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 <[email protected]> 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:* [email protected] [mailto: > [email protected]] *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 <[email protected]> wrote: > > Set verbose to 5 and see if you get a CLI output. > > > ------------------------------ > > *From:* [email protected] [mailto: > [email protected]] *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 <[email protected]> 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:* [email protected] [mailto: > [email protected]] *On Behalf Of *David Nickel > *Sent:* Wednesday, May 05, 2010 8:23 AM > *To:* [email protected] > *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
