
Max-Forwards: 70
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 494
Contact: <sip:1001@192.168.3.173;transport=udp>;expires=3599;+sip.instance="<urn:uuid:af16bccc-f148-00fe-91eb-78c65b9d3d31>"
User-Agent: Linphone Desktop/ (Ubuntu 22.04.1 LTS, Qt 5.15.3) LinphoneCore/4.4.21
Authorization: Digest realm="asterisk", nonce="1674332930/df807965872c3191c1685c3a733f96d8", algorithm=MD5, opaque="0687bf9f261c9f3d", username="1002", uri="sip:0612345678@domain.local:5060", response="bf4421cff9b28229f9b928c9a7a06528", cnonce="5JnwzOzN2e790UHz", nc=00000001, qop=auth
v=0
o=1002 3348 600 IN IP4 192.168.3.173
s=Talk
c=IN IP4 192.168.3.173
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 96 97 98 0 8 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr
<--- Transmitting SIP response (300 bytes) to UDP:192.168.3.173:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.3.173:5060;rport=5060;received=192.168.3.173;branch=z9hG4bK.uUBBaJGQ1
Call-ID: w~ARmS6~3G
From: <sip:1001@domain.local>;tag=tGlzd585N
To: <sip:0612345678@domain.local>
CSeq: 21 INVITE
Server: Asterisk PBX 18.15.0
Content-Length: 0
-- Executing [0612345678@work:1] Dial("PJSIP/1002-0000004a", "PJSIP/0612345678@trunk-sfr") in new stack
-- Called PJSIP/0612345678@trunk-sfr
<--- Transmitting SIP request (1132 bytes) to UDP:92.91.179.40:5062 --->
INVITE sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org SIP/2.0
Via: SIP/2.0/UDP 93.4.216.9:5060;rport;branch=z9hG4bKPjrbz92zKKeWKGLuj2-X4WF0vPlpecPpo2
From: <sip:+33287654321@ims.mnc010.mcc208.3gppnetwork.org>;tag=hCHqGKT2rc0KomF9-.lBvKeaT6oTKtWR
To: <sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org>
Contact: <sip:+33287654321@93.4.216.9:5060>
Call-ID: 5UrSOYtvmclD2JBK3SFYEadnlj6WSRZZ
CSeq: 18726 INVITE
Route: <sip:corbas.p-cscf.sfr.net:5062;lr>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.0
Content-Type: application/sdp
Content-Length: 361
v=0
o=- 906875053 906875053 IN IP4 93.4.216.9
s=Asterisk
c=IN IP4 93.4.216.9
t=0 0
m=audio 10044 RTP/AVP 0 107 110 9 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:110 speex/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv
<--- Received SIP response (430 bytes) from UDP:92.91.179.40:5062 --->
SIP/2.0 100 Trying
Call-ID: 5UrSOYtvmclD2JBK3SFYEadnlj6WSRZZ
Via: SIP/2.0/UDP 93.4.216.9:5060;received=93.4.216.9;branch=z9hG4bKPjrbz92zKKeWKGLuj2-X4WF0vPlpecPpo2;rport=5060
To: <sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org;user=phone>
From: <sip:+33287654321@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=hCHqGKT2rc0KomF9-.lBvKeaT6oTKtWR
CSeq: 18726 INVITE
Date: Sat, 21 Jan 2023 20:28:37 GMT
Content-Length: 0
<--- Received SIP response (1104 bytes) from UDP:92.91.179.40:5062 --->
SIP/2.0 183 Session Progress
Call-ID: 5UrSOYtvmclD2JBK3SFYEadnlj6WSRZZ
Via: SIP/2.0/UDP 93.4.216.9:5060;received=93.4.216.9;branch=z9hG4bKPjrbz92zKKeWKGLuj2-X4WF0vPlpecPpo2;rport=5060
To: <sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=62ba2c57-63cc4af5183a903-gm-po-lucentPCSF-012866
From: <sip:+33287654321@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=hCHqGKT2rc0KomF9-.lBvKeaT6oTKtWR
CSeq: 18726 INVITE
Require: 100rel
Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,PRACK,INFO,REFER,UPDATE
Contact: <sip:lucentNGFS-118977@pcgw-0006.imsgroup-003.cor1asbc03.ims.sfr.net:5062;x-afi=8>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel";audio
Content-Type: application/sdp
P-Early-Media: inactive
RSeq: 1
Server: Alcatel-Lucent-HPSS/3.0.3
Supported: precondition
Content-Length: 256
v=0
o=LucentPCSF 830991649 830991649 IN IP4 imsgroup-003.cor1asbc03.ims.sfr.net
s=-
c=IN IP4 92.91.224.78
t=0 0
m=audio 17964 RTP/AVP 8 101
b=AS:80
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=maxptime:20
> 0x7f9f46497400 -- Strict RTP learning after remote address set to: 92.91.224.78:17964
-- PJSIP/trunk-sfr-0000004b is making progress passing it to PJSIP/1002-0000004a
> 0x7f9f464b93b0 -- Strict RTP learning after remote address set to: 192.168.3.173:7078
<--- Transmitting SIP response (753 bytes) to UDP:192.168.3.173:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.3.173:5060;rport=5060;received=192.168.3.173;branch=z9hG4bK.uUBBaJGQ1
Call-ID: w~ARmS6~3G
From: <sip:1001@domain.local>;tag=tGlzd585N
To: <sip:0612345678@domain.local>;tag=SgToqSznO0Eeq00C5ISb861.oeq7skfU
CSeq: 21 INVITE
Server: Asterisk PBX 18.15.0
Contact: <sip:192.168.3.254:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Type: application/sdp
Content-Length: 226
v=0
o=- 3348 602 IN IP4 192.168.3.254
s=Asterisk
c=IN IP4 192.168.3.254
t=0 0
m=audio 10034 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
> 0x7f9f464b93b0 -- Strict RTP switching to RTP target address 192.168.3.173:7078 as source
<--- Received SIP response (1104 bytes) from UDP:92.91.179.40:5062 --->
SIP/2.0 183 Session Progress
Call-ID: 5UrSOYtvmclD2JBK3SFYEadnlj6WSRZZ
Via: SIP/2.0/UDP 93.4.216.9:5060;received=93.4.216.9;branch=z9hG4bKPjrbz92zKKeWKGLuj2-X4WF0vPlpecPpo2;rport=5060
To: <sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=62ba2c57-63cc4af5183a903-gm-po-lucentPCSF-012866
From: <sip:+33287654321@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=hCHqGKT2rc0KomF9-.lBvKeaT6oTKtWR
CSeq: 18726 INVITE
Require: 100rel
Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,PRACK,INFO,REFER,UPDATE
Contact: <sip:lucentNGFS-118977@pcgw-0006.imsgroup-003.cor1asbc03.ims.sfr.net:5062;x-afi=8>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel";audio
Content-Type: application/sdp
P-Early-Media: inactive
RSeq: 1
Server: Alcatel-Lucent-HPSS/3.0.3
Supported: precondition
Content-Length: 256
v=0
o=LucentPCSF 830991649 830991649 IN IP4 imsgroup-003.cor1asbc03.ims.sfr.net
s=-
c=IN IP4 92.91.224.78
t=0 0
m=audio 17964 RTP/AVP 8 101
b=AS:80
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=maxptime:20
-- PJSIP/trunk-sfr-0000004b is making progress passing it to PJSIP/1002-0000004a
<--- Transmitting SIP response (753 bytes) to UDP:192.168.3.173:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.3.173:5060;rport=5060;received=192.168.3.173;branch=z9hG4bK.uUBBaJGQ1
Call-ID: w~ARmS6~3G
From: <sip:1001@domain.local>;tag=tGlzd585N
To: <sip:0612345678@domain.local>;tag=SgToqSznO0Eeq00C5ISb861.oeq7skfU
CSeq: 21 INVITE
Server: Asterisk PBX 18.15.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.3.254:5060>
Content-Type: application/sdp
Content-Length: 226
v=0
o=- 3348 602 IN IP4 192.168.3.254
s=Asterisk
c=IN IP4 192.168.3.254
t=0 0
m=audio 10034 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<--- Received SIP response (1104 bytes) from UDP:92.91.179.40:5062 --->
SIP/2.0 183 Session Progress
Call-ID: 5UrSOYtvmclD2JBK3SFYEadnlj6WSRZZ
Via: SIP/2.0/UDP 93.4.216.9:5060;received=93.4.216.9;branch=z9hG4bKPjrbz92zKKeWKGLuj2-X4WF0vPlpecPpo2;rport=5060
To: <sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=62ba2c57-63cc4af5183a903-gm-po-lucentPCSF-012866
From: <sip:+33287654321@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=hCHqGKT2rc0KomF9-.lBvKeaT6oTKtWR
CSeq: 18726 INVITE
Require: 100rel
Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,PRACK,INFO,REFER,UPDATE
Contact: <sip:lucentNGFS-118977@pcgw-0006.imsgroup-003.cor1asbc03.ims.sfr.net:5062;x-afi=8>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel";audio
Content-Type: application/sdp
P-Early-Media: inactive
RSeq: 1
Server: Alcatel-Lucent-HPSS/3.0.3
Supported: precondition
Content-Length: 256
v=0
o=LucentPCSF 830991649 830991649 IN IP4 imsgroup-003.cor1asbc03.ims.sfr.net
s=-
c=IN IP4 92.91.224.78
t=0 0
m=audio 17964 RTP/AVP 8 101
b=AS:80
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=maxptime:20
-- PJSIP/trunk-sfr-0000004b is making progress passing it to PJSIP/1002-0000004a
<--- Transmitting SIP response (753 bytes) to UDP:192.168.3.173:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.3.173:5060;rport=5060;received=192.168.3.173;branch=z9hG4bK.uUBBaJGQ1
Call-ID: w~ARmS6~3G
From: <sip:1001@domain.local>;tag=tGlzd585N
To: <sip:0612345678@domain.local>;tag=SgToqSznO0Eeq00C5ISb861.oeq7skfU
CSeq: 21 INVITE
Server: Asterisk PBX 18.15.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.3.254:5060>
Content-Type: application/sdp
Content-Length: 226
v=0
o=- 3348 602 IN IP4 192.168.3.254
s=Asterisk
c=IN IP4 192.168.3.254
t=0 0
m=audio 10034 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<--- Received SIP response (1104 bytes) from UDP:92.91.179.40:5062 --->
SIP/2.0 183 Session Progress
Call-ID: 5UrSOYtvmclD2JBK3SFYEadnlj6WSRZZ
Via: SIP/2.0/UDP 93.4.216.9:5060;received=93.4.216.9;branch=z9hG4bKPjrbz92zKKeWKGLuj2-X4WF0vPlpecPpo2;rport=5060
To: <sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=62ba2c57-63cc4af5183a903-gm-po-lucentPCSF-012866
From: <sip:+33287654321@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=hCHqGKT2rc0KomF9-.lBvKeaT6oTKtWR
CSeq: 18726 INVITE
Require: 100rel
Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,PRACK,INFO,REFER,UPDATE
Contact: <sip:lucentNGFS-118977@pcgw-0006.imsgroup-003.cor1asbc03.ims.sfr.net:5062;x-afi=8>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel";audio
Content-Type: application/sdp
P-Early-Media: inactive
RSeq: 1
Server: Alcatel-Lucent-HPSS/3.0.3
Supported: precondition
Content-Length: 256
v=0
o=LucentPCSF 830991649 830991649 IN IP4 imsgroup-003.cor1asbc03.ims.sfr.net
s=-
c=IN IP4 92.91.224.78
t=0 0
m=audio 17964 RTP/AVP 8 101
b=AS:80
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=maxptime:20
-- PJSIP/trunk-sfr-0000004b is making progress passing it to PJSIP/1002-0000004a
<--- Transmitting SIP response (753 bytes) to UDP:192.168.3.173:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.3.173:5060;rport=5060;received=192.168.3.173;branch=z9hG4bK.uUBBaJGQ1
Call-ID: w~ARmS6~3G
From: <sip:1001@domain.local>;tag=tGlzd585N
To: <sip:0612345678@domain.local>;tag=SgToqSznO0Eeq00C5ISb861.oeq7skfU
CSeq: 21 INVITE
Server: Asterisk PBX 18.15.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.3.254:5060>
Content-Type: application/sdp
Content-Length: 226
v=0
o=- 3348 602 IN IP4 192.168.3.254
s=Asterisk
c=IN IP4 192.168.3.254
t=0 0
m=audio 10034 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
> 0x7f9f464b93b0 -- Strict RTP learning complete - Locking on source address 192.168.3.173:7078
<--- Received SIP response (1104 bytes) from UDP:92.91.179.40:5062 --->
SIP/2.0 183 Session Progress
Call-ID: 5UrSOYtvmclD2JBK3SFYEadnlj6WSRZZ
Via: SIP/2.0/UDP 93.4.216.9:5060;received=93.4.216.9;branch=z9hG4bKPjrbz92zKKeWKGLuj2-X4WF0vPlpecPpo2;rport=5060
To: <sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=62ba2c57-63cc4af5183a903-gm-po-lucentPCSF-012866
From: <sip:+33287654321@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=hCHqGKT2rc0KomF9-.lBvKeaT6oTKtWR
CSeq: 18726 INVITE
Require: 100rel
Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,PRACK,INFO,REFER,UPDATE
Contact: <sip:lucentNGFS-118977@pcgw-0006.imsgroup-003.cor1asbc03.ims.sfr.net:5062;x-afi=8>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel";audio
Content-Type: application/sdp
P-Early-Media: inactive
RSeq: 1
Server: Alcatel-Lucent-HPSS/3.0.3
Supported: precondition
Content-Length: 256
v=0
o=LucentPCSF 830991649 830991649 IN IP4 imsgroup-003.cor1asbc03.ims.sfr.net
s=-
c=IN IP4 92.91.224.78
t=0 0
m=audio 17964 RTP/AVP 8 101
b=AS:80
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=maxptime:20
-- PJSIP/trunk-sfr-0000004b is making progress passing it to PJSIP/1002-0000004a
<--- Transmitting SIP response (753 bytes) to UDP:192.168.3.173:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.3.173:5060;rport=5060;received=192.168.3.173;branch=z9hG4bK.uUBBaJGQ1
Call-ID: w~ARmS6~3G
From: <sip:1001@domain.local>;tag=tGlzd585N
To: <sip:0612345678@domain.local>;tag=SgToqSznO0Eeq00C5ISb861.oeq7skfU
CSeq: 21 INVITE
Server: Asterisk PBX 18.15.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.3.254:5060>
Content-Type: application/sdp
Content-Length: 226
v=0
o=- 3348 602 IN IP4 192.168.3.254
s=Asterisk
c=IN IP4 192.168.3.254
t=0 0
m=audio 10034 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<--- Received SIP response (1104 bytes) from UDP:92.91.179.40:5062 --->
SIP/2.0 183 Session Progress
Call-ID: 5UrSOYtvmclD2JBK3SFYEadnlj6WSRZZ
Via: SIP/2.0/UDP 93.4.216.9:5060;received=93.4.216.9;branch=z9hG4bKPjrbz92zKKeWKGLuj2-X4WF0vPlpecPpo2;rport=5060
To: <sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=62ba2c57-63cc4af5183a903-gm-po-lucentPCSF-012866
From: <sip:+33287654321@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=hCHqGKT2rc0KomF9-.lBvKeaT6oTKtWR
CSeq: 18726 INVITE
Require: 100rel
Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,PRACK,INFO,REFER,UPDATE
Contact: <sip:lucentNGFS-118977@pcgw-0006.imsgroup-003.cor1asbc03.ims.sfr.net:5062;x-afi=8>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel";audio
Content-Type: application/sdp
P-Early-Media: inactive
RSeq: 1
Server: Alcatel-Lucent-HPSS/3.0.3
Supported: precondition
Content-Length: 256
v=0
o=LucentPCSF 830991649 830991649 IN IP4 imsgroup-003.cor1asbc03.ims.sfr.net
s=-
c=IN IP4 92.91.224.78
t=0 0
m=audio 17964 RTP/AVP 8 101
b=AS:80
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=maxptime:20
-- PJSIP/trunk-sfr-0000004b is making progress passing it to PJSIP/1002-0000004a
<--- Transmitting SIP response (753 bytes) to UDP:192.168.3.173:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.3.173:5060;rport=5060;received=192.168.3.173;branch=z9hG4bK.uUBBaJGQ1
Call-ID: w~ARmS6~3G
From: <sip:1001@domain.local>;tag=tGlzd585N
To: <sip:0612345678@domain.local>;tag=SgToqSznO0Eeq00C5ISb861.oeq7skfU
CSeq: 21 INVITE
Server: Asterisk PBX 18.15.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.3.254:5060>
Content-Type: application/sdp
Content-Length: 226
v=0
o=- 3348 602 IN IP4 192.168.3.254
s=Asterisk
c=IN IP4 192.168.3.254
t=0 0
m=audio 10034 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<--- Received SIP response (496 bytes) from UDP:92.91.179.40:5062 --->
SIP/2.0 500 Server Internal Error
Call-ID: 5UrSOYtvmclD2JBK3SFYEadnlj6WSRZZ
Via: SIP/2.0/UDP 93.4.216.9:5060;received=93.4.216.9;branch=z9hG4bKPjrbz92zKKeWKGLuj2-X4WF0vPlpecPpo2;rport=5060
To: <sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=62ba2c57-63cc4af5183a903-gm-po-lucentPCSF-012866
From: <sip:+33287654321@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=hCHqGKT2rc0KomF9-.lBvKeaT6oTKtWR
CSeq: 18726 INVITE
Server: Alcatel-Lucent-HPSS/3.0.3
Content-Length: 0
<--- Transmitting SIP request (535 bytes) to UDP:92.91.179.40:5062 --->
ACK sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org SIP/2.0
Via: SIP/2.0/UDP 93.4.216.9:5060;rport;branch=z9hG4bKPjrbz92zKKeWKGLuj2-X4WF0vPlpecPpo2
From: <sip:+33287654321@ims.mnc010.mcc208.3gppnetwork.org>;tag=hCHqGKT2rc0KomF9-.lBvKeaT6oTKtWR
To: <sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org>;tag=62ba2c57-63cc4af5183a903-gm-po-lucentPCSF-012866
Call-ID: 5UrSOYtvmclD2JBK3SFYEadnlj6WSRZZ
CSeq: 18726 ACK
Route: <sip:corbas.p-cscf.sfr.net:5062;lr>
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.0
Content-Length: 0
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'PJSIP/1002-0000004a' status is 'CHANUNAVAIL'
<--- Transmitting SIP response (485 bytes) to UDP:192.168.3.173:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.3.173:5060;rport=5060;received=192.168.3.173;branch=z9hG4bK.uUBBaJGQ1
Call-ID: w~ARmS6~3G
From: <sip:1001@domain.local>;tag=tGlzd585N
To: <sip:0612345678@domain.local>;tag=SgToqSznO0Eeq00C5ISb861.oeq7skfU
CSeq: 21 INVITE
Server: Asterisk PBX 18.15.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Reason: Q.850;cause=38
Content-Length: 0
<--- Received SIP request (433 bytes) from UDP:192.168.3.173:5060 --->
ACK sip:0612345678@domain.local:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.173:5060;branch=z9hG4bK.uUBBaJGQ1;rport
Call-ID: w~ARmS6~3G
From: <sip:1001@domain.local>;tag=tGlzd585N
To: <sip:0612345678@domain.local>;tag=SgToqSznO0Eeq00C5ISb861.oeq7skfU
Contact: <sip:1001@192.168.3.173;transport=udp>;expires=3599;+sip.instance="<urn:uuid:af16bccc-f148-00fe-91eb-78c65b9d3d31>"
Max-Forwards: 70
CSeq: 21 ACK