Could not resolve socket address error

Asterisk Asterisk Support
1

Hi,

I installed Asterisk/11.13.1 on my VPS Debian 5.0.8 with IP public address.

I can’t make a call from one of my ISP’s, and it always show “Could not resolve socket address” error.

From asterisk log, there is “masquerade” address (00d5cc24I4), but from asterisk cli “show peers” it show 202.73.225.63 , is this “masquerade” address causes the problem ? My asterisk server is on VPS with public address

[Oct 22 11:54:02] ERROR[16489][C-0000000f]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("00d5cc24I4", "1024", ...): No address associated with hostname [Oct 22 11:54:02] WARNING[16489][C-0000000f]: chan_sip.c:18026 check_via: Could not resolve socket address for '00d5cc24I4:1024' == Using SIP RTP CoS mark 5 [Oct 22 11:54:02] ERROR[16489][C-0000000f]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("00d5cc04I4", "(null)", ...): No address associated with hostname [Oct 22 11:54:02] WARNING[16489][C-0000000f]: chan_sip.c:10803 process_sdp_c: Unable to lookup RTP Audio host in c= line, 'IN IP4 00d5cc04I4' [Oct 22 11:54:02] ERROR[16489][C-0000000f]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("00d5cc04I4", "(null)", ...): No address associated with hostname [Oct 22 11:54:02] WARNING[16489][C-0000000f]: chan_sip.c:10803 process_sdp_c: Unable to lookup RTP Audio host in c= line, 'IN IP4 00d5cc04I4' [Oct 22 11:54:02] WARNING[16489][C-0000000f]: chan_sip.c:10394 process_sdp: Insufficient information in SDP (c=)...

CLI>sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description 6000/6000 202.73.225.63 D Yes Yes 2649 Unmonitored 6001/6001 180.252.201.139 D Yes Yes 5060 Unmonitored

I have installed zoiper, linphone etc but the problem still occured.

How to deal with this ISP ?

2

This look like a DNS issue.

3

I’ve set DNS to 8.8.8.8/8.8.4.4 on both server and client, they can resolve hostnames. I also change this settings (based on my search result, there’s DNS issue with Asterisk 1.8):

srvlookup=no ; // sip.conf file enable=yes ; // dnsmgr.conf file
SIP packet capture :

[code]
INVITE sip:6001@110.232… SIP/2.0
Via: SIP/2.0/UDP 00d5cc24I4:5060;rport;branch=z9hG4bKPjpza1V4nidK-WyyWqJVlXBQC5ENo7NO9k
From: “6000” sip:6000@110.232....;tag=EjOpUg.RIS43P0fpCFIRRlkTbRwDZ21K
To: sip:6001@110.232…
Contact: “6000” sip:6000@00d5cc24I4:5060
Call-ID: B8MvXY3HnYHR0M4LdXldhNb1ekVKvJne
CSeq: 11477 INVITE
Route: sip:110.232....;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: ABTO Software VoIP SDK/darwin
Authorization: Digest username=“6000”, realm=“asterisk”, nonce=“29e9a78e”, uri=“sip:6001@110.232…”, response=“31b33a13fd153b67e91e1364f152d31b”, algorithm=MD5
Content-Type: application/sdp
Content-Length: 364
v=0
o=- 3623047108 3623047108 IN IP4 00d5cc04I4
s=sip_voip
c=IN IP4 00d5cc04I4
b=AS:84
t=0 0
a=X-nat:0
m=audio 5062 RTP/AVP 0 8 18 101
c=IN IP4 00d5cc04I4
b=TIAS:64000
a=rtcp:5063 IN IP4 192.168.0.101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 00d5cc24I4:5060;branch=z9hG4bKPjpza1V4nidK-WyyWqJVlXBQC5ENo7NO9k;received=114.79.12.34;rport=22208
From: “6000” sip:6000@110.232....;tag=EjOpUg.RIS43P0fpCFIRRlkTbRwDZ21K
To: sip:6001@110.232…;tag=as3fd8576d
Call-ID: B8MvXY3HnYHR0M4LdXldhNb1ekVKvJne
CSeq: 11477 INVITE
Server: Asterisk PBX 11.12.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Content-Length: 0[/code]
I keep getting “488 Not acceptable here” error (I forgot to mention this on my first post), any clue what should I do next ?

4

Not acceptable means that none of the codecs that they have offered are in your allowed list of codecs for that peer.

5

This problem is only happened if I switch ISP network (I have three ISP and this problem only occured on one ISP, the other two ISP normal, I can make and received call) and I never change codec settings on server or client.

Now I only allow GSM codec on Server and Client, but I still got error “488 Not acceptable here”.

[code]root@sip2:~# asterisk -rx “sip show settings”

Global Signalling Settings:

Codecs: (gsm)
Codec Order: gsm:20
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: No[/code]

Below I compared Failed and Success packet.

Failed SIP packet (INVITE—>488 Not acceptable here)

[code]INVITE sip:6001@110.232… SIP/2.0
Via: SIP/2.0/UDP 00d5cc24I4:5060;rport;branch=z9hG4bKPjdNoof5reexY2w7TkAssPLvvAi971-aMX
From: “6000” sip:6000@110.232.....;tag=vCBZzYzvDNNI5Bc7MtuC1Cn3GReN6fxt
To: sip:6001@110.232…
Contact: “6000” sip:6000@00d5cc24I4:5060

v=0
o=- 3623111004 3623111004 IN IP4 00d5cc04I4
s=sip_voip
c=IN IP4 00d5cc04I4
b=AS:30
t=0 0
a=X-nat:0
m=audio 5064 RTP/AVP 3 101
c=IN IP4 00d5cc04I4
b=TIAS:1320
a=rtcp:5065 IN IP4 192.168.0.101
a=sendrecv
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 00d5cc24I4:5060;branch=z9hG4bKPjdNoof5reexY2w7TkAssPLvvAi971-aMX;received=114.79.13.112;rport=25107
From: “6000” sip:6000@110.232.....;tag=vCBZzYzvDNNI5Bc7MtuC1Cn3GReN6fxt[/code]

Success SIP packet (INVITE—>Trying)

[code]INVITE sip:6001@110.232…:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.115:5060;rport;branch=z9hG4bKPjdxK0vROjTluUSg3-W7XJe5HvHConysB9
Max-Forwards: 70
From: “6000” sip:6000@110.232.....:5060;tag=fj7fNdh.1BbhJ6IP0h-Rvz8X0LXA3agO
To: sip:6001@110.232…:5060
Contact: “6000” sip:6000@192.168.0.115:5060

v=0
o=- 3623111540 3623111540 IN IP4 180.252.201.139
s=sip_voip
c=IN IP4 180.252.201.139
b=AS:30
t=0 0
a=X-nat:0
m=audio 5068 RTP/AVP 3 101
c=IN IP4 192.168.0.115
b=TIAS:13200
a=rtcp:5069 IN IP4 192.168.0.115
a=sendrecv
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.115:5060;branch=z9hG4bKPjdxK0vROjTluUSg3-W7XJe5HvHConysB9;received=180.252.201.139;rport=1032
From: “6000” sip:6000@110.232.....:5060;tag=fj7fNdh.1BbhJ6IP0h-Rvz8X0LXA3agO [/code]

6

enable debug level 5. Enable the full log in logger.conf, and use the full log, not a screen scrape. That should give information on why Asterisk didn’t like the offer.

Trying tends to be sent even if the call will then get rejected.

7

Here’s asterisk full log :
[FAILED]

[quote]
SIP Header

[code]<— SIP read from UDP:114.79.13.40:7430 —>
INVITE sip:600@110.232… SIP/2.0
Via: SIP/2.0/UDP 00d5cc24I4:5060;rport;branch=z9hG4bKPjsNgKfpJLIpM1ooTVO5Em-VLKzZCZP18R
Max-Forwards: 70
From: “6000” sip:6000@110.232.....;tag=UpwbPuuTzcHI-704U4FhuJ4ZZOPHub9R
To: sip:600@110.232…
Contact: “6000” sip:6000@00d5cc24I4:5060
Call-ID: .J-eGtdilRFt-UP84twohqJBMpfyvdqV
CSeq: 2156 INVITE
Route: sip:110.232.....;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: ABTO Software VoIP SDK/darwin
Authorization: Digest username=“6000”, realm=“asterisk”, nonce=“3a325ff1”, uri=“sip:600@110.232…”, response=“0e48b91fb82485e05408da526a8d8eda”, algorithm=MD5
Content-Type: application/sdp
Content-Length: 292

v=0
o=- 3623369868 3623369868 IN IP4 00d5cc04I4
s=sip_voip
c=IN IP4 00d5cc04I4
b=AS:30
t=0 0
a=X-nat:0
m=audio 5068 RTP/AVP 3 101
c=IN IP4 00d5cc04I4
b=TIAS:13200
a=rtcp:5069 IN IP4 192.168.0.100
a=sendrecv
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15[/code]
/var/log/asterisk/full :

[Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Header 0 [ 36]: INVITE sip:600@110.232..... SIP/2.0 [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Header 1 [ 87]: Via: SIP/2.0/UDP 00d5cc24I4:5060;rport;branch=z9hG4bKPjsNgKfpJLIpM1ooTVO5Em-VLKzZCZP18R [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Header 3 [ 74]: From: "6000" <sip:6000@110.232.....>;tag=UpwbPuuTzcHI-704U4FhuJ4ZZOPHub9R [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Header 4 [ 25]: To: sip:600@110.232..... [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Header 5 [ 42]: Contact: "6000" <sip:6000@00d5cc24I4:5060> [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Header 6 [ 41]: Call-ID: .J-eGtdilRFt-UP84twohqJBMpfyvdqV [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Header 7 [ 17]: CSeq: 2156 INVITE [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Header 8 [ 29]: Route: <sip:110.232.....;lr> [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Header 9 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Header 10 [ 46]: Supported: replaces, 100rel, timer, norefersub [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Header 11 [ 21]: Session-Expires: 1800 [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Header 12 [ 10]: Min-SE: 90 [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Header 13 [ 41]: User-Agent: ABTO Software VoIP SDK/darwin [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Header 14 [162]: Authorization: Digest username="6000", realm="asterisk", nonce="3a325ff1", uri="sip:600@110.232.....", response="0e48b91fb82485e05408da526a8d8eda", algorithm=MD5 [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Header 15 [ 29]: Content-Type: application/sdp [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Header 16 [ 21]: Content-Length: 292 [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Header 17 [ 0]: [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Body 0 [ 3]: v=0 [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Body 1 [ 43]: o=- 3623369868 3623369868 IN IP4 00d5cc04I4 [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Body 2 [ 10]: s=sip_voip [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Body 3 [ 19]: c=IN IP4 00d5cc04I4 [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Body 4 [ 7]: b=AS:30 [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Body 5 [ 5]: t=0 0 [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Body 6 [ 9]: a=X-nat:0 [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Body 7 [ 26]: m=audio 5068 RTP/AVP 3 101 [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Body 8 [ 19]: c=IN IP4 00d5cc04I4 [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Body 9 [ 12]: b=TIAS:13200 [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Body 10 [ 32]: a=rtcp:5069 IN IP4 192.168.0.100 [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Body 11 [ 10]: a=sendrecv [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Body 12 [ 19]: a=rtpmap:3 GSM/8000 [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Body 13 [ 33]: a=rtpmap:101 telephone-event/8000 [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Body 14 [ 15]: a=fmtp:101 0-15 [Oct 27 10:37:47] VERBOSE[27635] chan_sip.c: --- (17 headers 15 lines) --- [Oct 27 10:37:47] DEBUG[27635][C-00000003] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Oct 27 10:37:47] DEBUG[27635][C-00000003] netsock2.c: Splitting '00d5cc24I4:5060' into... [Oct 27 10:37:47] DEBUG[27635][C-00000003] netsock2.c: ...host '00d5cc24I4' and port '5060'. [Oct 27 10:37:47] ERROR[27635][C-00000003] netsock2.c: getaddrinfo("00d5cc24I4", "5060", ...): Name or service not known [Oct 27 10:37:47] WARNING[27635][C-00000003] chan_sip.c: Could not resolve socket address for '00d5cc24I4:5060' [Oct 27 10:37:47] DEBUG[27635][C-00000003] chan_sip.c: NAT detected for (null) / 114.79.13.40 [Oct 27 10:37:47] VERBOSE[27635][C-00000003] chan_sip.c: Sending to 114.79.13.40:7430 (NAT) [Oct 27 10:37:47] DEBUG[27635][C-00000003] chan_sip.c: Initializing initreq for method INVITE - callid .J-eGtdilRFt-UP84twohqJBMpfyvdqV [Oct 27 10:37:47] VERBOSE[27635][C-00000003] chan_sip.c: Using INVITE request as basis request - .J-eGtdilRFt-UP84twohqJBMpfyvdqV [Oct 27 10:37:47] DEBUG[27635][C-00000003] netsock2.c: Splitting '110.232.....' into... [Oct 27 10:37:47] DEBUG[27635][C-00000003] netsock2.c: ...host '110.232.....' and port ''. [Oct 27 10:37:47] VERBOSE[27635][C-00000003] chan_sip.c: Found peer '6000' for '6000' from 114.79.13.40:7430 [Oct 27 10:37:47] DEBUG[27635][C-00000003] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xb6b0e28c' [Oct 27 10:37:47] DEBUG[27635][C-00000003] res_rtp_asterisk.c: Allocated port 18110 for RTP instance '0xb6b0e28c' [Oct 27 10:37:47] DEBUG[27635][C-00000003] rtp_engine.c: RTP instance '0xb6b0e28c' is setup and ready to go [Oct 27 10:37:47] DEBUG[27635][C-00000003] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xb6b0e28c' [Oct 27 10:37:47] VERBOSE[27635][C-00000003] netsock2.c: == Using SIP RTP CoS mark 5 [Oct 27 10:37:47] DEBUG[27635][C-00000003] chan_sip.c: Setting NAT on RTP to On [Oct 27 10:37:47] DEBUG[27635][C-00000003] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Oct 27 10:37:47] DEBUG[27635][C-00000003] chan_sip.c: Processing session-level SDP o=- 3623369868 3623369868 IN IP4 00d5cc04I4... OK. [Oct 27 10:37:47] DEBUG[27635][C-00000003] chan_sip.c: Processing session-level SDP s=sip_voip... UNSUPPORTED OR FAILED. [Oct 27 10:37:47] DEBUG[27635][C-00000003] netsock2.c: Splitting '00d5cc04I4' into... [Oct 27 10:37:47] DEBUG[27635][C-00000003] netsock2.c: ...host '00d5cc04I4' and port ''. [Oct 27 10:37:47] ERROR[27635][C-00000003] netsock2.c: getaddrinfo("00d5cc04I4", "(null)", ...): Name or service not known [Oct 27 10:37:47] WARNING[27635][C-00000003] chan_sip.c: Unable to lookup RTP Audio host in c= line, 'IN IP4 00d5cc04I4' [Oct 27 10:37:47] DEBUG[27635][C-00000003] chan_sip.c: Processing session-level SDP c=IN IP4 00d5cc04I4... UNSUPPORTED OR FAILED. [Oct 27 10:37:47] DEBUG[27635][C-00000003] chan_sip.c: Processing session-level SDP b=AS:30... UNSUPPORTED OR FAILED. [Oct 27 10:37:47] DEBUG[27635][C-00000003] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Oct 27 10:37:47] DEBUG[27635][C-00000003] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Oct 27 10:37:47] VERBOSE[27635][C-00000003] chan_sip.c: Found RTP audio format 3 [Oct 27 10:37:47] DEBUG[27635][C-00000003] rtp_engine.c: Setting payload 3 based on m type on 0xb4d1fb78 [Oct 27 10:37:47] VERBOSE[27635][C-00000003] chan_sip.c: Found RTP audio format 101 [Oct 27 10:37:47] DEBUG[27635][C-00000003] rtp_engine.c: Setting payload 101 based on m type on 0xb4d1fb78 [Oct 27 10:37:47] DEBUG[27635][C-00000003] netsock2.c: Splitting '00d5cc04I4' into... [Oct 27 10:37:47] DEBUG[27635][C-00000003] netsock2.c: ...host '00d5cc04I4' and port ''. [Oct 27 10:37:47] ERROR[27635][C-00000003] netsock2.c: getaddrinfo("00d5cc04I4", "(null)", ...): Name or service not known [Oct 27 10:37:47] WARNING[27635][C-00000003] chan_sip.c: Unable to lookup RTP Audio host in c= line, 'IN IP4 00d5cc04I4' [Oct 27 10:37:47] DEBUG[27635][C-00000003] chan_sip.c: Processing media-level (audio) SDP c=IN IP4 00d5cc04I4... UNSUPPORTED OR FAILED. [Oct 27 10:37:47] DEBUG[27635][C-00000003] chan_sip.c: Processing media-level (audio) SDP b=TIAS:13200... UNSUPPORTED OR FAILED. [Oct 27 10:37:47] DEBUG[27635][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtcp:5069 IN IP4 192.168.0.100... UNSUPPORTED OR FAILED. [Oct 27 10:37:47] DEBUG[27635][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 27 10:37:47] VERBOSE[27635][C-00000003] chan_sip.c: Found audio description format GSM for ID 3 [Oct 27 10:37:47] DEBUG[27635][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:3 GSM/8000... OK. [Oct 27 10:37:47] VERBOSE[27635][C-00000003] chan_sip.c: Found audio description format telephone-event for ID 101 [Oct 27 10:37:47] DEBUG[27635][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Oct 27 10:37:47] DEBUG[27635][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED. [Oct 27 10:37:47] WARNING[27635][C-00000003] chan_sip.c: Insufficient information in SDP (c=)... [Oct 27 10:37:47] VERBOSE[27635][C-00000003] chan_sip.c: [Oct 27 10:37:47] DEBUG[27635][C-00000003] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #82 [Oct 27 10:37:47] DEBUG[27635][C-00000003] chan_sip.c: Trying to put 'SIP/2.0 488' onto UDP socket destined for 114.79.13.40:7430 [Oct 27 10:37:47] VERBOSE[27635][C-00000003] chan_sip.c: Scheduling destruction of SIP dialog '.J-eGtdilRFt-UP84twohqJBMpfyvdqV' in 32000 ms (Method: INVITE) [Oct 27 10:37:47] DEBUG[27635][C-00000003] chan_sip.c: No compatible codecs for this SIP call. [Oct 27 10:37:47] DEBUG[27635][C-00000003] chan_sip.c: SIP message could not be handled, bad request: .J-eGtdilRFt-UP84twohqJBMpfyvdqV [Oct 27 10:37:47] VERBOSE[27635] chan_sip.c: [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Header 0 [ 33]: ACK sip:600@110.232..... SIP/2.0 [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Header 1 [ 87]: Via: SIP/2.0/UDP 00d5cc24I4:5060;rport;branch=z9hG4bKPjsNgKfpJLIpM1ooTVO5Em-VLKzZCZP18R [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Header 3 [ 74]: From: "6000" <sip:6000@110.232.....>;tag=UpwbPuuTzcHI-704U4FhuJ4ZZOPHub9R [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Header 4 [ 40]: To: sip:600@110.232.....;tag=as2de937a9 [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Header 5 [ 41]: Call-ID: .J-eGtdilRFt-UP84twohqJBMpfyvdqV [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Header 6 [ 14]: CSeq: 2156 ACK [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Header 7 [ 29]: Route: <sip:110.232.....;lr> [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Header 8 [ 18]: Content-Length: 0 [Oct 27 10:37:47] DEBUG[27635] chan_sip.c: Header 9 [ 0]: [Oct 27 10:37:47] VERBOSE[27635] chan_sip.c: --- (9 headers 0 lines) --- [Oct 27 10:37:47] DEBUG[27635][C-00000003] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Oct 27 10:37:47] DEBUG[27635][C-00000003] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #82 [Oct 27 10:37:47] DEBUG[27635][C-00000003] chan_sip.c: Stopping retransmission on '.J-eGtdilRFt-UP84twohqJBMpfyvdqV' of Response 2156: Match Found [/quote]

[SUCCESS]

[quote]SIP Header :

[code]<— SIP read from UDP:180.252.202.202:1024 —>
INVITE sip:6001@110.232…:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.114:5060;rport;branch=z9hG4bKPjSbA0DKY3VLsrz8nXFbMyN-pBw02Jjege
Max-Forwards: 70
From: “6000” sip:6000@110.232.....:5060;tag=fJ7JnXa8-sq5R968sxeuoxw7T–EqQ0.
To: sip:6001@110.232…:5060
Contact: “6000” sip:6000@192.168.0.114:5060
Call-ID: RdDGnn1rcWXtkcpbtiaW12O5sCCDnIuD
CSeq: 5068 INVITE
Route: sip:110.232.....;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: ABTO Software VoIP SDK/darwin
Authorization: Digest username=“6000”, realm=“asterisk”, nonce=“2a5e92c7”, uri=“sip:6001@110.232…”, response=“6f91d668634bcf1c69e0e4e6afe8a34f”, algorithm=MD5
Content-Type: application/sdp
Content-Length: 305

v=0
o=- 3623371782 3623371782 IN IP4 180.252.202.202
s=sip_voip
c=IN IP4 180.252.202.202
b=AS:30
t=0 0
a=X-nat:0
m=audio 5066 RTP/AVP 3 101
c=IN IP4 192.168.0.114
b=TIAS:13200
a=rtcp:5067 IN IP4 192.168.0.114
a=sendrecv
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15[/code]
/var/log/asterisk/full :

[code][Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 0 [ 42]: INVITE sip:6001@110.232…:5060 SIP/2.0
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 192.168.0.114:5060;rport;branch=z9hG4bKPjSbA0DKY3VLsrz8nXFbMyN-pBw02Jjege
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 3 [ 79]: From: “6000” sip:6000@110.232.....:5060;tag=fJ7JnXa8-sq5R968sxeuoxw7T–EqQ0.
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 4 [ 31]: To: sip:6001@110.232…:5060
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 5 [ 45]: Contact: “6000” sip:6000@192.168.0.114:5060
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 6 [ 41]: Call-ID: RdDGnn1rcWXtkcpbtiaW12O5sCCDnIuD
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 7 [ 17]: CSeq: 5068 INVITE
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 8 [ 29]: Route: sip:110.232.....;lr
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 9 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 10 [ 46]: Supported: replaces, 100rel, timer, norefersub
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 11 [ 21]: Session-Expires: 1800
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 12 [ 10]: Min-SE: 90
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 13 [ 41]: User-Agent: ABTO Software VoIP SDK/darwin
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 14 [163]: Authorization: Digest username=“6000”, realm=“asterisk”, nonce=“2a5e92c7”, uri=“sip:6001@110.232…”, response=“6f91d668634bcf1c69e0e4e6afe8a34f”, algorithm=MD5
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 15 [ 29]: Content-Type: application/sdp
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 16 [ 21]: Content-Length: 305
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 17 [ 0]:
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Body 0 [ 3]: v=0
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Body 1 [ 48]: o=- 3623371782 3623371782 IN IP4 180.252.202.202
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Body 2 [ 10]: s=sip_voip
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Body 3 [ 24]: c=IN IP4 180.252.202.202
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Body 4 [ 7]: b=AS:30
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Body 5 [ 5]: t=0 0
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Body 6 [ 9]: a=X-nat:0
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Body 7 [ 26]: m=audio 5066 RTP/AVP 3 101
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Body 8 [ 22]: c=IN IP4 192.168.0.114
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Body 9 [ 12]: b=TIAS:13200
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Body 10 [ 32]: a=rtcp:5067 IN IP4 192.168.0.114
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Body 11 [ 10]: a=sendrecv
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Body 12 [ 19]: a=rtpmap:3 GSM/8000
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Body 13 [ 33]: a=rtpmap:101 telephone-event/8000
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Body 14 [ 15]: a=fmtp:101 0-15
[Oct 27 11:09:40] VERBOSE[27635] chan_sip.c: — (17 headers 15 lines) —

[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Oct 27 11:09:40] DEBUG[27635][C-00000007] netsock2.c: Splitting ‘192.168.0.114:5060’ into…
[Oct 27 11:09:40] DEBUG[27635][C-00000007] netsock2.c: …host ‘192.168.0.114’ and port ‘5060’.
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: NAT detected for 192.168.0.114 / 180.252.202.202
[Oct 27 11:09:40] VERBOSE[27635][C-00000007] chan_sip.c: Sending to 180.252.202.202:1024 (NAT)
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: Initializing initreq for method INVITE - callid RdDGnn1rcWXtkcpbtiaW12O5sCCDnIuD
[Oct 27 11:09:40] VERBOSE[27635][C-00000007] chan_sip.c: Using INVITE request as basis request - RdDGnn1rcWXtkcpbtiaW12O5sCCDnIuD
[Oct 27 11:09:40] DEBUG[27635][C-00000007] netsock2.c: Splitting ‘110.232…:5060’ into…
[Oct 27 11:09:40] DEBUG[27635][C-00000007] netsock2.c: …host ‘110.232…’ and port ‘’.
[Oct 27 11:09:40] VERBOSE[27635][C-00000007] chan_sip.c: Found peer ‘6000’ for ‘6000’ from 180.252.202.202:1024
[Oct 27 11:09:40] DEBUG[27635][C-00000007] rtp_engine.c: Using engine ‘asterisk’ for RTP instance ‘0xb6b2bf3c’
[Oct 27 11:09:40] DEBUG[27635][C-00000007] res_rtp_asterisk.c: Allocated port 10358 for RTP instance ‘0xb6b2bf3c’
[Oct 27 11:09:40] DEBUG[27635][C-00000007] rtp_engine.c: RTP instance ‘0xb6b2bf3c’ is setup and ready to go
[Oct 27 11:09:40] DEBUG[27635][C-00000007] res_rtp_asterisk.c: Setup RTCP on RTP instance ‘0xb6b2bf3c’
[Oct 27 11:09:40] VERBOSE[27635][C-00000007] netsock2.c: == Using SIP RTP CoS mark 5
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: Setting NAT on RTP to On
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: Processing session-level SDP v=0… UNSUPPORTED OR FAILED.
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: Processing session-level SDP o=- 3623371782 3623371782 IN IP4 180.252.202.202… OK.
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: Processing session-level SDP s=sip_voip… UNSUPPORTED OR FAILED.
[Oct 27 11:09:40] DEBUG[27635][C-00000007] netsock2.c: Splitting ‘180.252.202.202’ into…
[Oct 27 11:09:40] DEBUG[27635][C-00000007] netsock2.c: …host ‘180.252.202.202’ and port ‘’.
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: Processing session-level SDP c=IN IP4 180.252.202.202… OK.
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: Processing session-level SDP b=AS:30… UNSUPPORTED OR FAILED.
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: Processing session-level SDP t=0 0… UNSUPPORTED OR FAILED.
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: Processing session-level SDP a=X-nat:0… UNSUPPORTED OR FAILED.
[Oct 27 11:09:40] VERBOSE[27635][C-00000007] chan_sip.c: Found RTP audio format 3
[Oct 27 11:09:40] DEBUG[27635][C-00000007] rtp_engine.c: Setting payload 3 based on m type on 0xb4d1fb78
[Oct 27 11:09:40] VERBOSE[27635][C-00000007] chan_sip.c: Found RTP audio format 101
[Oct 27 11:09:40] DEBUG[27635][C-00000007] rtp_engine.c: Setting payload 101 based on m type on 0xb4d1fb78
[Oct 27 11:09:40] DEBUG[27635][C-00000007] netsock2.c: Splitting ‘192.168.0.114’ into…
[Oct 27 11:09:40] DEBUG[27635][C-00000007] netsock2.c: …host ‘192.168.0.114’ and port ‘’.
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: Processing media-level (audio) SDP c=IN IP4 192.168.0.114… OK.
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: Processing media-level (audio) SDP b=TIAS:13200… UNSUPPORTED OR FAILED.
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: Processing media-level (audio) SDP a=rtcp:5067 IN IP4 192.168.0.114… UNSUPPORTED OR FAILED.
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: Processing media-level (audio) SDP a=sendrecv… OK.
[Oct 27 11:09:40] VERBOSE[27635][C-00000007] chan_sip.c: Found audio description format GSM for ID 3
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:3 GSM/8000… OK.
[Oct 27 11:09:40] VERBOSE[27635][C-00000007] chan_sip.c: Found audio description format telephone-event for ID 101
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000… OK.
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15… UNSUPPORTED OR FAILED.
[Oct 27 11:09:40] VERBOSE[27635][C-00000007] chan_sip.c: Capabilities: us - (gsm), peer - audio=(gsm)/video=(nothing)/text=(nothing), combined - (gsm)
[Oct 27 11:09:40] VERBOSE[27635][C-00000007] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Oct 27 11:09:40] DEBUG[27635][C-00000007] res_rtp_asterisk.c: Setting RTCP address on RTP instance ‘0xb6b2bf3c’
[Oct 27 11:09:40] VERBOSE[27635][C-00000007] chan_sip.c: Peer audio RTP is at port 192.168.0.114:5066
[Oct 27 11:09:40] DEBUG[27635][C-00000007] rtp_engine.c: Copying payload 3 from 0xb4d1fb78 to 0xb6b2c0e8
[Oct 27 11:09:40] DEBUG[27635][C-00000007] rtp_engine.c: Copying payload 101 from 0xb4d1fb78 to 0xb6b2c0e8
[Oct 27 11:09:40] DEBUG[27635][C-00000007] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance ‘0xb6b2bf3c’
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: We’re settling with these formats: (gsm)
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: Checking SIP call limits for device 6000
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: Updating call counter for incoming call
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: Call from peer ‘6000’ is 1 out of 2147483647
[Oct 27 11:09:40] DEBUG[27635][C-00000007] netsock2.c: Splitting ‘110.232…:5060’ into…
[Oct 27 11:09:40] DEBUG[27635][C-00000007] netsock2.c: …host ‘110.232…’ and port ‘’.
[Oct 27 11:09:40] DEBUG[27635][C-00000007] netsock2.c: Splitting ‘110.232…:5060’ into…
[Oct 27 11:09:40] DEBUG[27635][C-00000007] netsock2.c: …host ‘110.232…’ and port ‘’.
[Oct 27 11:09:40] VERBOSE[27635][C-00000007] chan_sip.c: Looking for 6001 in DLPN_DialPlan1 (domain 110.232…)
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: Incoming INVITE with ‘timer’ option supported
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: INVITE also has “Session-Expires” header.
[Oct 27 11:09:40] DEBUG[27614] devicestate.c: No provider found, checking channel drivers for SIP - 6000
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: Session-Expires: 1800
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: INVITE also has “Min-SE” header.
[Oct 27 11:09:40] DEBUG[27614] chan_sip.c: Checking device state for peer 6000
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: Received Min-SE: 90
[Oct 27 11:09:40] DEBUG[27614] devicestate.c: Changing state for SIP/6000 - state 2 (In use)
[Oct 27 11:09:40] DEBUG[27614] devicestate.c: device ‘SIP/6000’ state ‘2’
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: *** Our native formats are (gsm)
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: *** Joint capabilities are (gsm)
[Oct 27 11:09:40] DEBUG[27647] app_queue.c: Device ‘SIP/6000’ changed to state ‘2’ (In use) but we don’t care because they’re not a member of any queue.
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: *** Our capabilities are (gsm)
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: *** AST_CODEC_CHOOSE formats are gsm
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: This channel will not be able to handle video.
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: build_route: Contact hop: “6000” sip:6000@192.168.0.114:5060
[Oct 27 11:09:40] DEBUG[27616] app_queue.c: Extension ‘6000@default’ changed to state ‘2’ (In use) but we don’t care because they’re not a member of any queue.
[Oct 27 11:09:40] VERBOSE[27635][C-00000007] chan_sip.c: list_route: hop: sip:6000@192.168.0.114:5060
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: SIP/6000-00000004: New call is still down… Trying…
[Oct 27 11:09:40] VERBOSE[27635][C-00000007] chan_sip.c:
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: Trying to put ‘SIP/2.0 100’ onto UDP socket destined for 180.252.202.202:1024
[Oct 27 11:09:40] DEBUG[27614] devicestate.c: No provider found, checking channel drivers for SIP - 6000
[Oct 27 11:09:40] DEBUG[27614] chan_sip.c: Checking device state for peer 6000
[Oct 27 11:09:40] DEBUG[27614] devicestate.c: Changing state for SIP/6000 - state 2 (In use)
[Oct 27 11:09:40] DEBUG[27614] devicestate.c: device ‘SIP/6000’ state ‘2’
[Oct 27 11:09:40] DEBUG[28265][C-00000007] pbx.c: Result of ‘HINT’ is ‘SIP/6001’
[Oct 27 11:09:40] DEBUG[28265][C-00000007] pbx.c: Launching ‘Dial’
[Oct 27 11:09:40] VERBOSE[28265][C-00000007] pbx.c: – Executing [6001@DLPN_DialPlan1:1] Dial(“SIP/6000-00000004”, “SIP/6001”) in new stack
[Oct 27 11:09:40] DEBUG[27647] app_queue.c: Device ‘SIP/6000’ changed to state ‘2’ (In use) but we don’t care because they’re not a member of any queue.
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: Asked to create a SIP channel with formats: (gsm)
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: Allocating new SIP dialog for 46ffb97a301955f24fe0887236890c1f@192.168.1.2:5060 - INVITE (No RTP)
[Oct 27 11:09:40] DEBUG[28265][C-00000007] rtp_engine.c: Using engine ‘asterisk’ for RTP instance ‘0xb700429c’
[Oct 27 11:09:40] DEBUG[28265][C-00000007] res_rtp_asterisk.c: Allocated port 11660 for RTP instance ‘0xb700429c’
[Oct 27 11:09:40] DEBUG[28265][C-00000007] rtp_engine.c: RTP instance ‘0xb700429c’ is setup and ready to go
[Oct 27 11:09:40] DEBUG[28265][C-00000007] res_rtp_asterisk.c: Setup RTCP on RTP instance ‘0xb700429c’
[Oct 27 11:09:40] VERBOSE[28265][C-00000007] netsock2.c: == Using SIP RTP CoS mark 5
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: Setting NAT on RTP to On
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: OBPROXY: Not applying OBproxy to this call
[Oct 27 11:09:40] DEBUG[28265][C-00000007] acl.c: For destination ‘180.252.202.202’, our source address is ‘110.232…’.
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 110.232…:5060
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: Setting NAT on RTP to On
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: SIP call-id changed from ‘46ffb97a301955f24fe0887236890c1f@192.168.1.2:5060’ to ‘0de99ff86babcb0553c229074bed3ea3@110.232…:5060’
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: *** Our native formats are (gsm)
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: *** Joint capabilities are (gsm)
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: *** Our capabilities are (gsm)
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: *** AST_CODEC_CHOOSE formats are gsm
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: *** Our preferred formats from the incoming channel are (gsm)
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: This channel will not be able to handle video.
[Oct 27 11:09:40] DEBUG[28265][C-00000007] channel_internal_api.c: Channel Call ID changing from [C-00000007] to [C-00000007]
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: Outgoing Call for 6001
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: Updating call counter for outgoing call
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: Call to peer ‘6001’ is 1 out of 2147483647
[Oct 27 11:09:40] DEBUG[27614] devicestate.c: No provider found, checking channel drivers for SIP - 6001
[Oct 27 11:09:40] DEBUG[27614] chan_sip.c: Checking device state for peer 6001
[Oct 27 11:09:40] DEBUG[27614] devicestate.c: Changing state for SIP/6001 - state 6 (Ringing)
[Oct 27 11:09:40] DEBUG[27614] devicestate.c: device ‘SIP/6001’ state ‘6’
[Oct 27 11:09:40] DEBUG[27647] app_queue.c: Device ‘SIP/6001’ changed to state ‘6’ (Ringing) but we don’t care because they’re not a member of any queue.
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: ** Our capability: (gsm) Video flag: False Text flag: False
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: ** Our prefcodec: (gsm)
[Oct 27 11:09:40] DEBUG[27616] app_queue.c: Extension ‘6001@default’ changed to state ‘6’ (Ringing) but we don’t care because they’re not a member of any queue.
[Oct 27 11:09:40] VERBOSE[28265][C-00000007] chan_sip.c: Audio is at 11660
[Oct 27 11:09:40] VERBOSE[28265][C-00000007] chan_sip.c: Adding codec 100002 (gsm) to SDP
[Oct 27 11:09:40] VERBOSE[28265][C-00000007] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: – Done with adding codecs to SDP
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: Done building SDP. Settling with this capability: (gsm)
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: Initializing initreq for method INVITE - callid 0de99ff86babcb0553c229074bed3ea3@110.232…:5060
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: Header 0 [ 42]: INVITE sip:6001@192.168.0.113:5060 SIP/2.0
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 110.232…:5060;branch=z9hG4bK558277b8;rport
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: Header 3 [ 52]: From: “6000” sip:6000@110.232.....;tag=as44d2ab56
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: Header 4 [ 33]: To: sip:6001@192.168.0.113:5060
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: Header 5 [ 38]: Contact: sip:6000@110.232.....:5060
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: Header 6 [ 60]: Call-ID: 0de99ff86babcb0553c229074bed3ea3@110.232…:5060
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: Header 8 [ 24]: User-Agent: Asterisk PBX
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: Header 9 [ 35]: Date: Mon, 27 Oct 2014 04:09:40 GMT
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp
[Oct 27 11:09:40] VERBOSE[28265][C-00000007] chan_sip.c: Reliably Transmitting (NAT) to 180.252.202.202:5060:
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #179
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: Trying to put ‘INVITE sip:’ onto UDP socket destined for 180.252.202.202:5060
[Oct 27 11:09:40] VERBOSE[28265][C-00000007] app_dial.c: – Called SIP/6001
[Oct 27 11:09:40] WARNING[27635] chan_sip.c: Retransmission timeout reached on transmission 5aea57d41c225b269921590636bab33b for seqno 1 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Setting SIP_ALREADYGONE on dialog 5aea57d41c225b269921590636bab33b
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Auto destroying SIP dialog ‘5aea57d41c225b269921590636bab33b’
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Destroying SIP dialog 5aea57d41c225b269921590636bab33b
[Oct 27 11:09:40] VERBOSE[27635] chan_sip.c: Really destroying SIP dialog ‘5aea57d41c225b269921590636bab33b’ Method: INVITE
[Oct 27 11:09:40] DEBUG[27635] rtp_engine.c: Destroyed RTP instance ‘0xb6b24744’
[Oct 27 11:09:40] VERBOSE[27635] chan_sip.c:
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 110.232…:5060;rport=5060;received=110.232…;branch=z9hG4bK558277b8
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 2 [ 60]: Call-ID: 0de99ff86babcb0553c229074bed3ea3@110.232…:5060
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 3 [ 57]: From: “6000” sip:6000@110.232.....:5060;tag=as44d2ab56
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 4 [ 28]: To: sip:6001@192.168.0.113
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 6 [ 18]: Content-Length: 0
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 7 [ 0]:
[Oct 27 11:09:40] VERBOSE[27635] chan_sip.c: — (7 headers 0 lines) —
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: *** SIP TIMER: Cancelling retransmission #179 - INVITE (got response)
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on ‘0de99ff86babcb0553c229074bed3ea3@110.232…:5060’ Request 102: Found
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: SIP response 100 to standard invite
[Oct 27 11:09:40] VERBOSE[27635] chan_sip.c:
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 110.232…:5060;rport=5060;received=110.232…;branch=z9hG4bK558277b8
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 2 [ 60]: Call-ID: 0de99ff86babcb0553c229074bed3ea3@110.232…:5060
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 3 [ 57]: From: “6000” sip:6000@110.232.....:5060;tag=as44d2ab56
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 4 [ 65]: To: sip:6001@192.168.0.113;tag=Xlek-BFg5q90zCXnx-0smXegmtEVy4OC
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 6 [ 45]: Contact: “6001” sip:6001@192.168.0.113:5060
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 7 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 8 [ 18]: Content-Length: 0
[Oct 27 11:09:40] DEBUG[27635] chan_sip.c: Header 9 [ 0]:
[Oct 27 11:09:40] VERBOSE[27635] chan_sip.c: — (9 headers 0 lines) —
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on ‘0de99ff86babcb0553c229074bed3ea3@110.232…:5060’ Request 102: Found
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: SIP response 180 to standard invite
[Oct 27 11:09:40] DEBUG[27635][C-00000007] chan_sip.c: build_route: Contact hop: “6001” sip:6001@192.168.0.113:5060
[Oct 27 11:09:40] VERBOSE[27635][C-00000007] chan_sip.c: list_route: hop: sip:6001@192.168.0.113:5060
[Oct 27 11:09:40] DEBUG[27614] devicestate.c: No provider found, checking channel drivers for SIP - 6001
[Oct 27 11:09:40] DEBUG[27614] chan_sip.c: Checking device state for peer 6001
[Oct 27 11:09:40] DEBUG[27614] devicestate.c: Changing state for SIP/6001 - state 6 (Ringing)
[Oct 27 11:09:40] VERBOSE[28265][C-00000007] app_dial.c: – SIP/6001-00000005 is ringing
[Oct 27 11:09:40] DEBUG[27614] devicestate.c: device ‘SIP/6001’ state ‘6’
[Oct 27 11:09:40] DEBUG[28265][C-00000007] rtp_engine.c: Setting early bridge SDP of ‘SIP/6000-00000004’ with that of ‘SIP/6001-00000005’
[Oct 27 11:09:40] DEBUG[27647] app_queue.c: Device ‘SIP/6001’ changed to state ‘6’ (Ringing) but we don’t care because they’re not a member of any queue.
[Oct 27 11:09:40] VERBOSE[28265][C-00000007] chan_sip.c:
[Oct 27 11:09:40] DEBUG[28265][C-00000007] chan_sip.c: Trying to put ‘SIP/2.0 180’ onto UDP socket destined for 180.252.202.202:1024
[Oct 27 11:09:42] VERBOSE[27635] chan_sip.c:
[Oct 27 11:09:42] DEBUG[27635] chan_sip.c: Header 0 [ 19]: SIP/2.0 603 Decline
[Oct 27 11:09:42] DEBUG[27635] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 110.232…:5060;rport=5060;received=110.232…;branch=z9hG4bK558277b8
[Oct 27 11:09:42] DEBUG[27635] chan_sip.c: Header 2 [ 60]: Call-ID: 0de99ff86babcb0553c229074bed3ea3@110.232…:5060
[Oct 27 11:09:42] DEBUG[27635] chan_sip.c: Header 3 [ 57]: From: “6000” sip:6000@110.232.....:5060;tag=as44d2ab56
[Oct 27 11:09:42] DEBUG[27635] chan_sip.c: Header 4 [ 65]: To: sip:6001@192.168.0.113;tag=Xlek-BFg5q90zCXnx-0smXegmtEVy4OC
[Oct 27 11:09:42] DEBUG[27635] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE
[Oct 27 11:09:42] DEBUG[27635] chan_sip.c: Header 6 [ 90]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
[Oct 27 11:09:42] DEBUG[27635] chan_sip.c: Header 7 [ 18]: Content-Length: 0
[Oct 27 11:09:42] DEBUG[27635] chan_sip.c: Header 8 [ 0]:
[Oct 27 11:09:42] VERBOSE[27635] chan_sip.c: — (8 headers 0 lines) —
[Oct 27 11:09:42] DEBUG[27635][C-00000007] chan_sip.c: Acked pending invite 102
[Oct 27 11:09:42] DEBUG[27635][C-00000007] chan_sip.c: Stopping retransmission on ‘0de99ff86babcb0553c229074bed3ea3@110.232…:5060’ of Request 102: Match Found
[Oct 27 11:09:42] VERBOSE[27635][C-00000007] chan_sip.c: – Got SIP response 603 “Decline” back from 180.252.202.202:5060
[Oct 27 11:09:42] DEBUG[27635][C-00000007] res_rtp_asterisk.c: Setting RTCP address on RTP instance ‘0xb700429c’
[Oct 27 11:09:42] DEBUG[27635][C-00000007] chan_sip.c: Strict routing enforced for session 0de99ff86babcb0553c229074bed3ea3@110.232…:5060
[Oct 27 11:09:42] VERBOSE[27635][C-00000007] chan_sip.c: set_destination: Parsing sip:6001@192.168.0.113:5060 for address/port to send to
[Oct 27 11:09:42] DEBUG[27635][C-00000007] netsock2.c: Splitting ‘192.168.0.113:5060’ into…
[Oct 27 11:09:42] DEBUG[27635][C-00000007] netsock2.c: …host ‘192.168.0.113’ and port ‘5060’.
[Oct 27 11:09:42] VERBOSE[27635][C-00000007] chan_sip.c: set_destination: set destination to 192.168.0.113:5060
[Oct 27 11:09:42] VERBOSE[27635][C-00000007] chan_sip.c: Transmitting (NAT) to 180.252.202.202:5060:

[Oct 27 11:09:42] DEBUG[27635][C-00000007] chan_sip.c: Trying to put ‘ACK sip:600’ onto UDP socket destined for 180.252.202.202:5060
[Oct 27 11:09:42] DEBUG[27635][C-00000007] chan_sip.c: Setting SIP_ALREADYGONE on dialog 0de99ff86babcb0553c229074bed3ea3@110.232…:5060
[Oct 27 11:09:42] VERBOSE[28265][C-00000007] app_dial.c: – SIP/6001-00000005 is busy
[Oct 27 11:09:42] DEBUG[28265][C-00000007] channel.c: Hanging up channel ‘SIP/6001-00000005’
[Oct 27 11:09:42] DEBUG[28265][C-00000007] chan_sip.c: Hangup call SIP/6001-00000005, SIP callid 0de99ff86babcb0553c229074bed3ea3@110.232…:5060
[Oct 27 11:09:42] DEBUG[28265][C-00000007] chan_sip.c: update_call_counter(6001) - decrement call limit counter on hangup
[Oct 27 11:09:42] DEBUG[28265][C-00000007] chan_sip.c: Updating call counter for outgoing call
[Oct 27 11:09:42] DEBUG[28265][C-00000007] chan_sip.c: Call to peer ‘6001’ removed from call limit 2147483647
[Oct 27 11:09:42] DEBUG[28265][C-00000007] res_rtp_asterisk.c: Setting RTCP address on RTP instance ‘0xb700429c’
[Oct 27 11:09:42] DEBUG[27614] devicestate.c: No provider found, checking channel drivers for SIP - 6001
[Oct 27 11:09:42] DEBUG[27614] chan_sip.c: Checking device state for peer 6001
[Oct 27 11:09:42] DEBUG[27614] devicestate.c: Changing state for SIP/6001 - state 1 (Not in use)
[Oct 27 11:09:42] DEBUG[27614] devicestate.c: device ‘SIP/6001’ state ‘1’
[Oct 27 11:09:42] VERBOSE[28265][C-00000007] app_dial.c: == Everyone is busy/congested at this time (1:1/0/0)
[Oct 27 11:09:42] DEBUG[27614] devicestate.c: No provider found, checking channel drivers for SIP - 6001
[Oct 27 11:09:42] DEBUG[27614] chan_sip.c: Checking device state for peer 6001
[Oct 27 11:09:42] DEBUG[27614] devicestate.c: Changing state for SIP/6001 - state 1 (Not in use)
[Oct 27 11:09:42] DEBUG[27614] devicestate.c: device ‘SIP/6001’ state ‘1’
[Oct 27 11:09:42] DEBUG[28265][C-00000007] app_dial.c: Exiting with DIALSTATUS=BUSY.
[Oct 27 11:09:42] DEBUG[27647] app_queue.c: Device ‘SIP/6001’ changed to state ‘1’ (Not in use) but we don’t care because they’re not a member of any queue.
[Oct 27 11:09:42] DEBUG[27647] app_queue.c: Device ‘SIP/6001’ changed to state ‘1’ (Not in use) but we don’t care because they’re not a member of any queue.
[Oct 27 11:09:42] VERBOSE[28265][C-00000007] pbx.c: – Auto fallthrough, channel ‘SIP/6000-00000004’ status is ‘BUSY’
[Oct 27 11:09:42] DEBUG[27616] app_queue.c: Extension ‘6001@default’ changed to state ‘1’ (Not in use) but we don’t care because they’re not a member of any queue.
[Oct 27 11:09:42] VERBOSE[28265][C-00000007] chan_sip.c:

[Oct 27 11:09:42] DEBUG[28265][C-00000007] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #182
[Oct 27 11:09:42] DEBUG[28265][C-00000007] chan_sip.c: Trying to put ‘SIP/2.0 486’ onto UDP socket destined for 180.252.202.202:1024
[Oct 27 11:09:42] DEBUG[28265][C-00000007] chan_sip.c: Setting SIP_ALREADYGONE on dialog RdDGnn1rcWXtkcpbtiaW12O5sCCDnIuD
[Oct 27 11:09:42] DEBUG[28265][C-00000007] channel.c: Soft-Hanging up channel ‘SIP/6000-00000004’
[Oct 27 11:09:42] DEBUG[27614] devicestate.c: No provider found, checking channel drivers for SIP - 6000
[Oct 27 11:09:42] DEBUG[27614] chan_sip.c: Checking device state for peer 6000
[Oct 27 11:09:42] DEBUG[27614] devicestate.c: Changing state for SIP/6000 - state 2 (In use)
[Oct 27 11:09:42] DEBUG[28265][C-00000007] channel.c: Soft-Hanging up channel ‘SIP/6000-00000004’
[Oct 27 11:09:42] DEBUG[27614] devicestate.c: device ‘SIP/6000’ state ‘2’
[Oct 27 11:09:42] DEBUG[27635] chan_sip.c: Destroying SIP dialog 0de99ff86babcb0553c229074bed3ea3@110.232…:5060
[Oct 27 11:09:42] VERBOSE[27635] chan_sip.c: Really destroying SIP dialog ‘0de99ff86babcb0553c229074bed3ea3@110.232…:5060’ Method: INVITE
[Oct 27 11:09:42] DEBUG[27635] rtp_engine.c: Destroyed RTP instance ‘0xb700429c’
[Oct 27 11:09:42] DEBUG[28265][C-00000007] channel.c: Hanging up channel ‘SIP/6000-00000004’
[Oct 27 11:09:42] DEBUG[27647] app_queue.c: Device ‘SIP/6000’ changed to state ‘2’ (In use) but we don’t care because they’re not a member of any queue.
[Oct 27 11:09:42] DEBUG[28265][C-00000007] chan_sip.c: Hangup call SIP/6000-00000004, SIP callid RdDGnn1rcWXtkcpbtiaW12O5sCCDnIuD
[Oct 27 11:09:42] DEBUG[28265][C-00000007] chan_sip.c: update_call_counter(6000) - decrement call limit counter on hangup
[Oct 27 11:09:42] DEBUG[28265][C-00000007] chan_sip.c: Updating call counter for incoming call
[Oct 27 11:09:42] DEBUG[28265][C-00000007] chan_sip.c: Call from peer ‘6000’ removed from call limit 2147483647
[Oct 27 11:09:42] DEBUG[28265][C-00000007] res_rtp_asterisk.c: Setting RTCP address on RTP instance ‘0xb6b2bf3c’
[Oct 27 11:09:42] DEBUG[27614] devicestate.c: No provider found, checking channel drivers for SIP - 6000
[Oct 27 11:09:42] DEBUG[27614] chan_sip.c: Checking device state for peer 6000
[Oct 27 11:09:42] DEBUG[27614] devicestate.c: Changing state for SIP/6000 - state 1 (Not in use)
[Oct 27 11:09:42] DEBUG[27614] devicestate.c: device ‘SIP/6000’ state ‘1’
[Oct 27 11:09:42] DEBUG[27647] app_queue.c: Device ‘SIP/6000’ changed to state ‘1’ (Not in use) but we don’t care because they’re not a member of any queue.
[Oct 27 11:09:42] DEBUG[27614] devicestate.c: No provider found, checking channel drivers for SIP - 6000
[Oct 27 11:09:42] DEBUG[27614] chan_sip.c: Checking device state for peer 6000
[Oct 27 11:09:42] DEBUG[27616] app_queue.c: Extension ‘6000@default’ changed to state ‘1’ (Not in use) but we don’t care because they’re not a member of any queue.
[Oct 27 11:09:42] DEBUG[27614] devicestate.c: Changing state for SIP/6000 - state 1 (Not in use)
[Oct 27 11:09:42] DEBUG[27614] devicestate.c: device ‘SIP/6000’ state ‘1’
[Oct 27 11:09:42] DEBUG[27647] app_queue.c: Device ‘SIP/6000’ changed to state ‘1’ (Not in use) but we don’t care because they’re not a member of any queue.
[Oct 27 11:09:42] VERBOSE[27635] chan_sip.c:

[Oct 27 11:09:42] DEBUG[27635] chan_sip.c: Header 0 [ 39]: ACK sip:6001@110.232…:5060 SIP/2.0
[Oct 27 11:09:42] DEBUG[27635] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 192.168.0.114:5060;rport;branch=z9hG4bKPjSbA0DKY3VLsrz8nXFbMyN-pBw02Jjege
[Oct 27 11:09:42] DEBUG[27635] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70
[Oct 27 11:09:42] DEBUG[27635] chan_sip.c: Header 3 [ 79]: From: “6000” sip:6000@110.232.....:5060;tag=fJ7JnXa8-sq5R968sxeuoxw7T–EqQ0.
[Oct 27 11:09:42] DEBUG[27635] chan_sip.c: Header 4 [ 46]: To: sip:6001@110.232…:5060;tag=as3f344903
[Oct 27 11:09:42] DEBUG[27635] chan_sip.c: Header 5 [ 41]: Call-ID: RdDGnn1rcWXtkcpbtiaW12O5sCCDnIuD
[Oct 27 11:09:42] DEBUG[27635] chan_sip.c: Header 6 [ 14]: CSeq: 5068 ACK
[Oct 27 11:09:42] DEBUG[27635] chan_sip.c: Header 7 [ 29]: Route: sip:110.232.....;lr
[Oct 27 11:09:42] DEBUG[27635] chan_sip.c: Header 8 [ 18]: Content-Length: 0
[Oct 27 11:09:42] DEBUG[27635] chan_sip.c: Header 9 [ 0]:
[Oct 27 11:09:42] VERBOSE[27635] chan_sip.c: — (9 headers 0 lines) —
[Oct 27 11:09:42] DEBUG[27635][C-00000007] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Oct 27 11:09:42] DEBUG[27635][C-00000007] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #182
[Oct 27 11:09:42] DEBUG[27635][C-00000007] chan_sip.c: Stopping retransmission on ‘RdDGnn1rcWXtkcpbtiaW12O5sCCDnIuD’ of Response 5068: Match Found
[Oct 27 11:09:42] DEBUG[27635] chan_sip.c: Session timer stopped: -1 - RdDGnn1rcWXtkcpbtiaW12O5sCCDnIuD
[Oct 27 11:09:42] DEBUG[27635] chan_sip.c: Destroying SIP dialog RdDGnn1rcWXtkcpbtiaW12O5sCCDnIuD
[Oct 27 11:09:42] VERBOSE[27635] chan_sip.c: Really destroying SIP dialog ‘RdDGnn1rcWXtkcpbtiaW12O5sCCDnIuD’ Method: ACK
[Oct 27 11:09:42] DEBUG[27635] rtp_engine.c: Destroyed RTP instance ‘0xb6b2bf3c’
[/code][/quote]

8

Back to the DNS problem. Provide valid DNS or other means of resolving the address 00d5cc04I4.