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...if (pri->debug & PRI_DEBUG_Q931_STATE) pri_message(pri, "Timed out looking for release complete\n"); PRI Debug was not on in the asterisk console. Any ideas? My asterisk restarts about twice a day, and drops any current calls in the process. Regards, Mark McQuiggan _ This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified th...

I am using a SIP softphone (X-lite, SIPPS or Firefly) connected to an Asterisk v 1.0.3 PBX. The PBX is also connected via a ISDN-PRI crossover cable to a Avaya Definity Generic 3 PBX via a TE405P card. All outside of the office calls go through the Definity. Here's the issue: Calls to internal SIP extensions, Definity extensions, other offices within our private network (through the

...erson's name on his/her call display (I suppose that this could be called "Called ID"). For example, if I called Joe Blow from my extension 46480 by dialling 41234, I would see "Joe Blow - 41234" (the Called ID) on my call display after a pause, and Joe would see "Mark McQuiggan - 46480". Is this possible on Asterisk using SIP phones (x-ten Eyebeam or Polycom)? At present, if I did the same as above on an Eyebeam, Joe Blow would see my name and extension, but I would see 41234@10.10.10.10. All documentation that I have read deals with Caller ID. Am I missing somet...

...person's name on his/her call display (I suppose that this could be called "Called ID"). For example, if I called Joe Blow from my extension 46480 by dialling 41234, I would see "Joe Blow - 41234" (the Called ID) on my call display after a pause, and Joe would see "Mark McQuiggan - 46480". Is this possible on Asterisk using SIP phones (x-ten Eyebeam or Polycom)? At present, if I did the same as above on an Eyebeam, Joe Blow would see my name and extension, but I would see 41234@10.10.10.10. All documentation that I have read deals with Caller ID. Am I missing somethin...

...es between voicemail accounts. At worst case, I would like our Asterisk users to be able to bounce to an AUDIX mailbox for voicemail storage. At best, I would like the users to use Comedian mail, with AUDIX messages from our head office forwarded automagically to Comedian. Please, help. Mark McQuiggan _ This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any...

...)). I am waiting to see if it solves the D-channel drops. As an aside, I have noted that calls that are hung up before completion, from either the Definity or the BCM, clear with a cause of -1. This can hang a channel on the Definity side as well. Mark. ________________________________ From: McQuiggan, Mark xt46480 Sent: Monday, March 13, 2006 4:20 PM To: 'asterisk-users@lists.digium.com' Subject: Calls not tearing down properly I have the following configuration: Bell CO. <---ISDN---> Definity <---ZAP PRI---> Asterisk <--ZAP PRI---> Nortel BCM. With SIP users...