[OpenSIPS-Users] Fw: Outbound Call issue

Brian Southworth brian.southworth at clocom.uk
Wed Jun 21 06:31:26 EDT 2017 

Hello Bogdan,

 
Sorry I know it has been  while work had become busy.

 
Here is the SIP debug from what I can see opensips is sending a cancel 101 and terminating the call before its even passed to the carrier

 
[Jun 21 10:10:45] <------------->

[Jun 21 10:10:45] --- (15 headers 18 lines) ---

[Jun 21 10:10:45] Sending to 34.250.75.163:5060 (no NAT)

[Jun 21 10:10:45] Sending to 34.250.75.163:5060 (no NAT)

[Jun 21 10:10:45] Using INVITE request as basis request - a6c89c83-4d38dcf8 at 192.168.1.48

[Jun 21 10:10:45] Found peer 'opensips' for '1007' from 34.250.75.163:5060

[Jun 21 10:10:45]   == Using SIP RTP CoS mark 5

[Jun 21 10:10:45] Found RTP audio format 0

[Jun 21 10:10:45] Found RTP audio format 2

[Jun 21 10:10:45] Found RTP audio format 8

[Jun 21 10:10:45] Found RTP audio format 9

[Jun 21 10:10:45] Found RTP audio format 18

[Jun 21 10:10:45] Found RTP audio format 96

[Jun 21 10:10:45] Found RTP audio format 97

[Jun 21 10:10:45] Found RTP audio format 98

[Jun 21 10:10:45] Found RTP audio format 101

[Jun 21 10:10:45] Found audio description format PCMU for ID 0

[Jun 21 10:10:45] Found audio description format G726-32 for ID 2

[Jun 21 10:10:45] Found audio description format PCMA for ID 8

[Jun 21 10:10:45] Found audio description format G722 for ID 9

[Jun 21 10:10:45] Found audio description format G729a for ID 18

[Jun 21 10:10:45] Found unknown media description format G726-40 for ID 96

[Jun 21 10:10:45] Found unknown media description format G726-24 for ID 97

[Jun 21 10:10:45] Found unknown media description format G726-16 for ID 98

[Jun 21 10:10:45] Found audio description format telephone-event for ID 101

[Jun 21 10:10:45] Capabilities: us - (alaw), peer - audio=(ulaw|g726|alaw|g722|g729)/video=(nothing)/text=(nothing), combined - (alaw)

[Jun 21 10:10:45] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)

[Jun 21 10:10:45] Peer audio RTP is at port 46.102.201.115:16496

[Jun 21 10:10:45] Looking for 07476243394 in from-sip (domain opensips

[Jun 21 10:10:45] sip_route_dump: route/path hop: <sip:1007 at office ip:5060>

[Jun 21 10:10:45]

[Jun 21 10:10:45] <--- Transmitting (NAT) to opensips:5060 --->

[Jun 21 10:10:45] SIP/2.0 100 Trying

[Jun 21 10:10:45] Via: SIP/2.0/UDP opensips:5060;branch=z9hG4bK2185.5135a617.0;received=opensips;rport=5060

[Jun 21 10:10:45] Via: SIP/2.0/UDP office ip:5060;branch=z9hG4bK-af4637c

[Jun 21 10:10:45] From: "opensips" <sip:1007 at opensips>;tag=6cf20f07e3486c44o3

[Jun 21 10:10:45] To: "Brian07476243394" <sip:07476243394 at opensips>

[Jun 21 10:10:45] Call-ID: a6c89c83-4d38dcf8 at 192.168.1.48

[Jun 21 10:10:45] CSeq: 101 INVITE

[Jun 21 10:10:45] Server: Asterisk PBX GIT-master-b05d2fd

[Jun 21 10:10:45] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

[Jun 21 10:10:45] Supported: replaces, timer

[Jun 21 10:10:45] Contact: <sip:07476243394 at 52.19.193.46:5060>

[Jun 21 10:10:45] Content-Length: 0

[Jun 21 10:10:45]

[Jun 21 10:10:45]

[Jun 21 10:10:45] <------------>

[Jun 21 10:10:45]     -- Executing [07476243394 at from-sip:1] Set("SIP/opensips-00000027", "CALLERID(num)=01611234565") in new stack

[Jun 21 10:10:45]     -- Executing [07476243394 at from-sip:2] Ringing("SIP/opensips-00000027", "") in new stack

[Jun 21 10:10:45]

[Jun 21 10:10:45] <--- Transmitting (NAT) to opensips:5060 --->

[Jun 21 10:10:45] SIP/2.0 180 Ringing

[Jun 21 10:10:45] Via: SIP/2.0/UDP opensips:5060;branch=z9hG4bK2185.5135a617.0;received=opensips;rport=5060

[Jun 21 10:10:45] Via: SIP/2.0/UDP office ip:5060;branch=z9hG4bK-af4637c

[Jun 21 10:10:45] From: "opensips" <sip:1007 at opensips>;tag=6cf20f07e3486c44o3

[Jun 21 10:10:45] To: "Brian07476243394" <sip:07476243394 at opensips>;tag=as29b37eb3

[Jun 21 10:10:45] Call-ID: a6c89c83-4d38dcf8 at 192.168.1.48

[Jun 21 10:10:45] CSeq: 101 INVITE

[Jun 21 10:10:45] Server: Asterisk PBX GIT-master-b05d2fd

[Jun 21 10:10:45] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

[Jun 21 10:10:45] Supported: replaces, timer

[Jun 21 10:10:45] Contact: <sip:07476243394 at 52.19.193.46:5060>

[Jun 21 10:10:45] Content-Length: 0

[Jun 21 10:10:45]

[Jun 21 10:10:45]

[Jun 21 10:10:45] <------------>

[Jun 21 10:10:45]

[Jun 21 10:10:45] <--- SIP read from UDP:openips:5060 --->

[Jun 21 10:10:45] CANCEL sip:07476243394 at opensips SIP/2.0

[Jun 21 10:10:45] Via: SIP/2.0/UDP 34.250.75.163:5060;branch=z9hG4bK2185.5135a617.0

[Jun 21 10:10:45] From: "opensips" <sip:1007 at 34.250.75.163>;tag=6cf20f07e3486c44o3

[Jun 21 10:10:45] Call-ID: a6c89c83-4d38dcf8 at 192.168.1.48

[Jun 21 10:10:45] To: "Brian07476243394" <sip:07476243394 at 34.250.75.163>


[Jun 21 10:10:45] CSeq: 101 CANCEL

[Jun 21 10:10:45] Max-Forwards: 70

[Jun 21 10:10:45] User-Agent: OpenSIPS (2.2.3 (x86_64/linux))

[Jun 21 10:10:45] Content-Length: 0

[Jun 21 10:10:45]

[Jun 21 10:10:45] <------------->

[Jun 21 10:10:45] --- (9 headers 0 lines) ---

[Jun 21 10:10:45] Sending to opensips:5060 (NAT)

[Jun 21 10:10:45]

[Jun 21 10:10:45] <--- Reliably Transmitting (NAT) to opensips:5060 --->


[Jun 21 10:10:45] SIP/2.0 487 Request Terminated

[Jun 21 10:10:45] Via: SIP/2.0/UDP opensips:5060;branch=z9hG4bK2185.5135a617.0;received=34.250.75.163;rport=5060

[Jun 21 10:10:45] Via: SIP/2.0/UDP office IP:5060;branch=z9hG4bK-af4637c

[Jun 21 10:10:45] From: "opensips" <sip:1007 at opensips>;tag=6cf20f07e3486c44o3

[Jun 21 10:10:45] To: "Brian07476243394" <sip:07476243394 at opensips>;tag=as29b37eb3

[Jun 21 10:10:45] Call-ID: a6c89c83-4d38dcf8 at 192.168.1.48

[Jun 21 10:10:45] CSeq: 101 INVITE

[Jun 21 10:10:45] Server: Asterisk PBX GIT-master-b05d2fd

[Jun 21 10:10:45] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

[Jun 21 10:10:45] Supported: replaces, timer

[Jun 21 10:10:45] Content-Length: 0

[Jun 21 10:10:45]

[Jun 21 10:10:45]

[Jun 21 10:10:45] <------------>

[Jun 21 10:10:45]

[Jun 21 10:10:45] <--- Transmitting (NAT) to opensips:5060 --->

[Jun 21 10:10:45] SIP/2.0 200 OK

[Jun 21 10:10:45] Via: SIP/2.0/UDP opensips:5060;branch=z9hG4bK2185.5135a617.0;received=opensips;rport=5060

[Jun 21 10:10:45] From: "opensips" <sip:1007 at opensips>;tag=6cf20f07e3486c44o3

[Jun 21 10:10:45] To: "Brian07476243394" <sip:07476243394 at opensips>;tag=as29b37eb3

[Jun 21 10:10:45] Call-ID: a6c89c83-4d38dcf8 at 192.168.1.48

[Jun 21 10:10:45] CSeq: 101 CANCEL

[Jun 21 10:10:45] Server: Asterisk PBX GIT-master-b05d2fd

[Jun 21 10:10:45] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

[Jun 21 10:10:45] Supported: replaces, timer

[Jun 21 10:10:45] Content-Length: 0

[Jun 21 10:10:45]

[Jun 21 10:10:45]

[Jun 21 10:10:45] <------------>

[Jun 21 10:10:45]

[Jun 21 10:10:45] <--- SIP read from UDP:opensips:5060 --->

[Jun 21 10:10:45] ACK sip:07476243394 at opensips SIP/2.0

[Jun 21 10:10:45] Via: SIP/2.0/UDP opensips:5060;branch=z9hG4bK2185.5135a617.0

[Jun 21 10:10:45] From: "opensips" <sip:1007 at opensips>;tag=6cf20f07e3486c44o3

[Jun 21 10:10:45] Call-ID: a6c89c83-4d38dcf8 at 192.168.1.48

[Jun 21 10:10:45] To: "Brian07476243394" <sip:07476243394 at opensips>;tag=as29b37eb3

[Jun 21 10:10:45] CSeq: 101 ACK

[Jun 21 10:10:45] Max-Forwards: 70

[Jun 21 10:10:45] User-Agent: OpenSIPS (2.2.3 (x86_64/linux))

[Jun 21 10:10:45] Content-Length: 0

[Jun 21 10:10:45]

[Jun 21 10:10:45] <------------->

 
 
I for security reasons I have edited any IP’s if you need the unedited version I will send it in an email directly

 
Regards,

 
Brian Southworth

Communications Developer


111 Wilmslow Road

Handforth

Wilmslow

SK9 3ER

 
T: 03333 446677

DDI:01625 837112

W: www.clocom.uk <http://www.clocom.uk/> 

 
	 

	 

	 

	 

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