01-20-2008 05:45 PM - edited 03-15-2019 08:19 AM
Hi,
Am attempting to get outbound calls via SIP to my ISP working, inbound SIP works fine.
Calling an external number gives an error tone on the handset.
The objective is that all calls except emergency (000 in Australia) go out via the SIP service. 0 (zero) is dialed to make an external call. My SIP provider is: sip.internode.on.net.
I've attached the config, and the debug cssip calls log is provided below.
What I've been able to glean from the debug is that either the number isn't being translated correctly since the leading zero (0) is supposed to be stripped, or that my end has been unable to negotiate a protocol with the external sip server, or both.
I have a UC520 with c500-advipservicesk9-mz.124-11.XW4.
All help appreciated.
Damian Halloran
----------debug cssip calls----------
(111.111.111.111 represents external IP address of my router, 04123456789 is a mobile number, 0391234567 is my office number)
----------------Call 1------------------------
006922: Jan 21 01:11:11.723: //5890/706DCA6D82CB/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x858164CC
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 301
Called Number : 004123456789
Source IP Address (Sig ): 111.111.111.111
Destn SIP Req Addr:Port : 203.2.134.1:5060
Destn SIP Resp Addr:Port : 203.2.134.1:5060
Destination Name : sip.internode.on.net
006923: Jan 21 01:11:11.723: //5890/706DCA6D82CB/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : No Codec
Negotiated Codec Bytes : 0
Nego. Codec payload : 255 (tx), 255 (rx)
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 111.111.111.111
Source IP Port (Media): 19230
Destn IP Address (Media): 0.0.0.0
Destn IP Port (Media): 0
Orig Destn IP Address:Port (Media): 0.0.0.0:0
-----------------------end call 1------------------------
----------------------Call 2-----------------------------
006930: Jan 21 01:13:38.803: //5902/D5C42C0D82ED/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x85813E00
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 301
Called Number : 00391234567
Source IP Address (Sig ): 111.111.111.111
Destn SIP Req Addr:Port : 203.2.134.1:5060
Destn SIP Resp Addr:Port : 203.2.134.1:5060
Destination Name : sip.internode.on.net
006931: Jan 21 01:13:38.803: //5902/D5C42C0D82ED/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : No Codec
Negotiated Codec Bytes : 0
Nego. Codec payload : 255 (tx), 255 (rx)
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 111.111.111.111
Source IP Port (Media): 16912
Destn IP Address (Media): 0.0.0.0
Destn IP Port (Media): 0
Orig Destn IP Address:Port (Media): 0.0.0.0:0
006932: Jan 21 01:13:38.803: //5902/D5C42C0D82ED/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 102
Disconnect Cause (SIP) : 408
---------------debug cssip calls-------------
01-20-2008 06:02 PM
01-20-2008 08:53 PM
The problem is now 90% fixed.
Problem one was that I was sending the internal extension no. to my ISP and they wanted the telephone number of the service.
Second was that the number translations were wrong.
I can now make calls but only if I dial the area code. If I try to make local calls without the area code the call fails.
Following is the translation ruleset:
voice translation-rule 2
rule 1 /^09/ /039/
rule 2 /^05/ /035/
rule 3 /^0/ //
Applied to:
voice translation-profile SIPOutgoing
translate calling 3
translate called 2
And when testing this is the result:
test voice trans 2 052554616
Matched with rule 2
Original number: 052554616 Translated number: 0352554616
Original number type: none Translated number type: none
Original number plan: none Translated number plan: none
cellardoor#test voice trans 2 00352554616
Matched with rule 3
Original number: 00352554616 Translated number: 0352554616
Original number type: none Translated number type: none
Original number plan: none Translated number plan: none
As can be seen both numbers produce the same result but the former does not work and doesn't even get into SIP (nothing shows in ccsip debug all for that call), but the latter does work.
Any assistance appreciated.
Damian Halloran
01-22-2008 03:48 AM
The problem is now fixed completely to the point where a zero "0" doesn't need to be dialed to get an outside line which is fine for this very small install.
The reason why it wasn't showing in the debug was because the number was the incorrect length for the dial-peer rule I had created.
All good now.
BTW anyone is Australia trying to use CCA to set up one of these boxes - don't. There is a bug which mucks up the dialing plans in the config. The bug is apparently known by Cisco. I believe it is to do with our use of 000 for emergency, and our traditional use of 0 for outside lines.
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