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-boun...@lists.digium.com [mailto:asterisk-users-boun...@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 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users