PROGRAMMERPC
27-08-14, 20:17
Kαλησπέρα σας και από μένα,
έχω στήσει έναν asterisk 1.8 με asterisk-gui και έχω δυο προβλήματα.
Tο πρώτο είναι ότι μόλις απαντάω κλίση εισερχόμενη από το trunk τις viva μετά από 6 δευτερόλεπτα κλείνει..
Kαι το δεύτερο είναι ότι από το trunk του ephone δεν μπορώ να λάβω κληση καθόλου.
Βοηθήστε με σας παρακαλω! Δουλεύω με Gui!
Ευχαριστώ.
cli:
Verbosity is at least 3
Core debug is at least 1
[Aug 27 19:58:29] WARNING[10792]: chan_sip.c:25474 handle_request_subscribe: SUBSCRIBE failure: unrecognized format:'application/watcherinfo+xml' pvt: subscribed: 0, stateid: -1, laststate: 0,dialogver: 0, subscribecont: 'default', subscribeuri: ''
== Using SIP RTP CoS mark 5
-- Executing [s@DID_trunk_1:1] Goto("SIP/trunk_viva-00000065", "ringroups-custom-1,s,1") in new stack
-- Goto (ringroups-custom-1,s,1)
-- Executing [s@ringroups-custom-1:1] NoOp("SIP/trunk_viva-00000065", "Loukas") in new stack
-- Executing [s@ringroups-custom-1:2] Dial("SIP/trunk_viva-00000065", "SIP/6000&SIP/6001&SIP/6003&IAX2/6003&IAX2/6001&IAX2/6000,40,i") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/6000
== Using SIP RTP CoS mark 5
-- Called SIP/6001
== Using SIP RTP CoS mark 5
-- Called SIP/6003
[Aug 27 19:59:19] WARNING[21897]: app_dial.c:2341 dial_exec_full: Unable to create channel of type 'IAX2' (cause 20 - Unknown)
[Aug 27 19:59:19] WARNING[21897]: app_dial.c:2341 dial_exec_full: Unable to create channel of type 'IAX2' (cause 20 - Unknown)
[Aug 27 19:59:19] WARNING[21897]: app_dial.c:2341 dial_exec_full: Unable to create channel of type 'IAX2' (cause 20 - Unknown)
-- SIP/6000-00000066 connected line has changed. Saving it until answer for SIP/trunk_viva-00000065
-- SIP/6001-00000067 connected line has changed. Saving it until answer for SIP/trunk_viva-00000065
-- SIP/6003-00000068 connected line has changed. Saving it until answer for SIP/trunk_viva-00000065
-- SIP/6000-00000066 is ringing
-- Got SIP response 486 "Busy Here" back from 83.235.190.161:17183
-- SIP/6003-00000068 is busy
-- SIP/6001-00000067 is ringing
== Using UDPTL CoS mark 5
-- SIP/6000-00000066 connected line has changed. Saving it until answer for SIP/trunk_viva-00000065
-- SIP/6000-00000066 answered SIP/trunk_viva-00000065
-- Locally bridging SIP/trunk_viva-00000065 and SIP/6000-00000066
[Aug 27 19:59:36] WARNING[10792]: chan_sip.c:3736 retrans_pkt: Retransmission timeout reached on transmission 063c00765680cd0a65aa39887a9d79fb@83.235.24.87 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response
[Aug 27 19:59:36] WARNING[10792]: chan_sip.c:3765 retrans_pkt: Hanging up call 063c00765680cd0a65aa39887a9d79fb@83.235.24.87 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
== Spawn extension (ringroups-custom-1, s, 2) exited non-zero on 'SIP/trunk_viva-00000065'
debug:
--- (8 headers 0 lines) ---
Retransmitting #4 (NAT) to 83.235.24.86:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 83.235.24.86;branch=z9hG4bK324a.0512f147.0;received=83.235.24.86;rport=5060
Via: SIP/2.0/UDP 83.235.24.87:5060;received=83.235.24.87;branch=z9hG4bK186862c7;rport=5060
Record-Route: <sip:83.235.24.86:5060;lr=on;ftag=as76df670a;vsf=AAAAAAAAAAAAAAAAAAAATlpYUx1SXDI0Ljg3>
From: "numberwhocallingme" <sip:numberwhocallingme@viva.gr>;tag=as76df670a
To: <sip:myvoipnumber@83.235.24.86:5060>;tag=as647edeb4
Call-ID: 545ee0cf6648ec546724db6963efc93f@83.235.24.87
CSeq: 102 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:s@77.49.80.112:5070>
Content-Type: application/sdp
Content-Length: 317
v=0
o=root 298456689 298456689 IN IP4 77.49.80.112
s=Asterisk PBX 1.8.15-cert7
c=IN IP4 77.49.80.112
t=0 0
m=audio 12790 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:77.49.80.112:60080 --->
<------------->
Retransmitting #5 (NAT) to 83.235.24.86:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 83.235.24.86;branch=z9hG4bK324a.0512f147.0;received=83.235.24.86;rport=5060
Via: SIP/2.0/UDP 83.235.24.87:5060;received=83.235.24.87;branch=z9hG4bK186862c7;rport=5060
Record-Route: <sip:83.235.24.86:5060;lr=on;ftag=as76df670a;vsf=AAAAAAAAAAAAAAAAAAAATlpYUx1SXDI0Ljg3>
From: "numberwhocallingme" <sip:numberwhocallingme@viva.gr>;tag=as76df670a
To: <sip:myvoipnumber@83.235.24.86:5060>;tag=as647edeb4
Call-ID: 545ee0cf6648ec546724db6963efc93f@83.235.24.87
CSeq: 102 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:s@77.49.80.112:5070>
Content-Type: application/sdp
Content-Length: 317
v=0
o=root 298456689 298456689 IN IP4 77.49.80.112
s=Asterisk PBX 1.8.15-cert7
c=IN IP4 77.49.80.112
t=0 0
m=audio 12790 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
Retransmitting #6 (NAT) to 83.235.24.86:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 83.235.24.86;branch=z9hG4bK324a.0512f147.0;received=83.235.24.86;rport=5060
Via: SIP/2.0/UDP 83.235.24.87:5060;received=83.235.24.87;branch=z9hG4bK186862c7;rport=5060
Record-Route: <sip:83.235.24.86:5060;lr=on;ftag=as76df670a;vsf=AAAAAAAAAAAAAAAAAAAATlpYUx1SXDI0Ljg3>
From: "numberwhocallingme" <sip:numberwhocallingme@viva.gr>;tag=as76df670a
To: <sip:myvoipnumber@83.235.24.86:5060>;tag=as647edeb4
Call-ID: 545ee0cf6648ec546724db6963efc93f@83.235.24.87
CSeq: 102 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:s@77.49.80.112:5070>
Content-Type: application/sdp
Content-Length: 317
v=0
o=root 298456689 298456689 IN IP4 77.49.80.112
s=Asterisk PBX 1.8.15-cert7
c=IN IP4 77.49.80.112
t=0 0
m=audio 12790 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
[Aug 27 20:09:06] WARNING[10792]: chan_sip.c:3736 retrans_pkt: Retransmission timeout reached on transmission 545ee0cf6648ec546724db6963efc93f@83.235.24.87 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[Aug 27 20:09:06] WARNING[10792]: chan_sip.c:3765 retrans_pkt: Hanging up call 545ee0cf6648ec546724db6963efc93f@83.235.24.87 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
Scheduling destruction of SIP dialog '0462b63b0c58d2d07a754e844747adcc@77.49.80.112:5070' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:192.168.1.98:5060> for address/port to send to
set_destination: set destination to 192.168.1.98:5060
Reliably Transmitting (NAT) to 77.49.80.112:50266:
BYE sip:192.168.1.98:5060 SIP/2.0
Via: SIP/2.0/UDP 77.49.80.112:5070;branch=z9hG4bK049a0a78;rport
Max-Forwards: 70
From: "numberwhocallingme" <sip:numberwhocallingme@77.49.80.112:5070>;tag=as3a2d634b
To: <sip:6000@192.168.1.98:5060;ob>;tag=2ee58d1c2c404b0d88412e295073c75e
Call-ID: 0462b63b0c58d2d07a754e844747adcc@77.49.80.112:5070
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
== Spawn extension (ringroups-custom-1, s, 2) exited non-zero on 'SIP/trunk_viva-0000006d'
Scheduling destruction of SIP dialog '545ee0cf6648ec546724db6963efc93f@83.235.24.87' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:83.235.24.86:5060;lr=on;ftag=as76df670a;vsf=AAAAAAAAAAAAAAAAAAAATlpYUx1SXDI0Ljg3> for address/port to send to
set_destination: set destination to 83.235.24.86:5060
Reliably Transmitting (NAT) to 83.235.24.86:5060:
BYE sip:numberwhocallingme@83.235.24.87 SIP/2.0
Via: SIP/2.0/UDP 77.49.80.112:5070;branch=z9hG4bK3a662d30;rport
Route: <sip:83.235.24.86:5060;lr=on;ftag=as76df670a;vsf=AAAAAAAAAAAAAAAAAAAATlpYUx1SXDI0Ljg3>
Max-Forwards: 70
From: <sip:myvoipnumber@83.235.24.86:5060>;tag=as647edeb4
To: "numberwhocallingme" <sip:numberwhocallingme@viva.gr>;tag=as76df670a
Call-ID: 545ee0cf6648ec546724db6963efc93f@83.235.24.87
CSeq: 102 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0
---
<--- SIP read from UDP:77.49.80.112:50266 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 77.49.80.112:5070;rport=5070;received=77.49.80.112;branch=z9hG4bK049a0a78
Call-ID: 0462b63b0c58d2d07a754e844747adcc@77.49.80.112:5070
From: "numberwhocallingme" <sip:numberwhocallingme@77.49.80.112>;tag=as3a2d634b
To: <sip:6000@192.168.1.98;ob>;tag=2ee58d1c2c404b0d88412e295073c75e
CSeq: 103 BYE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '0462b63b0c58d2d07a754e844747adcc@77.49.80.112:5070' Method: INVITE
Retransmitting #1 (NAT) to 83.235.24.86:5060:
BYE sip:numberwhocallingme@83.235.24.87 SIP/2.0
Via: SIP/2.0/UDP 77.49.80.112:5070;branch=z9hG4bK3a662d30;rport
Route: <sip:83.235.24.86:5060;lr=on;ftag=as76df670a;vsf=AAAAAAAAAAAAAAAAAAAATlpYUx1SXDI0Ljg3>
Max-Forwards: 70
From: <sip:myvoipnumber@83.235.24.86:5060>;tag=as647edeb4
To: "numberwhocallingme" <sip:numberwhocallingme@viva.gr>;tag=as76df670a
Call-ID: 545ee0cf6648ec546724db6963efc93f@83.235.24.87
CSeq: 102 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0
---
Really destroying SIP dialog '4d133b932e55f0846b4f6a090873b1e1@77.49.80.112:5070' Method: INVITE
<--- SIP read from UDP:83.235.24.86:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 77.49.80.112:5070;received=77.49.80.112;branch=z9hG4bK3a662d30;rport=5070
From: <sip:myvoipnumber@83.235.24.86:5060>;tag=as647edeb4
To: "numberwhocallingme" <sip:numberwhocallingme@viva.gr>;tag=as76df670a
Call-ID: 545ee0cf6648ec546724db6963efc93f@83.235.24.87
CSeq: 102 BYE
Server: m1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '545ee0cf6648ec546724db6963efc93f@83.235.24.87' Method: INVITE
<--- SIP read from UDP:83.235.24.86:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 77.49.80.112:5070;received=77.49.80.112;branch=z9hG4bK3a662d30;rport=5070
From: <sip:myvoipnumber@83.235.24.86:5060>;tag=as647edeb4
To: "numberwhocallingme" <sip:numberwhocallingme@viva.gr>;tag=as76df670a
Call-ID: 545ee0cf6648ec546724db6963efc93f@83.235.24.87
CSeq: 102 BYE
Server: m1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP:79.131.37.170:5060 --->
<------------->
Really destroying SIP dialog '4656cf707dbdb3c67809038c7eec70a3@77.49.80.112:5070' Method: NOTIFY
Reliably Transmitting (NAT) to 83.235.24.86:5060:
OPTIONS sip:voip.viva.gr SIP/2.0
Via: SIP/2.0/UDP 77.49.80.112:5070;branch=z9hG4bK13af4b79;rport
Max-Forwards: 70
From: "asterisk" <sip:myvoipnumber@77.49.80.112:5070>;tag=as48b9a5cf
To: <sip:voip.viva.gr>
Contact: <sip:myvoipnumber@77.49.80.112:5070>
Call-ID: 080e8e32272c9d786374803419bdbc7b@77.49.80.112:5070
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Wed, 27 Aug 2014 17:09:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
Reliably Transmitting (NAT) to 91.217.155.70:5060:
OPTIONS sip:sip.ephone.gr SIP/2.0
Via: SIP/2.0/UDP 77.49.80.112:5070;branch=z9hG4bK1f87070a;rport
Max-Forwards: 70
From: "asterisk" <sip:9438607905@77.49.80.112:5070>;tag=as7b0824eb
To: <sip:sip.ephone.gr>
Contact: <sip:9438607905@77.49.80.112:5070>
Call-ID: 2c39710932a50d33455f783f24350c61@77.49.80.112:5070
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Wed, 27 Aug 2014 17:09:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:83.235.24.86:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 77.49.80.112:5070;branch=z9hG4bK13af4b79;rport=5070
From: "asterisk" <sip:myvoipnumber@77.49.80.112:5070>;tag=as48b9a5cf
To: <sip:voip.viva.gr>;tag=ae428d0fc435bd570aaae338d70dfd1f.f041
Call-ID: 080e8e32272c9d786374803419bdbc7b@77.49.80.112:5070
CSeq: 102 OPTIONS
Server: Viva VoIP
Content-Length: 0
<------------->
έχω στήσει έναν asterisk 1.8 με asterisk-gui και έχω δυο προβλήματα.
Tο πρώτο είναι ότι μόλις απαντάω κλίση εισερχόμενη από το trunk τις viva μετά από 6 δευτερόλεπτα κλείνει..
Kαι το δεύτερο είναι ότι από το trunk του ephone δεν μπορώ να λάβω κληση καθόλου.
Βοηθήστε με σας παρακαλω! Δουλεύω με Gui!
Ευχαριστώ.
cli:
Verbosity is at least 3
Core debug is at least 1
[Aug 27 19:58:29] WARNING[10792]: chan_sip.c:25474 handle_request_subscribe: SUBSCRIBE failure: unrecognized format:'application/watcherinfo+xml' pvt: subscribed: 0, stateid: -1, laststate: 0,dialogver: 0, subscribecont: 'default', subscribeuri: ''
== Using SIP RTP CoS mark 5
-- Executing [s@DID_trunk_1:1] Goto("SIP/trunk_viva-00000065", "ringroups-custom-1,s,1") in new stack
-- Goto (ringroups-custom-1,s,1)
-- Executing [s@ringroups-custom-1:1] NoOp("SIP/trunk_viva-00000065", "Loukas") in new stack
-- Executing [s@ringroups-custom-1:2] Dial("SIP/trunk_viva-00000065", "SIP/6000&SIP/6001&SIP/6003&IAX2/6003&IAX2/6001&IAX2/6000,40,i") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/6000
== Using SIP RTP CoS mark 5
-- Called SIP/6001
== Using SIP RTP CoS mark 5
-- Called SIP/6003
[Aug 27 19:59:19] WARNING[21897]: app_dial.c:2341 dial_exec_full: Unable to create channel of type 'IAX2' (cause 20 - Unknown)
[Aug 27 19:59:19] WARNING[21897]: app_dial.c:2341 dial_exec_full: Unable to create channel of type 'IAX2' (cause 20 - Unknown)
[Aug 27 19:59:19] WARNING[21897]: app_dial.c:2341 dial_exec_full: Unable to create channel of type 'IAX2' (cause 20 - Unknown)
-- SIP/6000-00000066 connected line has changed. Saving it until answer for SIP/trunk_viva-00000065
-- SIP/6001-00000067 connected line has changed. Saving it until answer for SIP/trunk_viva-00000065
-- SIP/6003-00000068 connected line has changed. Saving it until answer for SIP/trunk_viva-00000065
-- SIP/6000-00000066 is ringing
-- Got SIP response 486 "Busy Here" back from 83.235.190.161:17183
-- SIP/6003-00000068 is busy
-- SIP/6001-00000067 is ringing
== Using UDPTL CoS mark 5
-- SIP/6000-00000066 connected line has changed. Saving it until answer for SIP/trunk_viva-00000065
-- SIP/6000-00000066 answered SIP/trunk_viva-00000065
-- Locally bridging SIP/trunk_viva-00000065 and SIP/6000-00000066
[Aug 27 19:59:36] WARNING[10792]: chan_sip.c:3736 retrans_pkt: Retransmission timeout reached on transmission 063c00765680cd0a65aa39887a9d79fb@83.235.24.87 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response
[Aug 27 19:59:36] WARNING[10792]: chan_sip.c:3765 retrans_pkt: Hanging up call 063c00765680cd0a65aa39887a9d79fb@83.235.24.87 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
== Spawn extension (ringroups-custom-1, s, 2) exited non-zero on 'SIP/trunk_viva-00000065'
debug:
--- (8 headers 0 lines) ---
Retransmitting #4 (NAT) to 83.235.24.86:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 83.235.24.86;branch=z9hG4bK324a.0512f147.0;received=83.235.24.86;rport=5060
Via: SIP/2.0/UDP 83.235.24.87:5060;received=83.235.24.87;branch=z9hG4bK186862c7;rport=5060
Record-Route: <sip:83.235.24.86:5060;lr=on;ftag=as76df670a;vsf=AAAAAAAAAAAAAAAAAAAATlpYUx1SXDI0Ljg3>
From: "numberwhocallingme" <sip:numberwhocallingme@viva.gr>;tag=as76df670a
To: <sip:myvoipnumber@83.235.24.86:5060>;tag=as647edeb4
Call-ID: 545ee0cf6648ec546724db6963efc93f@83.235.24.87
CSeq: 102 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:s@77.49.80.112:5070>
Content-Type: application/sdp
Content-Length: 317
v=0
o=root 298456689 298456689 IN IP4 77.49.80.112
s=Asterisk PBX 1.8.15-cert7
c=IN IP4 77.49.80.112
t=0 0
m=audio 12790 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:77.49.80.112:60080 --->
<------------->
Retransmitting #5 (NAT) to 83.235.24.86:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 83.235.24.86;branch=z9hG4bK324a.0512f147.0;received=83.235.24.86;rport=5060
Via: SIP/2.0/UDP 83.235.24.87:5060;received=83.235.24.87;branch=z9hG4bK186862c7;rport=5060
Record-Route: <sip:83.235.24.86:5060;lr=on;ftag=as76df670a;vsf=AAAAAAAAAAAAAAAAAAAATlpYUx1SXDI0Ljg3>
From: "numberwhocallingme" <sip:numberwhocallingme@viva.gr>;tag=as76df670a
To: <sip:myvoipnumber@83.235.24.86:5060>;tag=as647edeb4
Call-ID: 545ee0cf6648ec546724db6963efc93f@83.235.24.87
CSeq: 102 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:s@77.49.80.112:5070>
Content-Type: application/sdp
Content-Length: 317
v=0
o=root 298456689 298456689 IN IP4 77.49.80.112
s=Asterisk PBX 1.8.15-cert7
c=IN IP4 77.49.80.112
t=0 0
m=audio 12790 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
Retransmitting #6 (NAT) to 83.235.24.86:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 83.235.24.86;branch=z9hG4bK324a.0512f147.0;received=83.235.24.86;rport=5060
Via: SIP/2.0/UDP 83.235.24.87:5060;received=83.235.24.87;branch=z9hG4bK186862c7;rport=5060
Record-Route: <sip:83.235.24.86:5060;lr=on;ftag=as76df670a;vsf=AAAAAAAAAAAAAAAAAAAATlpYUx1SXDI0Ljg3>
From: "numberwhocallingme" <sip:numberwhocallingme@viva.gr>;tag=as76df670a
To: <sip:myvoipnumber@83.235.24.86:5060>;tag=as647edeb4
Call-ID: 545ee0cf6648ec546724db6963efc93f@83.235.24.87
CSeq: 102 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:s@77.49.80.112:5070>
Content-Type: application/sdp
Content-Length: 317
v=0
o=root 298456689 298456689 IN IP4 77.49.80.112
s=Asterisk PBX 1.8.15-cert7
c=IN IP4 77.49.80.112
t=0 0
m=audio 12790 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
[Aug 27 20:09:06] WARNING[10792]: chan_sip.c:3736 retrans_pkt: Retransmission timeout reached on transmission 545ee0cf6648ec546724db6963efc93f@83.235.24.87 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[Aug 27 20:09:06] WARNING[10792]: chan_sip.c:3765 retrans_pkt: Hanging up call 545ee0cf6648ec546724db6963efc93f@83.235.24.87 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
Scheduling destruction of SIP dialog '0462b63b0c58d2d07a754e844747adcc@77.49.80.112:5070' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:192.168.1.98:5060> for address/port to send to
set_destination: set destination to 192.168.1.98:5060
Reliably Transmitting (NAT) to 77.49.80.112:50266:
BYE sip:192.168.1.98:5060 SIP/2.0
Via: SIP/2.0/UDP 77.49.80.112:5070;branch=z9hG4bK049a0a78;rport
Max-Forwards: 70
From: "numberwhocallingme" <sip:numberwhocallingme@77.49.80.112:5070>;tag=as3a2d634b
To: <sip:6000@192.168.1.98:5060;ob>;tag=2ee58d1c2c404b0d88412e295073c75e
Call-ID: 0462b63b0c58d2d07a754e844747adcc@77.49.80.112:5070
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
== Spawn extension (ringroups-custom-1, s, 2) exited non-zero on 'SIP/trunk_viva-0000006d'
Scheduling destruction of SIP dialog '545ee0cf6648ec546724db6963efc93f@83.235.24.87' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:83.235.24.86:5060;lr=on;ftag=as76df670a;vsf=AAAAAAAAAAAAAAAAAAAATlpYUx1SXDI0Ljg3> for address/port to send to
set_destination: set destination to 83.235.24.86:5060
Reliably Transmitting (NAT) to 83.235.24.86:5060:
BYE sip:numberwhocallingme@83.235.24.87 SIP/2.0
Via: SIP/2.0/UDP 77.49.80.112:5070;branch=z9hG4bK3a662d30;rport
Route: <sip:83.235.24.86:5060;lr=on;ftag=as76df670a;vsf=AAAAAAAAAAAAAAAAAAAATlpYUx1SXDI0Ljg3>
Max-Forwards: 70
From: <sip:myvoipnumber@83.235.24.86:5060>;tag=as647edeb4
To: "numberwhocallingme" <sip:numberwhocallingme@viva.gr>;tag=as76df670a
Call-ID: 545ee0cf6648ec546724db6963efc93f@83.235.24.87
CSeq: 102 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0
---
<--- SIP read from UDP:77.49.80.112:50266 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 77.49.80.112:5070;rport=5070;received=77.49.80.112;branch=z9hG4bK049a0a78
Call-ID: 0462b63b0c58d2d07a754e844747adcc@77.49.80.112:5070
From: "numberwhocallingme" <sip:numberwhocallingme@77.49.80.112>;tag=as3a2d634b
To: <sip:6000@192.168.1.98;ob>;tag=2ee58d1c2c404b0d88412e295073c75e
CSeq: 103 BYE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '0462b63b0c58d2d07a754e844747adcc@77.49.80.112:5070' Method: INVITE
Retransmitting #1 (NAT) to 83.235.24.86:5060:
BYE sip:numberwhocallingme@83.235.24.87 SIP/2.0
Via: SIP/2.0/UDP 77.49.80.112:5070;branch=z9hG4bK3a662d30;rport
Route: <sip:83.235.24.86:5060;lr=on;ftag=as76df670a;vsf=AAAAAAAAAAAAAAAAAAAATlpYUx1SXDI0Ljg3>
Max-Forwards: 70
From: <sip:myvoipnumber@83.235.24.86:5060>;tag=as647edeb4
To: "numberwhocallingme" <sip:numberwhocallingme@viva.gr>;tag=as76df670a
Call-ID: 545ee0cf6648ec546724db6963efc93f@83.235.24.87
CSeq: 102 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0
---
Really destroying SIP dialog '4d133b932e55f0846b4f6a090873b1e1@77.49.80.112:5070' Method: INVITE
<--- SIP read from UDP:83.235.24.86:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 77.49.80.112:5070;received=77.49.80.112;branch=z9hG4bK3a662d30;rport=5070
From: <sip:myvoipnumber@83.235.24.86:5060>;tag=as647edeb4
To: "numberwhocallingme" <sip:numberwhocallingme@viva.gr>;tag=as76df670a
Call-ID: 545ee0cf6648ec546724db6963efc93f@83.235.24.87
CSeq: 102 BYE
Server: m1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '545ee0cf6648ec546724db6963efc93f@83.235.24.87' Method: INVITE
<--- SIP read from UDP:83.235.24.86:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 77.49.80.112:5070;received=77.49.80.112;branch=z9hG4bK3a662d30;rport=5070
From: <sip:myvoipnumber@83.235.24.86:5060>;tag=as647edeb4
To: "numberwhocallingme" <sip:numberwhocallingme@viva.gr>;tag=as76df670a
Call-ID: 545ee0cf6648ec546724db6963efc93f@83.235.24.87
CSeq: 102 BYE
Server: m1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP:79.131.37.170:5060 --->
<------------->
Really destroying SIP dialog '4656cf707dbdb3c67809038c7eec70a3@77.49.80.112:5070' Method: NOTIFY
Reliably Transmitting (NAT) to 83.235.24.86:5060:
OPTIONS sip:voip.viva.gr SIP/2.0
Via: SIP/2.0/UDP 77.49.80.112:5070;branch=z9hG4bK13af4b79;rport
Max-Forwards: 70
From: "asterisk" <sip:myvoipnumber@77.49.80.112:5070>;tag=as48b9a5cf
To: <sip:voip.viva.gr>
Contact: <sip:myvoipnumber@77.49.80.112:5070>
Call-ID: 080e8e32272c9d786374803419bdbc7b@77.49.80.112:5070
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Wed, 27 Aug 2014 17:09:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
Reliably Transmitting (NAT) to 91.217.155.70:5060:
OPTIONS sip:sip.ephone.gr SIP/2.0
Via: SIP/2.0/UDP 77.49.80.112:5070;branch=z9hG4bK1f87070a;rport
Max-Forwards: 70
From: "asterisk" <sip:9438607905@77.49.80.112:5070>;tag=as7b0824eb
To: <sip:sip.ephone.gr>
Contact: <sip:9438607905@77.49.80.112:5070>
Call-ID: 2c39710932a50d33455f783f24350c61@77.49.80.112:5070
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Wed, 27 Aug 2014 17:09:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:83.235.24.86:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 77.49.80.112:5070;branch=z9hG4bK13af4b79;rport=5070
From: "asterisk" <sip:myvoipnumber@77.49.80.112:5070>;tag=as48b9a5cf
To: <sip:voip.viva.gr>;tag=ae428d0fc435bd570aaae338d70dfd1f.f041
Call-ID: 080e8e32272c9d786374803419bdbc7b@77.49.80.112:5070
CSeq: 102 OPTIONS
Server: Viva VoIP
Content-Length: 0
<------------->