[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



<--- 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) ---



<--- 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

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(CRM)system? visit http://www.agile.co.ke/crm.php and find out how our
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