rendered paste body<------------->
--- (13 headers 13 lines) ---
Sending to 192.168.1.105 : 16784 (NAT)
Using INVITE request as basis request - YzNkYjVjNDExMWNjOGFhMWIwNTNiNDgzNjE0MTI5OWQ.
Found peer '600'
Found RTP audio format 107
Found RTP audio format 119
Found RTP audio format 0
Found RTP audio format 98
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.105:11070
Found description format BV32 for ID 107
Found description format BV32-FEC for ID 119
Found description format iLBC for ID 98
Found description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.105:11070
Looking for 913094532511 in numberplan-custom-1 (domain 192.168.1.197)
list_route: hop: <sip:600@192.168.1.105:16784>
<--- Transmitting (no NAT) to 192.168.1.105:16784 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.105:16784;branch=z9hG4bK-d87543-cc49d04911134f07-1--d87543-;received=192.168.1.105;rport=16784
From: "Mark Chaney"<sip:600@192.168.1.197>;tag=381bf221
To: "913094532511"<sip:913094532511@192.168.1.197>
Call-ID: YzNkYjVjNDExMWNjOGFhMWIwNTNiNDgzNjE0MTI5OWQ.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:913094532511@192.168.1.197>
Content-Length: 0
<------------>
-- Executing [913094532511@numberplan-custom-1:1] Macro("SIP/600-0821fc00", "trunkdial|SIP/trunk_1/13094532511") in new stack
-- Executing [s@macro-trunkdial:1] Dial("SIP/600-0821fc00", "SIP/trunk_1/13094532511") in new stack
-- Called trunk_1/13094532511
[Mar 21 05:54:06] WARNING[2952]: chan_sip.c:11731 handle_response_invite: Received response: "Forbidden" from '"Mark SoftPhone" <sip:600@192.168.1.197>;tag=as6bbc7ec3'
-- SIP/trunk_1-082299e0 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-trunkdial:2] Goto("SIP/600-0821fc00", "s-CONGESTION|1") in new stack
-- Goto (macro-trunkdial,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-trunkdial:1] NoOp("SIP/600-0821fc00", "") in new stack
[Mar 21 05:54:16] WARNING[3450]: pbx.c:2460 __ast_pbx_run: Timeout, but no rule 't' in context 'numberplan-custom-1'
Scheduling destruction of SIP dialog 'YzNkYjVjNDExMWNjOGFhMWIwNTNiNDgzNjE0MTI5OWQ.' in 32000 ms (Method: INVITE)
localhost*CLI>
<--- Reliably Transmitting (no NAT) to 192.168.1.105:16784 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.105:16784;branch=z9hG4bK-d87543-cc49d04911134f07-1--d87543-;received=192.168.1.105;rport=16784
From: "Mark Chaney"<sip:600@192.168.1.197>;tag=381bf221
To: "913094532511"<sip:913094532511@192.168.1.197>;tag=as6675c67c
Call-ID: YzNkYjVjNDExMWNjOGFhMWIwNTNiNDgzNjE0MTI5OWQ.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
localhost*CLI>
<--- SIP read from 192.168.1.105:16784 --->
ACK sip:913094532511@192.168.1.197 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.105:16784;branch=z9hG4bK-d87543-cc49d04911134f07-1--d87543-;rport
To: "913094532511"<sip:913094532511@192.168.1.197>;tag=as6675c67c
From: "Mark Chaney"<sip:600@192.168.1.197>;tag=381bf221
Call-ID: YzNkYjVjNDExMWNjOGFhMWIwNTNiNDgzNjE0MTI5OWQ.
CSeq: 2 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
[Mar 21 05:54:28] NOTICE[2952]: chan_sip.c:7055 sip_reregister: -- Re-registration for 17772275104@callcentric.com
[Mar 21 05:54:28] NOTICE[2952]: chan_sip.c:11961 handle_response_register: Outbound Registration: Expiry for callcentric.com is 62 sec (Scheduling reregistration in 47 s)
localhost*CLI>
<--- SIP read from 192.168.1.105:16784 --->
<------------->
--- (0 headers 1 lines) ---
Really destroying SIP dialog 'YzNkYjVjNDExMWNjOGFhMWIwNTNiNDgzNjE0MTI5OWQ.' Method: ACK
localhost*CLI>
<--- SIP read from 192.168.1.105:16784 --->
<------------->
--- (0 headers 1 lines) ---
[Mar 21 05:55:15] NOTICE[2952]: chan_sip.c:7055 sip_reregister: -- Re-registration for 17772275104@callcentric.com
[Mar 21 05:55:15] NOTICE[2952]: chan_sip.c:11961 handle_response_register: Outbound Registration: Expiry for callcentric.com is 63 sec (Scheduling reregistration in 48 s)
localhost*CLI>
<--- SIP read from 192.168.1.105:16784 --->
<------------->
--- (0 headers 1 lines) ---
Reliably Transmitting (no NAT) to 192.168.1.105:16784:
NOTIFY sip:600@192.168.1.197 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.197:5060;branch=z9hG4bK512a7124;rport
From: "Mark Chaney"<sip:600@192.168.1.197>;tag=as00991225
To: "Mark Chaney"<sip:600@192.168.1.197>;tag=ea0f7d15
Contact: <sip:600@192.168.1.197>
Call-ID: YzQyMmUzOGY2MjAxZjMzMmVmMmEzZjZkZmQ5MTg3ZTQ.
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: terminated;reason=timeout
Content-Length: 0
---
localhost*CLI>
<--- SIP read from 192.168.1.105:16784 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.197:5060;branch=z9hG4bK512a7124;rport=5060
Contact: <sip:600@192.168.1.105:16784>
To: "Mark Chaney"<sip:600@192.168.1.197>;tag=ea0f7d15
From: "Mark Chaney"<sip:600@192.168.1.197>;tag=as00991225
Call-ID: YzQyMmUzOGY2MjAxZjMzMmVmMmEzZjZkZmQ5MTg3ZTQ.
CSeq: 102 NOTIFY
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog NOTIFY arrived
Really destroying SIP dialog 'YzQyMmUzOGY2MjAxZjMzMmVmMmEzZjZkZmQ5MTg3ZTQ.' Method: SUBSCRIBE
localhost*CLI>
<--- SIP read from 192.168.1.105:16784 --->
<------------->
--- (0 headers 1 lines) ---
[Mar 21 05:56:03] NOTICE[2952]: chan_sip.c:7055 sip_reregister: -- Re-registration for 17772275104@callcentric.com
[Mar 21 05:56:04] NOTICE[2952]: chan_sip.c:11961 handle_response_register: Outbound Registration: Expiry for callcentric.com is 60 sec (Scheduling reregistration in 45 s)
localhost*CLI>