[asterisk-users] All circuits are busy now on outgoing trunk call
James Mutuku
listmutuku at gmail.com
Wed Mar 21 06:30:36 CDT 2012
my sip traces are below
Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 192.168.9.250 port 17722
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.9.251:5060:
INVITE sip:0722490994 at 192.168.9.251 SIP/2.0
Via: SIP/2.0/UDP 192.168.9.250:5060;branch=z9hG4bK111ef687;rport
Max-Forwards: 70
From: "pbxserver" <sip:Unknown at 192.168.9.250>;tag=as66c75bd7
To: <sip:0722490994 at 192.168.9.251>
Contact: <sip:Unknown at 192.168.9.250>
Call-ID: 4ce934e47314480b45073a7d768cb0c8 at 192.168.9.250
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.13
Date: Wed, 21 Mar 2012 11:19:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286
v=0
o=root 1836379524 1836379524 IN IP4 192.168.9.250
s=Asterisk PBX 1.6.2.13
c=IN IP4 192.168.9.250
t=0 0
m=audio 17722 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
-- Called 6fxogateway/0722490994
[0K
<--- SIP read from UDP:192.168.9.251:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.9.250:5060;rport=5060;branch=z9hG4bK111ef687
To: <sip:0722490994 at 192.168.9.251>
From: "pbxserver" <sip:Unknown at 192.168.9.250>;tag=as66c75bd7
CSeq: 102 INVITE
Call-ID: 4ce934e47314480b45073a7d768cb0c8 at 192.168.9.250
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
[0K
<--- SIP read from UDP:192.168.9.251:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.9.250:5060;rport=5060;branch=z9hG4bK111ef687
To: <sip:0722490994 at 192.168.9.251>;tag=1332328154302b4aa3-f15d-4eb7-beee-97782e7cbd06
From: "pbxserver" <sip:Unknown at 192.168.9.250>;tag=as66c75bd7
CSeq: 102 INVITE
Call-ID: 4ce934e47314480b45073a7d768cb0c8 at 192.168.9.250
Server: Epygi Quadro SIP User Agent/v5.1.16 (QUADRO-FXO)
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 192.168.9.251:5060:
ACK sip:0722490994 at 192.168.9.251 SIP/2.0
Via: SIP/2.0/UDP 192.168.9.250:5060;branch=z9hG4bK111ef687;rport
Max-Forwards: 70
From: "pbxserver" <sip:Unknown at 192.168.9.250>;tag=as66c75bd7
To: <sip:0722490994 at 192.168.9.251>;tag=1332328154302b4aa3-f15d-4eb7-beee-97782e7cbd06
Contact: <sip:Unknown at 192.168.9.250>
Call-ID: 4ce934e47314480b45073a7d768cb0c8 at 192.168.9.250
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.13
Content-Length: 0
--
Best Regards,
James Mutuku Ndeti
Agile Systems Limited
+254722490994
www.agile.co.ke
www.zetu.co.ke
Has your organization implemented a customer relationship management
(CRM)system? visit http://www.agile.co.ke/crm.php and find out how our
CRM can help you achieve better customer satisfaction and sales
More information about the asterisk-users
mailing list