[asterisk-users] Step-by-Step Asterisk and Cisco 1760 Help - (Solved)

[Jimmy Ezell]

Jonathan And Dan,

Thank you both for the responses.  But the problem turned out to be that - I'm an idiot. I was placing both calls to our company number for the voicemail system which I thought was part of a hunt group.  As it turns out, only one call can come in at a time on that number, and so I was getting a busy signal because the line really was busy (go figure).  I did learn some things, and I post them here for the benefit of all.

First of all Dan thanks for showing me how to get at some debug information on the Cisco 1760.
I ran the command:
#show call history voice brief

I got several sections of text that looked like this (but more of them) on the screen:

17BD : 878 3699467830ms.572 +3470 +214330 pid:2212 Originate 2572210
 dur 00:03:30 tx:10567/1770108 rx:10542/1686720 10  (normal call clearing (16))
 Telephony 0/0 (878) [0/0] tx:210860/210860/0ms g711ulaw noise:-65dBm acom:14dBm
 long duration call detected:n long dur callduration :n/a timestamp:n/a

That first hex number is the call-id (17BD).  Seems there were three sections of text for each call. (One for calling and 2 sections for disconnect)

After I figured out which ones were associated with my failed call, I ran the second command that Dan suggested:
#show call history voice id <call-id> (show call history voice id 17BD)

And it spit out a lot of stuff but I eventually saw an error code:
InternalErrorCode=1.1.182.11.26.0

I found this webpage that helped to decode the error.
http://docwiki.cisco.com/wiki/Cisco_IOS_Voice_Troubleshooting_and_Monitoring_--_Cisco_VoIP_Internal_Error_Codes

The important bits seem to be:
-------------------------------------
182 	Hardware resources unavailable

26 	No application 	The system could not find an application to take the incoming call. Check your call application and dial peer configurations.
------------------------------------
 
Second the configuration changes that Jonathan suggested worked just fine.  Thank you for showing me another way to make this work with trunk groups.  See below for changes he suggested to my Cisco 1760 configuration:

trunk group Outbound
 description - Outbound calling hunt group
 hunt-scheme sequential
!
voice-port 0/0
 trunk-group Outbound 1
!
voice-port 2/0
 trunk-group Outbound 2
!
dial-peer voice 2212 pots
 trunkgroup Outbound
 description Outbound call hunting
 destination-pattern .T
!

One note here is that it failed the same way, (because the line really was busy) but "show call history voice id <call-id>" did not show any error codes in this trunk group configuration.  Not sure why.

Hope this helps someone and thanks again guys!

Jimmy

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Dan Austin
Sent: Thursday, April 23, 2009 10:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Step-by-Step Asterisk and Cisco 1760 Help


Jimmy wrote:
> Dan thank you, yes that seems to help.  It looks like the 
> bridging is happening now and I see the light come on in 
> the second FXO port, but then I get a busy signal after 
> that and the call still does not complete.  If I set the 
> second line as priority 1 it completes the first call on 
> that line and second call gets the busy on the first line.
> I even tried moving the lines to a different FXO card and 
> the result is the same.

> Here is my current config for the cisco dial-peers:


> dial-peer voice 2212 pots
>  preference 2
>  destination-pattern .T
>  port 2/0
>  forward-digits all
> !
> dial-peer voice 2211 pots
>  preference 1
>  destination-pattern .T
>  port 0/0
>  forward-digits all


> Thanks again Dan,  I think I am much closer now.

I think the suggestion by Jonathan will help you finish
off your problem, but what you have listed should also
have worked.

What does your SIP dial-peer look like?

After the second call fails, try issuing this command on
the cisco:
	#show call history voice brief
Then identify the call id of the failed call and use this:
	#show call history voice id <call-id>

That will at least tell you why the call failed.  I have not
worked a lot with the Cisco analog interfaces, but I have
setup a healthy number of ISDN ports, with the type of
roll-over that you are trying to setup.  I can try to help with
the Cisco debug logs if you want to take this off list.

Dan



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