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- Appel d'un sip interne vers un portable externe via trunk OVH,
- sonnerie KO poste appellant, appelé sonnerie OK, conversation KO, pas de raccroché détecté.
- Le poste SIP raccroche manuellement à la fin.
- SIP Debugging enabled
- <--- SIP read from UDP:192.168.1.150:5060 --->
- INVITE sip:06XXXXXXXX@192.168.1.200 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKa67cf3240a443c02
- From: "Florent TOTOR" <sip:101@192.168.1.200>;tag=ba6dba2dbe613f58
- To: <sip:06XXXXXXXX@192.168.1.200>
- Contact: <sip:101@192.168.1.150:5060;transport=udp>
- Supported: replaces, timer, path
- X-Grandstream-PBX: true
- P-Early-Media: Supported
- Call-ID: 1f25e87efe2b6f84@192.168.1.150
- CSeq: 33404 INVITE
- User-Agent: Grandstream GXP1200 1.2.5.3
- Max-Forwards: 70
- Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
- Content-Type: application/sdp
- Content-Length: 313
- v=0
- o=101 8000 8000 IN IP4 192.168.1.150
- s=SIP Call
- c=IN IP4 192.168.1.150
- t=0 0
- m=audio 5014 RTP/AVP 9 18 8 0 2 101
- a=sendrecv
- a=rtpmap:9 G722/8000
- a=rtpmap:18 G729/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:0 PCMU/8000
- a=rtpmap:2 G726-32/8000
- a=ptime:20
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-11
- <------------->
- --- (15 headers 15 lines) ---
- Sending to 192.168.1.150:5060 (no NAT)
- Using INVITE request as basis request - 1f25e87efe2b6f84@192.168.1.150
- Found peer '101' for '101' from 192.168.1.150:5060
- <--- Reliably Transmitting (NAT) to 192.168.1.150:5060 --->
- SIP/2.0 401 Unauthorized
- Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKa67cf3240a443c02;received=192.168.1.150;rport=5060
- From: "Florent TOTOR" <sip:101@192.168.1.200>;tag=ba6dba2dbe613f58
- To: <sip:06XXXXXXXX@192.168.1.200>;tag=as4c3d30a9
- Call-ID: 1f25e87efe2b6f84@192.168.1.150
- CSeq: 33404 INVITE
- Server: AskoziaPBX
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="75a6cfa2"
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog '1f25e87efe2b6f84@192.168.1.150' in 6400 ms (Method: INVITE)
- <--- SIP read from UDP:192.168.1.150:5060 --->
- ACK sip:06XXXXXXXX@192.168.1.200 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKa67cf3240a443c02
- From: "Florent TOTOR" <sip:101@192.168.1.200>;tag=ba6dba2dbe613f58
- To: <sip:06XXXXXXXX@192.168.1.200>;tag=as4c3d30a9
- Contact: <sip:101@192.168.1.150:5060;transport=udp>
- Supported: path
- X-Grandstream-PBX: true
- Call-ID: 1f25e87efe2b6f84@192.168.1.150
- CSeq: 33404 ACK
- User-Agent: Grandstream GXP1200 1.2.5.3
- Max-Forwards: 70
- Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
- Content-Length: 0
- <------------->
- --- (13 headers 0 lines) ---
- <--- SIP read from UDP:192.168.1.150:5060 --->
- INVITE sip:06XXXXXXXX@192.168.1.200 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKb82faf11bd292194
- From: "Florent TOTOR" <sip:101@192.168.1.200>;tag=ba6dba2dbe613f58
- To: <sip:06XXXXXXXX@192.168.1.200>
- Contact: <sip:101@192.168.1.150:5060;transport=udp>
- Supported: replaces, timer, path
- X-Grandstream-PBX: true
- P-Early-Media: Supported
- Authorization: Digest username="101", realm="asterisk", algorithm=MD5, uri="sip:06XXXXXXXX@192.168.1.200", nonce="75a6cfa2", response="56ea22283ad59cd23789e02653c31800"
- Call-ID: 1f25e87efe2b6f84@192.168.1.150
- CSeq: 33405 INVITE
- User-Agent: Grandstream GXP1200 1.2.5.3
- Max-Forwards: 70
- Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
- Content-Type: application/sdp
- Content-Length: 313
- v=0
- o=101 8000 8001 IN IP4 192.168.1.150
- s=SIP Call
- c=IN IP4 192.168.1.150
- t=0 0
- m=audio 5014 RTP/AVP 9 18 8 0 2 101
- a=sendrecv
- a=rtpmap:9 G722/8000
- a=rtpmap:18 G729/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:0 PCMU/8000
- a=rtpmap:2 G726-32/8000
- a=ptime:20
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-11
- <------------->
- --- (16 headers 15 lines) ---
- Sending to 192.168.1.150:5060 (NAT)
- Using INVITE request as basis request - 1f25e87efe2b6f84@192.168.1.150
- Found peer '101' for '101' from 192.168.1.150:5060
- Found RTP audio format 9
- Found RTP audio format 18
- Found RTP audio format 8
- Found RTP audio format 0
- Found RTP audio format 2
- Found RTP audio format 101
- Found audio description format G722 for ID 9
- Found audio description format G729 for ID 18
- Found audio description format PCMA for ID 8
- Found audio description format PCMU for ID 0
- Found audio description format G726-32 for ID 2
- Found audio description format telephone-event for ID 101
- Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x190c (ulaw|alaw|g726|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
- Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
- Peer audio RTP is at port 192.168.1.150:5014
- Looking for 06XXXXXXXX in SIP-PHONE-5171585574ee5e5a8ba8f6 (domain 192.168.1.200)
- list_route: hop: <sip:101@192.168.1.150:5060;transport=udp>
- <--- Transmitting (NAT) to 192.168.1.150:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKb82faf11bd292194;received=192.168.1.150;rport=5060
- From: "Florent TOTOR" <sip:101@192.168.1.200>;tag=ba6dba2dbe613f58
- To: <sip:06XXXXXXXX@192.168.1.200>
- Call-ID: 1f25e87efe2b6f84@192.168.1.150
- CSeq: 33405 INVITE
- Server: AskoziaPBX
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Contact: <sip:06XXXXXXXX@192.168.1.200:5060>
- Content-Length: 0
- <------------>
- Audio is at 5060
- Adding codec 0x4 (ulaw) to SDP
- Adding codec 0x8 (alaw) to SDP
- Adding codec 0x2 (gsm) to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- Reliably Transmitting (NAT) to 91.121.129.17:5060:
- INVITE sip:06XXXXXXXX@sip.ovh.net SIP/2.0
- Via: SIP/2.0/UDP 10.192.26.204:5060;branch=z9hG4bK3e7cd865;rport
- Max-Forwards: 70
- From: "Default Extension" <sip:0033183643585@sip.ovh.net>;tag=as4523e5ab
- To: <sip:06XXXXXXXX@sip.ovh.net>
- Contact: <sip:0033183643585@10.192.26.204:5060>
- Call-ID: 75f7814960e8c0081359b3700cc6f332@sip.ovh.net
- CSeq: 102 INVITE
- User-Agent: AskoziaPBX
- Date: Mon, 12 Dec 2011 11:48:06 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Type: application/sdp
- Content-Length: 285
- v=0
- o=root 1595019147 1595019147 IN IP4 10.192.26.204
- s=Asterisk PBX 1.8.4.4
- c=IN IP4 10.192.26.204
- t=0 0
- m=audio 10072 RTP/AVP 0 8 3 101
- a=rtpmap:0 PCMU/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=sendrecv
- ---
- Retransmitting #1 (NAT) to 91.121.129.17:5060:
- INVITE sip:06XXXXXXXX@sip.ovh.net SIP/2.0
- Via: SIP/2.0/UDP 10.192.26.204:5060;branch=z9hG4bK3e7cd865;rport
- Max-Forwards: 70
- From: "Default Extension" <sip:0033183643585@sip.ovh.net>;tag=as4523e5ab
- To: <sip:06XXXXXXXX@sip.ovh.net>
- Contact: <sip:0033183643585@10.192.26.204:5060>
- Call-ID: 75f7814960e8c0081359b3700cc6f332@sip.ovh.net
- CSeq: 102 INVITE
- User-Agent: AskoziaPBX
- Date: Mon, 12 Dec 2011 11:48:06 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Type: application/sdp
- Content-Length: 285
- v=0
- o=root 1595019147 1595019147 IN IP4 10.192.26.204
- s=Asterisk PBX 1.8.4.4
- c=IN IP4 10.192.26.204
- t=0 0
- m=audio 10072 RTP/AVP 0 8 3 101
- a=rtpmap:0 PCMU/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=sendrecv
- ---
- <--- SIP read from UDP:91.121.129.17:5060 --->
- SIP/2.0 407 authentication required
- Allow: UPDATE,REFER,INFO
- Call-ID: 75f7814960e8c0081359b3700cc6f332@sip.ovh.net
- Contact: <sip:06XXXXXXXX@91.121.129.17:5060;user=phone>
- CSeq: 102 INVITE
- From: "Default Extension" <sip:0033183643585@sip.ovh.net>;tag=as4523e5ab
- Proxy-Authenticate: Digest realm="sip.ovh.net",nonce="0973b5c766fddd071e1927cd6756ac7d",opaque="0972c2e8159490c",stale=false,algorithm=MD5
- Server: Cirpack/v4.42j (gw_sip)
- To: <sip:06XXXXXXXX@sip.ovh.net>;tag=00-07862-0973bb13-60144c9b0
- Via: SIP/2.0/UDP 10.192.26.204:5060;received=95.210.161.54;rport=5060;branch=z9hG4bK3e7cd865
- Content-Length: 0
- <------------->
- --- (11 headers 0 lines) ---
- Transmitting (NAT) to 91.121.129.17:5060:
- ACK sip:06XXXXXXXX@sip.ovh.net SIP/2.0
- Via: SIP/2.0/UDP 10.192.26.204:5060;branch=z9hG4bK3e7cd865;rport
- Max-Forwards: 70
- From: "Default Extension" <sip:0033183643585@sip.ovh.net>;tag=as4523e5ab
- To: <sip:06XXXXXXXX@sip.ovh.net>;tag=00-07862-0973bb13-60144c9b0
- Contact: <sip:0033183643585@10.192.26.204:5060>
- Call-ID: 75f7814960e8c0081359b3700cc6f332@sip.ovh.net
- CSeq: 102 ACK
- User-Agent: AskoziaPBX
- Content-Length: 0
- ---
- Audio is at 5060
- Adding codec 0x4 (ulaw) to SDP
- Adding codec 0x8 (alaw) to SDP
- Adding codec 0x2 (gsm) to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- Reliably Transmitting (NAT) to 91.121.129.17:5060:
- INVITE sip:06XXXXXXXX@sip.ovh.net SIP/2.0
- Via: SIP/2.0/UDP 10.192.26.204:5060;branch=z9hG4bK0b23a06a;rport
- Max-Forwards: 70
- From: "Default Extension" <sip:0033183643585@sip.ovh.net>;tag=as4523e5ab
- To: <sip:06XXXXXXXX@sip.ovh.net>
- Contact: <sip:0033183643585@10.192.26.204:5060>
- Call-ID: 75f7814960e8c0081359b3700cc6f332@sip.ovh.net
- CSeq: 103 INVITE
- User-Agent: AskoziaPBX
- Proxy-Authorization: Digest username="0033183643585", realm="sip.ovh.net", algorithm=MD5, uri="sip:06XXXXXXXX@sip.ovh.net", nonce="0973b5c766fddd071e1927cd6756ac7d", response="8d1362978804f22773300724ae9822ef", opaque="0972c2e8159490c"
- Date: Mon, 12 Dec 2011 11:48:07 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Type: application/sdp
- Content-Length: 285
- v=0
- o=root 1595019147 1595019148 IN IP4 10.192.26.204
- s=Asterisk PBX 1.8.4.4
- c=IN IP4 10.192.26.204
- t=0 0
- m=audio 10072 RTP/AVP 0 8 3 101
- a=rtpmap:0 PCMU/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=sendrecv
- ---
- <--- SIP read from UDP:91.121.129.17:5060 --->
- SIP/2.0 407 authentication required
- Allow: UPDATE,REFER,INFO
- Call-ID: 75f7814960e8c0081359b3700cc6f332@sip.ovh.net
- Contact: <sip:06XXXXXXXX@91.121.129.17:5060;user=phone>
- CSeq: 102 INVITE
- From: "Default Extension" <sip:0033183643585@sip.ovh.net>;tag=as4523e5ab
- Proxy-Authenticate: Digest realm="sip.ovh.net",nonce="0973b5c766fddd071e1927cd6756ac7d",opaque="0972c2e8159490c",stale=false,algorithm=MD5
- Server: Cirpack/v4.42j (gw_sip)
- To: <sip:06XXXXXXXX@sip.ovh.net>;tag=00-07862-0973bb13-60144c9b0
- Via: SIP/2.0/UDP 10.192.26.204:5060;received=95.210.161.54;rport=5060;branch=z9hG4bK3e7cd865
- Content-Length: 0
- <------------->
- --- (11 headers 0 lines) ---
- Transmitting (NAT) to 91.121.129.17:5060:
- ACK sip:06XXXXXXXX@sip.ovh.net SIP/2.0
- Via: SIP/2.0/UDP 10.192.26.204:5060;branch=z9hG4bK0b23a06a;rport
- Max-Forwards: 70
- From: "Default Extension" <sip:0033183643585@sip.ovh.net>;tag=as4523e5ab
- To: <sip:06XXXXXXXX@sip.ovh.net>
- Contact: <sip:0033183643585@10.192.26.204:5060>
- Call-ID: 75f7814960e8c0081359b3700cc6f332@sip.ovh.net
- CSeq: 102 ACK
- User-Agent: AskoziaPBX
- Content-Length: 0
- ---
- Retransmitting #1 (NAT) to 91.121.129.17:5060:
- INVITE sip:06XXXXXXXX@sip.ovh.net SIP/2.0
- Via: SIP/2.0/UDP 10.192.26.204:5060;branch=z9hG4bK0b23a06a;rport
- Max-Forwards: 70
- From: "Default Extension" <sip:0033183643585@sip.ovh.net>;tag=as4523e5ab
- To: <sip:06XXXXXXXX@sip.ovh.net>
- Contact: <sip:0033183643585@10.192.26.204:5060>
- Call-ID: 75f7814960e8c0081359b3700cc6f332@sip.ovh.net
- CSeq: 103 INVITE
- User-Agent: AskoziaPBX
- Proxy-Authorization: Digest username="0033183643585", realm="sip.ovh.net", algorithm=MD5, uri="sip:06XXXXXXXX@sip.ovh.net", nonce="0973b5c766fddd071e1927cd6756ac7d", response="8d1362978804f22773300724ae9822ef", opaque="0972c2e8159490c"
- Date: Mon, 12 Dec 2011 11:48:07 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Type: application/sdp
- Content-Length: 285
- v=0
- o=root 1595019147 1595019148 IN IP4 10.192.26.204
- s=Asterisk PBX 1.8.4.4
- c=IN IP4 10.192.26.204
- t=0 0
- m=audio 10072 RTP/AVP 0 8 3 101
- a=rtpmap:0 PCMU/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=sendrecv
- ---
- <--- SIP read from UDP:91.121.129.17:5060 --->
- SIP/2.0 100 Trying
- Allow: UPDATE,REFER,INFO
- Call-ID: 75f7814960e8c0081359b3700cc6f332@sip.ovh.net
- Contact: <sip:91.121.129.17:5060>
- CSeq: 103 INVITE
- From: "Default Extension" <sip:0033183643585@sip.ovh.net>;tag=as4523e5ab
- Server: Cirpack/v4.42j (gw_sip)
- To: <sip:06XXXXXXXX@sip.ovh.net>
- Via: SIP/2.0/UDP 10.192.26.204:5060;received=95.210.161.54;rport=5060;branch=z9hG4bK0b23a06a
- Content-Length: 0
- <------------->
- --- (10 headers 0 lines) ---
- <--- SIP read from UDP:192.168.1.150:5060 --->
- CANCEL sip:06XXXXXXXX@192.168.1.200 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKb82faf11bd292194
- From: "Florent TOTOR" <sip:101@192.168.1.200>;tag=ba6dba2dbe613f58
- To: <sip:06XXXXXXXX@192.168.1.200>
- Supported: path
- X-Grandstream-PBX: true
- Call-ID: 1f25e87efe2b6f84@192.168.1.150
- CSeq: 33405 CANCEL
- User-Agent: Grandstream GXP1200 1.2.5.3
- Max-Forwards: 70
- Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
- Content-Length: 0
- <------------->
- --- (12 headers 0 lines) ---
- Sending to 192.168.1.150:5060 (NAT)
- <--- Reliably Transmitting (NAT) to 192.168.1.150:5060 --->
- SIP/2.0 487 Request Terminated
- Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKb82faf11bd292194;received=192.168.1.150;rport=5060
- From: "Florent TOTOR" <sip:101@192.168.1.200>;tag=ba6dba2dbe613f58
- To: <sip:06XXXXXXXX@192.168.1.200>;tag=as0683db8c
- Call-ID: 1f25e87efe2b6f84@192.168.1.150
- CSeq: 33405 INVITE
- Server: AskoziaPBX
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Length: 0
- <------------>
- <--- Transmitting (NAT) to 192.168.1.150:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKb82faf11bd292194;received=192.168.1.150;rport=5060
- From: "Florent TOTOR" <sip:101@192.168.1.200>;tag=ba6dba2dbe613f58
- To: <sip:06XXXXXXXX@192.168.1.200>;tag=as0683db8c
- Call-ID: 1f25e87efe2b6f84@192.168.1.150
- CSeq: 33405 CANCEL
- Server: AskoziaPBX
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Length: 0
- <------------>
- Scheduling destruction of SIP dialog '75f7814960e8c0081359b3700cc6f332@sip.ovh.net' in 53440 ms (Method: INVITE)
- Retransmitting #1 (NAT) to 192.168.1.150:5060:
- SIP/2.0 487 Request Terminated
- Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKb82faf11bd292194;received=192.168.1.150;rport=5060
- From: "Florent TOTOR" <sip:101@192.168.1.200>;tag=ba6dba2dbe613f58
- To: <sip:06XXXXXXXX@192.168.1.200>;tag=as0683db8c
- Call-ID: 1f25e87efe2b6f84@192.168.1.150
- CSeq: 33405 INVITE
- Server: AskoziaPBX
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Length: 0
- ---
- <--- SIP read from UDP:192.168.1.150:5060 --->
- ACK sip:06XXXXXXXX@192.168.1.200 SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKb82faf11bd292194
- From: "Florent TOTOR" <sip:101@192.168.1.200>;tag=ba6dba2dbe613f58
- To: <sip:06XXXXXXXX@192.168.1.200>;tag=as0683db8c
- Contact: <sip:101@192.168.1.150:5060;transport=udp>
- Supported: path
- X-Grandstream-PBX: true
- Authorization: Digest username="101", realm="asterisk", algorithm=MD5, uri="sip:06XXXXXXXX@192.168.1.200", nonce="75a6cfa2", response="56ea22283ad59cd23789e02653c31800"
- Call-ID: 1f25e87efe2b6f84@192.168.1.150
- CSeq: 33405 ACK
- User-Agent: Grandstream GXP1200 1.2.5.3
- Max-Forwards: 70
- Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
- Content-Length: 0
- <------------->
- --- (14 headers 0 lines) ---
- Really destroying SIP dialog '1f25e87efe2b6f84@192.168.1.150' Method: ACK
- Really destroying SIP dialog '2a25b55a29a64948559af39e1330ea8e@192.168.1.2' Method: REGISTER
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