Ok. I'm confused. I was interpreting what you wrote to say that you are doing this:
1. pick up sip phone attached to pbx1 (1.2 box) 2. dial #5551212 3. command dial(iax2/trunk/5551212,30,r) gets executed on 1.2 box 4. 1.4 box should fall into _XXXXXXX and do DAHDI dial? If this is correct, where is the IAX command in your CLI output. _____ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Nickel Sent: Wednesday, May 05, 2010 10:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hash Dial Pattern Problems I am on the 1.2 box and see nothing with the verbose cranked up. I do see the following when tailing the asterisk full log during the calls: May 5 11:09:46 DEBUG[26538] chan_sip.c: Setting NAT on RTP to 0 May 5 11:09:46 DEBUG[26538] chan_sip.c: Checking SIP call limits for device 3000 May 5 11:09:46 DEBUG[26538] chan_sip.c: Stopping retransmission on 'Njg3MjI5N2IzNDk3NWYxZTMzMzFmMzEwNzc2ZDE1NTE.' of Response 2: Match Found On Wed, May 5, 2010 at 10:39 AM, Danny Nicholas <da...@debsinc.com> wrote: Ok - you have to be getting something or you wouldn't get that message. You are looking at CLI on the 1.2 or 1.4 box? If you're looking at the 1.4 side, you won't see anything until a connection is made (although you should see some kind of credential reject or something??) _____ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Nickel Sent: Wednesday, May 05, 2010 9:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hash Dial Pattern Problems Nothing..goes directly to "The person you are calling is unavailable". On Wed, May 5, 2010 at 9:46 AM, Danny Nicholas <da...@debsinc.com> wrote: Set verbose to 5 and see if you get a CLI output. _____ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Nickel Sent: Wednesday, May 05, 2010 8:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hash Dial Pattern Problems I am sorry it is not 1.6 but 1.4.30 (argh.. to many boxes) The other box is running 1.2.1 Thanks, David On Wed, May 5, 2010 at 9:28 AM, Danny Nicholas <da...@debsinc.com> wrote: Which 1.6 are you running? I dropped my 1.6.1.6 back to 1.4.30 because my other 2 1.4.30 boxes wouldn't talk to it properly. _____ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Nickel Sent: Wednesday, May 05, 2010 8:23 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Hash Dial Pattern Problems I have two Asterisk boxe. One is running 1.6 and the other 1.2 The users on the 1.2 system press # plus a local 7 digit number to place local calls through the trunk to the 1.6 box. For some reason this dial pattern fails right away with "unavailable". There is no activity in the CLI. Other patterns for the trunk work just fine. Dial pattern: #|. or #|NXXXXXX exten => _#.,1,Dial(IAX2/trunk/${EXTEN:1},30,r) exten => _#.,2,Congestion I have been beating my end with the problem for three days. Any suggestions would be much appreciated. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users