Outgoing call cut at the same moment that redirect

Hi.

I configure an extension of another SIP server as TRUNK in my asterisk server.

The problem is that i can receive incoming calls but i can’t make outgoing calls.

I try the same configuration of the user, password, proxy and hostname in one voip phone, and works ok.

I can see that at the same moment that the call is redirected, this one is closed by asterisk…

The sip server is the ip 192.168.1.1 and my asterisk server is 192.168.100.21

Here the copy of the log:

[Nov 3 17:45:52] VERBOSE[14741][C-00000161] chan_sip.c: Sending to 192.168.100.33:5060 (no NAT)
[Nov 3 17:45:52] VERBOSE[14741][C-00000161] chan_sip.c: Using INVITE request as basis request - 1764563286-5060-26@BJC.BGI.BAA.DD
[Nov 3 17:45:52] VERBOSE[14741][C-00000161] chan_sip.c: Found peer ‘1002’ for ‘1002’ from 192.168.100.33:5060
[Nov 3 17:45:52] VERBOSE[14741][C-00000161] chan_sip.c: Found RTP audio format 0
[Nov 3 17:45:52] VERBOSE[14741][C-00000161] chan_sip.c: Found RTP audio format 8
[Nov 3 17:45:52] VERBOSE[14741][C-00000161] chan_sip.c: Found RTP audio format 4
[Nov 3 17:45:52] VERBOSE[14741][C-00000161] chan_sip.c: Found RTP audio format 18
[Nov 3 17:45:52] VERBOSE[14741][C-00000161] chan_sip.c: Found RTP audio format 2
[Nov 3 17:45:52] VERBOSE[14741][C-00000161] chan_sip.c: Found RTP audio format 97
[Nov 3 17:45:52] VERBOSE[14741][C-00000161] chan_sip.c: Found RTP audio format 101
[Nov 3 17:45:52] VERBOSE[14741][C-00000161] chan_sip.c: Found audio description format PCMU for ID 0
[Nov 3 17:45:52] VERBOSE[14741][C-00000161] chan_sip.c: Found audio description format PCMA for ID 8
[Nov 3 17:45:52] VERBOSE[14741][C-00000161] chan_sip.c: Found audio description format G723 for ID 4
[Nov 3 17:45:52] VERBOSE[14741][C-00000161] chan_sip.c: Found audio description format G729 for ID 18
[Nov 3 17:45:52] VERBOSE[14741][C-00000161] chan_sip.c: Found audio description format G726-32 for ID 2
[Nov 3 17:45:52] VERBOSE[14741][C-00000161] chan_sip.c: Found audio description format iLBC for ID 97
[Nov 3 17:45:52] VERBOSE[14741][C-00000161] chan_sip.c: Found audio description format telephone-event for ID 101
[Nov 3 17:45:52] VERBOSE[14741][C-00000161] chan_sip.c: Capabilities: us - (ulaw|gsm), peer - audio=(ulaw|g726|g723|alaw|g729|ilbc)/video=(nothing)/text=(nothing), combined - (ulaw)
[Nov 3 17:45:52] VERBOSE[14741][C-00000161] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Nov 3 17:45:52] VERBOSE[14741][C-00000161] chan_sip.c: Peer audio RTP is at port 192.168.100.33:5004
[Nov 3 17:45:52] VERBOSE[14741][C-00000161] chan_sip.c: Looking for XXXXXXXXX in DLPN_DialPlan1 (domain 192.168.100.21)
[Nov 3 17:45:52] VERBOSE[14741][C-00000161] sip/route.c: sip_route_dump: route/path hop: sip:1002@192.168.100.33:5060
[Nov 3 17:45:52] VERBOSE[14741][C-00000161] chan_sip.c:
[Nov 3 17:45:52] VERBOSE[19201][C-00000161] chan_sip.c: Audio is at 14366
[Nov 3 17:45:52] VERBOSE[19201][C-00000161] chan_sip.c: Adding codec ulaw to SDP
[Nov 3 17:45:52] VERBOSE[19201][C-00000161] chan_sip.c: Adding codec alaw to SDP
[Nov 3 17:45:52] VERBOSE[19201][C-00000161] chan_sip.c: Adding codec gsm to SDP
[Nov 3 17:45:52] VERBOSE[19201][C-00000161] chan_sip.c: Adding codec g726 to SDP
[Nov 3 17:45:52] VERBOSE[19201][C-00000161] chan_sip.c: Adding codec g729 to SDP
[Nov 3 17:45:52] VERBOSE[19201][C-00000161] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Nov 3 17:45:52] VERBOSE[19201][C-00000161] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.1.1:5060:
[Nov 3 17:45:52] VERBOSE[14741] chan_sip.c:
[Nov 3 17:45:52] VERBOSE[14741] chan_sip.c: — (8 headers 0 lines) —
[Nov 3 17:45:52] VERBOSE[14741][C-00000161] chan_sip.c: Transmitting (no NAT) to 192.168.1.1:5060:
[Nov 3 17:45:52] VERBOSE[19201][C-00000161] chan_sip.c: Scheduling destruction of SIP dialog ‘4d36e7e6115403a55474571452cf632c@192.168.100.21:5060’ in 32000 ms (Method: INVITE)
[Nov 3 17:45:52] VERBOSE[19201][C-00000161] chan_sip.c: Scheduling destruction of SIP dialog '1764563286-5060-26@BJC.BGI.BAA.DD’ in 32000 ms (Method: INVITE)
[Nov 3 17:45:52] VERBOSE[19201][C-00000161] chan_sip.c:

I try with nat and not nat configuration but not works.

The log doesn’t show any call being closed, and is difficult to use without the sip set debug on info.

Hi.

Well, i check the debug option an i can view this information:

[Nov 4 16:13:16] VERBOSE[14741] chan_sip.c:
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 0 [ 43]: INVITE sip:XXXXXXXXX@192.168.100.21 SIP/2.0
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 192.168.100.32:5060;branch=z9hG4bK1815663290;rport
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 2 [ 54]: From: “Sergio” sip:1001@192.168.100.21;tag=259365463
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 3 [ 34]: To: sip:XXXXXXXXX@192.168.100.21
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 4 [ 42]: Call-ID: 1695553247-5060-64@BJC.BGI.BAA.DC
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 5 [ 16]: CSeq: 620 INVITE
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 6 [ 48]: Contact: “Sergio” sip:1001@192.168.100.32:5060
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 8 [ 40]: User-Agent: Grandstream GXP1625 1.0.1.12
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 9 [ 13]: Privacy: none
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 10 [ 56]: P-Preferred-Identity: “Sergio” sip:1001@192.168.100.21
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 11 [ 32]: Supported: replaces, path, timer
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 12 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 14 [ 47]: Accept: application/sdp, application/dtmf-relay
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 15 [ 21]: Content-Length: 337
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 16 [ 0]:
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 0 [ 3]: v=0
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 1 [ 38]: o=1001 8000 8000 IN IP4 192.168.100.32
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 2 [ 10]: s=SIP Call
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.100.32
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 4 [ 5]: t=0 0
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 5 [ 35]: m=audio 5004 RTP/AVP 0 8 18 9 2 101
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 6 [ 10]: a=sendrecv
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 8 [ 10]: a=ptime:20
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 9 [ 20]: a=rtpmap:8 PCMA/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 10 [ 21]: a=rtpmap:18 G729/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 11 [ 19]: a=fmtp:18 annexb=no
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 12 [ 20]: a=rtpmap:9 G722/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 13 [ 23]: a=rtpmap:2 G726-32/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 14 [ 33]: a=rtpmap:101 telephone-event/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 15 [ 15]: a=fmtp:101 0-15
[Nov 4 16:13:16] VERBOSE[14741] chan_sip.c: — (16 headers 16 lines) —
[Nov 4 16:13:16] DEBUG[14741] acl.c: For destination ‘192.168.100.32’, our source address is ‘192.168.100.21’.
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Setting AST_TRANSPORT_UDP with address 192.168.100.21:5060
[Nov 4 16:13:16] DEBUG[14741] netsock2.c: Splitting ‘192.168.100.32:5060’ into…
[Nov 4 16:13:16] DEBUG[14741] netsock2.c: …host ‘192.168.100.32’ and port ‘5060’.
[Nov 4 16:13:16] VERBOSE[14741] chan_sip.c: Sending to 192.168.100.32:5060 (no NAT)
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Allocating new SIP dialog for 1695553247-5060-64@BJC.BGI.BAA.DC - INVITE (No RTP)
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] sip/reqresp_parser.c: Begin: parsing SIP “Supported: replaces, path, timer”
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] sip/reqresp_parser.c: Found SIP option: -replaces-
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] sip/reqresp_parser.c: Matched SIP option: replaces
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] sip/reqresp_parser.c: Found SIP option: -path-
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] sip/reqresp_parser.c: Matched SIP option: path
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] sip/reqresp_parser.c: Found SIP option: -timer-
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] sip/reqresp_parser.c: Matched SIP option: timer
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: Splitting ‘192.168.100.32:5060’ into…
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: …host ‘192.168.100.32’ and port ‘5060’.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Sending to 192.168.100.32:5060 (no NAT)
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Initializing initreq for method INVITE - callid 1695553247-5060-64@BJC.BGI.BAA.DC
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Using INVITE request as basis request - 1695553247-5060-64@BJC.BGI.BAA.DC
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: Splitting ‘192.168.100.21’ into…
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: …host ‘192.168.100.21’ and port ‘’.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found peer ‘1001’ for ‘1001’ from 192.168.100.32:5060
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c:
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/UDP 192.168.100.32:5060;branch=z9hG4bK1815663290;received=192.168.100.32;rport=5060
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 2 [ 54]: From: “Sergio” sip:1001@192.168.100.21;tag=259365463
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 3 [ 49]: To: sip:XXXXXXXXX@192.168.100.21;tag=as45df46a2
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 4 [ 42]: Call-ID: 1695553247-5060-64@BJC.BGI.BAA.DC
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 5 [ 16]: CSeq: 620 INVITE
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 6 [ 20]: Server: Asterisk PBX
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“3453120a”
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 10 [ 17]: Content-Length: 0
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 11 [ 0]:
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1601
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Trying to put ‘SIP/2.0 401’ onto UDP socket destined for 192.168.100.32:5060
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Scheduling destruction of SIP dialog '1695553247-5060-64@BJC.BGI.BAA.DC’ in 32000 ms (Method: INVITE)
[Nov 4 16:13:16] VERBOSE[14741] chan_sip.c:
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 0 [ 40]: ACK sip:XXXXXXXXX@192.168.100.21 SIP/2.0
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 192.168.100.32:5060;branch=z9hG4bK1815663290;rport
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 2 [ 54]: From: “Sergio” sip:1001@192.168.100.21;tag=259365463
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 3 [ 49]: To: sip:XXXXXXXXX@192.168.100.21;tag=as45df46a2
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 4 [ 42]: Call-ID: 1695553247-5060-64@BJC.BGI.BAA.DC
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 5 [ 13]: CSeq: 620 ACK
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 6 [ 17]: Content-Length: 0
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 7 [ 0]:
[Nov 4 16:13:16] VERBOSE[14741] chan_sip.c: — (7 headers 0 lines) —
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1601
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Stopping retransmission on '1695553247-5060-64@BJC.BGI.BAA.DC’ of Response 620: Match Found
[Nov 4 16:13:16] VERBOSE[14741] chan_sip.c:
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 0 [ 43]: INVITE sip:XXXXXXXXX@192.168.100.21 SIP/2.0
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 1 [ 66]: Via: SIP/2.0/UDP 192.168.100.32:5060;branch=z9hG4bK573606872;rport
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 2 [ 54]: From: “Sergio” sip:1001@192.168.100.21;tag=259365463
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 3 [ 34]: To: sip:XXXXXXXXX@192.168.100.21
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 4 [ 42]: Call-ID: 1695553247-5060-64@BJC.BGI.BAA.DC
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 5 [ 16]: CSeq: 621 INVITE
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 6 [ 48]: Contact: “Sergio” sip:1001@192.168.100.32:5060
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 7 [169]: Authorization: Digest username=“1001”, realm=“asterisk”, nonce=“3453120a”, uri="sip:XXXXXXXXX@192.168.100.21", response=“53d2dac71b43c9a28a3ca6dd46435ef6”, algorithm=MD5
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 8 [ 16]: Max-Forwards: 70
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 9 [ 40]: User-Agent: Grandstream GXP1625 1.0.1.12
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 10 [ 13]: Privacy: none
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 11 [ 56]: P-Preferred-Identity: “Sergio” sip:1001@192.168.100.21
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 12 [ 32]: Supported: replaces, path, timer
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 13 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 15 [ 47]: Accept: application/sdp, application/dtmf-relay
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 16 [ 21]: Content-Length: 337
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 17 [ 0]:
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 0 [ 3]: v=0
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 1 [ 38]: o=1001 8000 8000 IN IP4 192.168.100.32
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 2 [ 10]: s=SIP Call
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.100.32
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 4 [ 5]: t=0 0
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 5 [ 35]: m=audio 5004 RTP/AVP 0 8 18 9 2 101
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 6 [ 10]: a=sendrecv
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 8 [ 10]: a=ptime:20
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 9 [ 20]: a=rtpmap:8 PCMA/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 10 [ 21]: a=rtpmap:18 G729/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 11 [ 19]: a=fmtp:18 annexb=no
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 12 [ 20]: a=rtpmap:9 G722/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 13 [ 23]: a=rtpmap:2 G726-32/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 14 [ 33]: a=rtpmap:101 telephone-event/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 15 [ 15]: a=fmtp:101 0-15
[Nov 4 16:13:16] VERBOSE[14741] chan_sip.c: — (17 headers 16 lines) —
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: Splitting ‘192.168.100.32:5060’ into…
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: …host ‘192.168.100.32’ and port ‘5060’.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Sending to 192.168.100.32:5060 (no NAT)
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Initializing initreq for method INVITE - callid 1695553247-5060-64@BJC.BGI.BAA.DC
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Using INVITE request as basis request - 1695553247-5060-64@BJC.BGI.BAA.DC
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: Splitting ‘192.168.100.21’ into…
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: …host ‘192.168.100.21’ and port ‘’.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found peer ‘1001’ for ‘1001’ from 192.168.100.32:5060
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Using engine ‘asterisk’ for RTP instance ‘0xf3b3b294’
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] res_rtp_asterisk.c: Allocated port 10066 for RTP instance ‘0xf3b3b294’
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: RTP instance ‘0xf3b3b294’ is setup and ready to go
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] res_rtp_asterisk.c: Setup RTCP on RTP instance ‘0xf3b3b294’
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] netsock2.c: Using SIP RTP CoS mark 5
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Setting NAT on RTP to Off
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing session-level SDP v=0… UNSUPPORTED OR FAILED.
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing session-level SDP o=1001 8000 8000 IN IP4 192.168.100.32… OK.
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing session-level SDP s=SIP Call… UNSUPPORTED OR FAILED.
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: Splitting ‘192.168.100.32’ into…
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: …host ‘192.168.100.32’ and port ‘’.
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.100.32… OK.
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing session-level SDP t=0 0… UNSUPPORTED OR FAILED.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found RTP audio format 0
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Setting payload 0 (0xf3d27eac) based on m type on 0xf489085c
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found RTP audio format 8
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Setting payload 8 (0xf3d2799c) based on m type on 0xf489085c
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found RTP audio format 18
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Setting payload 18 (0xf3b3e83c) based on m type on 0xf489085c
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found RTP audio format 9
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Setting payload 9 (0xf3d0a864) based on m type on 0xf489085c
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found RTP audio format 2
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Setting payload 2 (0xf3d14be4) based on m type on 0xf489085c
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found RTP audio format 101
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Setting payload 101 (0xf3d48144) based on m type on 0xf489085c
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing media-level (audio) SDP a=sendrecv… OK.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found audio description format PCMU for ID 0
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000… OK.
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing media-level (audio) SDP a=ptime:20… OK.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found audio description format PCMA for ID 8
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000… OK.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found audio description format G729 for ID 18
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000… OK.
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no… OK.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found audio description format G722 for ID 9
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000… OK.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found audio description format G726-32 for ID 2
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:2 G726-32/8000… OK.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found audio description format telephone-event for ID 101
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000… OK.
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15… UNSUPPORTED OR FAILED.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Capabilities: us - (ulaw|gsm), peer - audio=(ulaw|g726|alaw|g722|g729)/video=(nothing)/text=(nothing), combined - (ulaw)
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] res_rtp_asterisk.c: Setting RTCP address on RTP instance ‘0xf3b3b294’
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Peer audio RTP is at port 192.168.100.32:5004
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Copying payload 0 (0xf3d89d7c) from 0xf489085c to 0xf3b3b3bc
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Copying payload 2 (0xf3d0a864) from 0xf489085c to 0xf3b3b3bc
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Copying payload 8 (0xf3d27eac) from 0xf489085c to 0xf3b3b3bc
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Copying payload 9 (0xf3b3e83c) from 0xf489085c to 0xf3b3b3bc
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Copying payload 18 (0xf3d2799c) from 0xf489085c to 0xf3b3b3bc
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Copying payload 101 (0xf3d14be4) from 0xf489085c to 0xf3b3b3bc
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance ‘0xf3b3b294’
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: We’re settling with these formats: (ulaw)
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Checking SIP call limits for device 1001
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Updating call counter for incoming call
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Call from peer ‘1001’ is 1 out of 2147483647
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: Splitting ‘192.168.100.21’ into…
[Nov 4 16:13:16] DEBUG[14477] devicestate.c: No provider found, checking channel drivers for SIP - 1001
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: …host ‘192.168.100.21’ and port ‘’.
[Nov 4 16:13:16] DEBUG[14477] chan_sip.c: Checking device state for peer 1001
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: Splitting ‘192.168.100.21’ into…
[Nov 4 16:13:16] DEBUG[14477] devicestate.c: Changing state for SIP/1001 - state 2 (In use)
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: …host ‘192.168.100.21’ and port ‘’.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Looking for XXXXXXXXX in DLPN_DialPlan1 (domain 192.168.100.21)
[Nov 4 16:13:16] DEBUG[14814] app_queue.c: Device ‘SIP/1001’ changed to state ‘2’ (In use) but we don’t care because they’re not a member of any queue.
[Nov 4 16:13:16] DEBUG[14674] app_queue.c: Extension ‘1001@default’ changed to state ‘2’ (In use) but we don’t care because they’re not a member of any queue.
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Incoming INVITE with ‘timer’ option supported
[Nov 4 16:13:16] DEBUG[14394] threadpool.c: Increasing threadpool stasis-core’s size by 1
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: *** Our native formats are (ulaw)
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: *** Joint capabilities are (ulaw)
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: *** Our capabilities are (ulaw|gsm)
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: This channel will not be able to handle video.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] sip/route.c: sip_route_dump: route/path hop: sip:1001@192.168.100.32:5060
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: SIP/1001-0000005f: New call is still down… Trying…
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c:
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 1 [ 95]: Via: SIP/2.0/UDP 192.168.100.32:5060;branch=z9hG4bK573606872;received=192.168.100.32;rport=5060
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 2 [ 54]: From: “Sergio” sip:1001@192.168.100.21;tag=259365463
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 3 [ 34]: To: sip:XXXXXXXXX@192.168.100.21
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 4 [ 42]: Call-ID: 1695553247-5060-64@BJC.BGI.BAA.DC
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 5 [ 16]: CSeq: 621 INVITE
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 6 [ 20]: Server: Asterisk PBX
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 10 [ 44]: Contact: sip:XXXXXXXXX@192.168.100.21:5060
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 11 [ 17]: Content-Length: 0
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 12 [ 0]:
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Trying to put ‘SIP/2.0 100’ onto UDP socket destined for 192.168.100.32:5060
[Nov 4 16:13:16] DEBUG[14477] devicestate.c: No provider found, checking channel drivers for SIP - 1001
[Nov 4 16:13:16] DEBUG[14477] chan_sip.c: Checking device state for peer 1001
[Nov 4 16:13:16] DEBUG[14477] devicestate.c: Changing state for SIP/1001 - state 2 (In use)
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘ROUTER’ is ‘SIP/ROUTER’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘EXTEN’ is ‘XXXXXXXXX’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Launching ‘Macro’
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] pbx.c: Executing [XXXXXXXXX@DLPN_DialPlan1:1] Macro(“SIP/1001-0000005f”, “trunkdial-failover-0.3,SIP/ROUTER/XXXXXXXXX,ROUTER,”) in new stack
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘FMCIDNUM’ is NULL
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Function LEN() result is ‘0’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Expression result is ‘0’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Launching ‘GotoIf’
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] pbx.c: Executing [s@macro-trunkdial-failover-0.3:1] GotoIf(“SIP/1001-0000005f”, “0?1-fmsetcid,1”) in new stack
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Not taking any branch
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] app_macro.c: Executed application: GotoIf
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘GLOBAL_OUTBOUNDCIDNAME’ is ‘’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Function LEN() result is ‘0’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Expression result is ‘0’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Launching ‘GotoIf’
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] pbx.c: Executing [s@macro-trunkdial-failover-0.3:2] GotoIf(“SIP/1001-0000005f”, “0?1-setgbobname,1”) in new stack
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Not taking any branch
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] app_macro.c: Executed application: GotoIf
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Function CALLERID(num) result is ‘1001’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘CID_1001’ is NULL
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Function LEN() result is ‘0’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Expression result is ‘0’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Function CALLERID(num) result is ‘1001’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘CID_1001’ is NULL
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Function IF(0?:slight_smile: result is ‘’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Launching ‘Set’
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] pbx.c: Executing [s@macro-trunkdial-failover-0.3:3] Set(“SIP/1001-0000005f”, “CALLERID(num)=”) in new stack
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] app_macro.c: Executed application: Set
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘ARG5’ is NULL
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Function LEN() result is ‘0’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Expression result is ‘0’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘ARG5’ is NULL
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Function IF(0?:slight_smile: result is ‘’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Launching ‘Set’
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] pbx.c: Executing [s@macro-trunkdial-failover-0.3:4] Set(“SIP/1001-0000005f”, “CALLERID(all)=”) in new stack
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] app_macro.c: Executed application: Set
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Function CALLERID(num) result is ‘’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Function LEN() result is ‘0’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Expression result is ‘0’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Launching ‘GotoIf’
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] pbx.c: Executing [s@macro-trunkdial-failover-0.3:5] GotoIf(“SIP/1001-0000005f”, “0?1-dial,1”) in new stack
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Not taking any branch
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] app_macro.c: Executed application: GotoIf
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘ARG3’ is ‘ROUTER’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘CID_ROUTER’ is NULL
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Function LEN() result is ‘0’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Expression result is ‘0’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘ARG3’ is ‘ROUTER’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘CID_ROUTER’ is NULL
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘GLOBAL_OUTBOUNDCID’ is ‘’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Function IF(0?:slight_smile: result is ‘’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Launching ‘Set’
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] pbx.c: Executing [s@macro-trunkdial-failover-0.3:6] Set(“SIP/1001-0000005f”, “CALLERID(all)=”) in new stack
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] app_macro.c: Executed application: Set
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘ARG5’ is NULL
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Function LEN() result is ‘0’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Expression result is ‘0’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘ARG5’ is NULL
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Function IF(0?:slight_smile: result is ‘’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Launching ‘Set’
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] pbx.c: Executing [s@macro-trunkdial-failover-0.3:7] Set(“SIP/1001-0000005f”, “CALLERID(all)=”) in new stack
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] app_macro.c: Executed application: Set
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Launching ‘Goto’
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] pbx.c: Executing [s@macro-trunkdial-failover-0.3:8] Goto(“SIP/1001-0000005f”, “1-dial,1”) in new stack
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] pbx.c: Goto (macro-trunkdial-failover-0.3,1-dial,1)
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] app_macro.c: Executed application: Goto
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘ARG1’ is ‘SIP/ROUTER/XXXXXXXXX’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Launching ‘Dial’
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] pbx.c: Executing [1-dial@macro-trunkdial-failover-0.3:1] Dial(“SIP/1001-0000005f”, “SIP/ROUTER/XXXXXXXXX”) in new stack
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Asked to create a SIP channel with formats: (ulaw)
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Allocating new SIP dialog for 179f4c2e083d5b6842f220787003208a@192.168.100.22:5060 - INVITE (No RTP)
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] rtp_engine.c: Using engine ‘asterisk’ for RTP instance ‘0xf46466f4’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] res_rtp_asterisk.c: Allocated port 19714 for RTP instance ‘0xf46466f4’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] rtp_engine.c: RTP instance ‘0xf46466f4’ is setup and ready to go
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] res_rtp_asterisk.c: Setup RTCP on RTP instance ‘0xf46466f4’
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] netsock2.c: Using SIP RTP CoS mark 5
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Setting NAT on RTP to Off
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: OBPROXY: Not applying OBproxy to this call
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] acl.c: For destination ‘192.168.1.1’, our source address is ‘192.168.100.21’.
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Setting AST_TRANSPORT_UDP with address 192.168.100.21:5060
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Setting NAT on RTP to Off
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: SIP call-id changed from ‘179f4c2e083d5b6842f220787003208a@192.168.100.22:5060’ to ‘17ed7ac67cb18fc83bc73ff972eeb94e@192.168.100.21:5060’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: *** Our native formats are (ulaw)
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: *** Joint capabilities are (ulaw)
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: *** Our capabilities are (ulaw|alaw|gsm|g726|g729)
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: *** Our preferred formats from the incoming channel are (ulaw)
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: This channel will not be able to handle video.
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] channel_internal_api.c: Channel Call ID changing from [C-000001bf] to [C-000001bf]
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] rtp_engine.c: Copying payload 0 (0xf3d89d7c) from 0xf3b3b3bc to 0xf464681c
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] rtp_engine.c: Copying payload 2 (0xf3d0a864) from 0xf3b3b3bc to 0xf464681c
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] rtp_engine.c: Copying payload 8 (0xf3d27eac) from 0xf3b3b3bc to 0xf464681c
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] rtp_engine.c: Copying payload 9 (0xf3b3e83c) from 0xf3b3b3bc to 0xf464681c
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] rtp_engine.c: Copying payload 18 (0xf3d2799c) from 0xf3b3b3bc to 0xf464681c
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] rtp_engine.c: Copying payload 101 (0xf3d14be4) from 0xf3b3b3bc to 0xf464681c
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] rtp_engine.c: Seeded SDP of ‘SIP/ROUTER-00000060’ with that of ‘SIP/1001-0000005f’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Outgoing Call for XXXXXXXXX
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Updating call counter for outgoing call
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: ** Our capability: (ulaw|alaw|gsm|g726|g729) Video flag: False Text flag: False
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: ** Our prefcodec: (ulaw)
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] chan_sip.c: Audio is at 19714
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] chan_sip.c: Adding codec ulaw to SDP
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] chan_sip.c: Adding codec alaw to SDP
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] chan_sip.c: Adding codec gsm to SDP
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] chan_sip.c: Adding codec g726 to SDP
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] chan_sip.c: Adding codec g729 to SDP
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: – Done with adding codecs to SDP
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Done building SDP. Settling with this capability: (ulaw|alaw|gsm|g726|g729)
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Initializing initreq for method INVITE - callid 17ed7ac67cb18fc83bc73ff972eeb94e@192.168.100.21:5060
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 0 [ 40]: INVITE sip:XXXXXXXXX@192.168.1.1 SIP/2.0
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.100.21:5060;branch=z9hG4bK1f56d062
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 3 [ 61]: From: “asterisk” sip:asterisk@192.168.100.21;tag=as410dcd29
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 4 [ 31]: To: sip:XXXXXXXXX@192.168.1.1
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 5 [ 43]: Contact: sip:asterisk@192.168.100.21:5060
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 6 [ 61]: Call-ID: 17ed7ac67cb18fc83bc73ff972eeb94e@192.168.100.21:5060
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 8 [ 24]: User-Agent: Asterisk PBX
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 9 [ 35]: Date: Wed, 04 Nov 2015 15:13:16 GMT
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.1.1:5060:
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 0 [ 40]: INVITE sip:XXXXXXXXX@192.168.1.1 SIP/2.0
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.100.21:5060;branch=z9hG4bK1f56d062
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 3 [ 61]: From: “asterisk” sip:asterisk@192.168.100.21;tag=as410dcd29
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 4 [ 31]: To: sip:XXXXXXXXX@192.168.1.1
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 5 [ 43]: Contact: sip:asterisk@192.168.100.21:5060
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 6 [ 61]: Call-ID: 17ed7ac67cb18fc83bc73ff972eeb94e@192.168.100.21:5060
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 8 [ 24]: User-Agent: Asterisk PBX
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 9 [ 35]: Date: Wed, 04 Nov 2015 15:13:16 GMT
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 13 [ 19]: Content-Length: 374
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 14 [ 0]:
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 0 [ 3]: v=0
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 1 [ 48]: o=root 262529552 262529552 IN IP4 192.168.100.21
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 2 [ 21]: s=Asterisk PBX 13.1.0
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.100.21
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 4 [ 5]: t=0 0
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 5 [ 36]: m=audio 19714 RTP/AVP 0 8 3 2 18 101
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 8 [ 19]: a=rtpmap:3 GSM/8000
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 9 [ 23]: a=rtpmap:2 G726-32/8000
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 10 [ 21]: a=rtpmap:18 G729/8000
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 11 [ 19]: a=fmtp:18 annexb=no
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 12 [ 33]: a=rtpmap:101 telephone-event/8000
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 13 [ 15]: a=fmtp:101 0-16
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 14 [ 10]: a=ptime:20
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 15 [ 14]: a=maxptime:150
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 16 [ 10]: a=sendrecv
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1604
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Trying to put ‘INVITE sip:’ onto UDP socket destined for 192.168.1.1:5060
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] app_dial.c: Called SIP/ROUTER/XXXXXXXXX
[Nov 4 16:13:16] VERBOSE[14741] chan_sip.c:
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 1 [ 80]: Via: SIP/2.0/UDP 192.168.100.21:5060;branch=z9hG4bK1f56d062;received=192.168.1.2
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 2 [ 61]: From: “asterisk” sip:asterisk@192.168.100.21;tag=as410dcd29
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 3 [ 68]: To: sip:XXXXXXXXX@192.168.1.1;tag=490977b9c933825154b0b9671e5afc51
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 4 [ 61]: Call-ID: 17ed7ac67cb18fc83bc73ff972eeb94e@192.168.100.21:5060
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 6 [ 43]: User-Agent: DSL Router/DSL Router-00.96.315
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 7 [ 17]: Content-Length: 0
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 8 [ 0]:
[Nov 4 16:13:16] VERBOSE[14741] chan_sip.c: — (8 headers 0 lines) —
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Acked pending invite 102
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1604
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Stopping retransmission on ‘17ed7ac67cb18fc83bc73ff972eeb94e@192.168.100.21:5060’ of Request 102: Match Found
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: SIP response 404 to standard invite
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Transmitting (no NAT) to 192.168.1.1:5060:
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 0 [ 37]: ACK sip:XXXXXXXXX@192.168.1.1 SIP/2.0
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.100.21:5060;branch=z9hG4bK1f56d062
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 3 [ 61]: From: “asterisk” sip:asterisk@192.168.100.21;tag=as410dcd29
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 4 [ 68]: To: sip:XXXXXXXXX@192.168.1.1;tag=490977b9c933825154b0b9671e5afc51
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 5 [ 43]: Contact: sip:asterisk@192.168.100.21:5060
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 6 [ 61]: Call-ID: 17ed7ac67cb18fc83bc73ff972eeb94e@192.168.100.21:5060
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 7 [ 13]: CSeq: 102 ACK
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 8 [ 24]: User-Agent: Asterisk PBX
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 9 [ 17]: Content-Length: 0
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 10 [ 0]:
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Trying to put ‘ACK sip:644’ onto UDP socket destined for 192.168.1.1:5060
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] channel.c: Hanging up channel ‘SIP/ROUTER-00000060’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Hangup call SIP/ROUTER-00000060, SIP callid 17ed7ac67cb18fc83bc73ff972eeb94e@192.168.100.21:5060
[Nov 4 16:13:16] DEBUG[14664] cdr.c: Finalized CDR for SIP/1001-0000005f - start 1446649996.779049 answer 0.000000 end 1446649996.819008 dispo NO ANSWER
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] res_rtp_asterisk.c: Setting RTCP address on RTP instance ‘0xf46466f4’
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] chan_sip.c: Scheduling destruction of SIP dialog ‘17ed7ac67cb18fc83bc73ff972eeb94e@192.168.100.21:5060’ in 32000 ms (Method: INVITE)
[Nov 4 16:13:16] DEBUG[14664] cdr.c: Finalized CDR for SIP/ROUTER-00000060 - start 1446649996.789172 answer 0.000000 end 1446649996.819905 dispo NO ANSWER
[Nov 4 16:13:16] DEBUG[14664] cdr.c: CDR for SIP/ROUTER-00000060 is dialed and has no Party B; discarding

Hi.

Well, i check the debug option an i can view this information:

[Nov 4 16:13:16] VERBOSE[14741] chan_sip.c:
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 0 [ 43]: INVITE sip:XXXXXXXXX@192.168.100.21 SIP/2.0
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 192.168.100.32:5060;branch=z9hG4bK1815663290;rport
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 2 [ 54]: From: “Sergio” sip:1001@192.168.100.21;tag=259365463
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 3 [ 34]: To: sip:XXXXXXXXX@192.168.100.21
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 4 [ 42]: Call-ID: 1695553247-5060-64@BJC.BGI.BAA.DC
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 5 [ 16]: CSeq: 620 INVITE
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 6 [ 48]: Contact: “Sergio” sip:1001@192.168.100.32:5060
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 8 [ 40]: User-Agent: Grandstream GXP1625 1.0.1.12
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 9 [ 13]: Privacy: none
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 10 [ 56]: P-Preferred-Identity: “Sergio” sip:1001@192.168.100.21
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 11 [ 32]: Supported: replaces, path, timer
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 12 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 14 [ 47]: Accept: application/sdp, application/dtmf-relay
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 15 [ 21]: Content-Length: 337
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 16 [ 0]:
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 0 [ 3]: v=0
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 1 [ 38]: o=1001 8000 8000 IN IP4 192.168.100.32
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 2 [ 10]: s=SIP Call
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.100.32
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 4 [ 5]: t=0 0
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 5 [ 35]: m=audio 5004 RTP/AVP 0 8 18 9 2 101
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 6 [ 10]: a=sendrecv
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 8 [ 10]: a=ptime:20
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 9 [ 20]: a=rtpmap:8 PCMA/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 10 [ 21]: a=rtpmap:18 G729/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 11 [ 19]: a=fmtp:18 annexb=no
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 12 [ 20]: a=rtpmap:9 G722/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 13 [ 23]: a=rtpmap:2 G726-32/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 14 [ 33]: a=rtpmap:101 telephone-event/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 15 [ 15]: a=fmtp:101 0-15
[Nov 4 16:13:16] VERBOSE[14741] chan_sip.c: — (16 headers 16 lines) —
[Nov 4 16:13:16] DEBUG[14741] acl.c: For destination ‘192.168.100.32’, our source address is ‘192.168.100.21’.
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Setting AST_TRANSPORT_UDP with address 192.168.100.21:5060
[Nov 4 16:13:16] DEBUG[14741] netsock2.c: Splitting ‘192.168.100.32:5060’ into…
[Nov 4 16:13:16] DEBUG[14741] netsock2.c: …host ‘192.168.100.32’ and port ‘5060’.
[Nov 4 16:13:16] VERBOSE[14741] chan_sip.c: Sending to 192.168.100.32:5060 (no NAT)
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Allocating new SIP dialog for 1695553247-5060-64@BJC.BGI.BAA.DC - INVITE (No RTP)
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] sip/reqresp_parser.c: Begin: parsing SIP “Supported: replaces, path, timer”
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] sip/reqresp_parser.c: Found SIP option: -replaces-
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] sip/reqresp_parser.c: Matched SIP option: replaces
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] sip/reqresp_parser.c: Found SIP option: -path-
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] sip/reqresp_parser.c: Matched SIP option: path
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] sip/reqresp_parser.c: Found SIP option: -timer-
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] sip/reqresp_parser.c: Matched SIP option: timer
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: Splitting ‘192.168.100.32:5060’ into…
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: …host ‘192.168.100.32’ and port ‘5060’.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Sending to 192.168.100.32:5060 (no NAT)
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Initializing initreq for method INVITE - callid 1695553247-5060-64@BJC.BGI.BAA.DC
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Using INVITE request as basis request - 1695553247-5060-64@BJC.BGI.BAA.DC
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: Splitting ‘192.168.100.21’ into…
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: …host ‘192.168.100.21’ and port ‘’.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found peer ‘1001’ for ‘1001’ from 192.168.100.32:5060
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c:
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/UDP 192.168.100.32:5060;branch=z9hG4bK1815663290;received=192.168.100.32;rport=5060
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 2 [ 54]: From: “Sergio” sip:1001@192.168.100.21;tag=259365463
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 3 [ 49]: To: sip:XXXXXXXXX@192.168.100.21;tag=as45df46a2
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 4 [ 42]: Call-ID: 1695553247-5060-64@BJC.BGI.BAA.DC
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 5 [ 16]: CSeq: 620 INVITE
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 6 [ 20]: Server: Asterisk PBX
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“3453120a”
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 10 [ 17]: Content-Length: 0
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 11 [ 0]:
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1601
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Trying to put ‘SIP/2.0 401’ onto UDP socket destined for 192.168.100.32:5060
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Scheduling destruction of SIP dialog '1695553247-5060-64@BJC.BGI.BAA.DC’ in 32000 ms (Method: INVITE)
[Nov 4 16:13:16] VERBOSE[14741] chan_sip.c:
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 0 [ 40]: ACK sip:XXXXXXXXX@192.168.100.21 SIP/2.0
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 192.168.100.32:5060;branch=z9hG4bK1815663290;rport
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 2 [ 54]: From: “Sergio” sip:1001@192.168.100.21;tag=259365463
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 3 [ 49]: To: sip:XXXXXXXXX@192.168.100.21;tag=as45df46a2
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 4 [ 42]: Call-ID: 1695553247-5060-64@BJC.BGI.BAA.DC
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 5 [ 13]: CSeq: 620 ACK
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 6 [ 17]: Content-Length: 0
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 7 [ 0]:
[Nov 4 16:13:16] VERBOSE[14741] chan_sip.c: — (7 headers 0 lines) —
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1601
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Stopping retransmission on '1695553247-5060-64@BJC.BGI.BAA.DC’ of Response 620: Match Found
[Nov 4 16:13:16] VERBOSE[14741] chan_sip.c:
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 0 [ 43]: INVITE sip:XXXXXXXXX@192.168.100.21 SIP/2.0
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 1 [ 66]: Via: SIP/2.0/UDP 192.168.100.32:5060;branch=z9hG4bK573606872;rport
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 2 [ 54]: From: “Sergio” sip:1001@192.168.100.21;tag=259365463
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 3 [ 34]: To: sip:XXXXXXXXX@192.168.100.21
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 4 [ 42]: Call-ID: 1695553247-5060-64@BJC.BGI.BAA.DC
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 5 [ 16]: CSeq: 621 INVITE
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 6 [ 48]: Contact: “Sergio” sip:1001@192.168.100.32:5060
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 7 [169]: Authorization: Digest username=“1001”, realm=“asterisk”, nonce=“3453120a”, uri="sip:XXXXXXXXX@192.168.100.21", response=“53d2dac71b43c9a28a3ca6dd46435ef6”, algorithm=MD5
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 8 [ 16]: Max-Forwards: 70
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 9 [ 40]: User-Agent: Grandstream GXP1625 1.0.1.12
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 10 [ 13]: Privacy: none
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 11 [ 56]: P-Preferred-Identity: “Sergio” sip:1001@192.168.100.21
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 12 [ 32]: Supported: replaces, path, timer
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 13 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 15 [ 47]: Accept: application/sdp, application/dtmf-relay
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 16 [ 21]: Content-Length: 337
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 17 [ 0]:
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 0 [ 3]: v=0
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 1 [ 38]: o=1001 8000 8000 IN IP4 192.168.100.32
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 2 [ 10]: s=SIP Call
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.100.32
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 4 [ 5]: t=0 0
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 5 [ 35]: m=audio 5004 RTP/AVP 0 8 18 9 2 101
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 6 [ 10]: a=sendrecv
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 8 [ 10]: a=ptime:20
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 9 [ 20]: a=rtpmap:8 PCMA/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 10 [ 21]: a=rtpmap:18 G729/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 11 [ 19]: a=fmtp:18 annexb=no
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 12 [ 20]: a=rtpmap:9 G722/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 13 [ 23]: a=rtpmap:2 G726-32/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 14 [ 33]: a=rtpmap:101 telephone-event/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 15 [ 15]: a=fmtp:101 0-15
[Nov 4 16:13:16] VERBOSE[14741] chan_sip.c: — (17 headers 16 lines) —
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: Splitting ‘192.168.100.32:5060’ into…
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: …host ‘192.168.100.32’ and port ‘5060’.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Sending to 192.168.100.32:5060 (no NAT)
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Initializing initreq for method INVITE - callid 1695553247-5060-64@BJC.BGI.BAA.DC
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Using INVITE request as basis request - 1695553247-5060-64@BJC.BGI.BAA.DC
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: Splitting ‘192.168.100.21’ into…
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: …host ‘192.168.100.21’ and port ‘’.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found peer ‘1001’ for ‘1001’ from 192.168.100.32:5060
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Using engine ‘asterisk’ for RTP instance ‘0xf3b3b294’
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] res_rtp_asterisk.c: Allocated port 10066 for RTP instance ‘0xf3b3b294’
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: RTP instance ‘0xf3b3b294’ is setup and ready to go
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] res_rtp_asterisk.c: Setup RTCP on RTP instance ‘0xf3b3b294’
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] netsock2.c: Using SIP RTP CoS mark 5
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Setting NAT on RTP to Off
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing session-level SDP v=0… UNSUPPORTED OR FAILED.
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing session-level SDP o=1001 8000 8000 IN IP4 192.168.100.32… OK.
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing session-level SDP s=SIP Call… UNSUPPORTED OR FAILED.
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: Splitting ‘192.168.100.32’ into…
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: …host ‘192.168.100.32’ and port ‘’.
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.100.32… OK.
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing session-level SDP t=0 0… UNSUPPORTED OR FAILED.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found RTP audio format 0
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Setting payload 0 (0xf3d27eac) based on m type on 0xf489085c
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found RTP audio format 8
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Setting payload 8 (0xf3d2799c) based on m type on 0xf489085c
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found RTP audio format 18
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Setting payload 18 (0xf3b3e83c) based on m type on 0xf489085c
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found RTP audio format 9
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Setting payload 9 (0xf3d0a864) based on m type on 0xf489085c
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found RTP audio format 2
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Setting payload 2 (0xf3d14be4) based on m type on 0xf489085c
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found RTP audio format 101
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Setting payload 101 (0xf3d48144) based on m type on 0xf489085c
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing media-level (audio) SDP a=sendrecv… OK.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found audio description format PCMU for ID 0
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000… OK.
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing media-level (audio) SDP a=ptime:20… OK.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found audio description format PCMA for ID 8
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000… OK.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found audio description format G729 for ID 18
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000… OK.
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no… OK.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found audio description format G722 for ID 9
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000… OK.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found audio description format G726-32 for ID 2
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:2 G726-32/8000… OK.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found audio description format telephone-event for ID 101
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000… OK.
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15… UNSUPPORTED OR FAILED.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Capabilities: us - (ulaw|gsm), peer - audio=(ulaw|g726|alaw|g722|g729)/video=(nothing)/text=(nothing), combined - (ulaw)
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] res_rtp_asterisk.c: Setting RTCP address on RTP instance ‘0xf3b3b294’
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Peer audio RTP is at port 192.168.100.32:5004
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Copying payload 0 (0xf3d89d7c) from 0xf489085c to 0xf3b3b3bc
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Copying payload 2 (0xf3d0a864) from 0xf489085c to 0xf3b3b3bc
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Copying payload 8 (0xf3d27eac) from 0xf489085c to 0xf3b3b3bc
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Copying payload 9 (0xf3b3e83c) from 0xf489085c to 0xf3b3b3bc
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Copying payload 18 (0xf3d2799c) from 0xf489085c to 0xf3b3b3bc
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Copying payload 101 (0xf3d14be4) from 0xf489085c to 0xf3b3b3bc
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance ‘0xf3b3b294’
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: We’re settling with these formats: (ulaw)
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Checking SIP call limits for device 1001
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Updating call counter for incoming call
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Call from peer ‘1001’ is 1 out of 2147483647
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: Splitting ‘192.168.100.21’ into…
[Nov 4 16:13:16] DEBUG[14477] devicestate.c: No provider found, checking channel drivers for SIP - 1001
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: …host ‘192.168.100.21’ and port ‘’.
[Nov 4 16:13:16] DEBUG[14477] chan_sip.c: Checking device state for peer 1001
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: Splitting ‘192.168.100.21’ into…
[Nov 4 16:13:16] DEBUG[14477] devicestate.c: Changing state for SIP/1001 - state 2 (In use)
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: …host ‘192.168.100.21’ and port ‘’.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Looking for XXXXXXXXX in DLPN_DialPlan1 (domain 192.168.100.21)
[Nov 4 16:13:16] DEBUG[14814] app_queue.c: Device ‘SIP/1001’ changed to state ‘2’ (In use) but we don’t care because they’re not a member of any queue.
[Nov 4 16:13:16] DEBUG[14674] app_queue.c: Extension ‘1001@default’ changed to state ‘2’ (In use) but we don’t care because they’re not a member of any queue.
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Incoming INVITE with ‘timer’ option supported
[Nov 4 16:13:16] DEBUG[14394] threadpool.c: Increasing threadpool stasis-core’s size by 1
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: *** Our native formats are (ulaw)
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: *** Joint capabilities are (ulaw)
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: *** Our capabilities are (ulaw|gsm)
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: This channel will not be able to handle video.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] sip/route.c: sip_route_dump: route/path hop: sip:1001@192.168.100.32:5060
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: SIP/1001-0000005f: New call is still down… Trying…
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c:
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 1 [ 95]: Via: SIP/2.0/UDP 192.168.100.32:5060;branch=z9hG4bK573606872;received=192.168.100.32;rport=5060
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 2 [ 54]: From: “Sergio” sip:1001@192.168.100.21;tag=259365463
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 3 [ 34]: To: sip:XXXXXXXXX@192.168.100.21
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 4 [ 42]: Call-ID: 1695553247-5060-64@BJC.BGI.BAA.DC

Well, i change the debug level and now i have more information but i’m trying to reply but can’t paste the debu text…