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Dear * users, in your opinion, when using a * as a public server, is good practice enabling nat=yes in sip.conf for all the peers? Can anyone imagine a scenario when enabling this parameter (even for peers that don't require it) can cause problems? Regards and thanks in advance, Alex -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi List, Is there any way by which we can remove asterisk from machine without deleting folder manually? I did google and gets various solution by no success. even after deleted asterisk will be there ..... ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello list, I need a hand to find the best dialplan failover solution when using two SIP Trunks. My reasons to do failover are: a) one of the two providers could be unreachable b) both providers may be UP but one of them could return a SIP error message (maybe caused by DOWN E1s) Googling I found a few possible solutions: 1.

Hello List, I think I've discovered a little bug in t.38 bug in 1.6.0.22 regarding the speed (T38MaxBitRate) used to send the faxes. Asterisk always responds with a=T38MaxBitRate:2400. I've tried with Patton and Grandstream devices and the result is always the same. Patton ignores the parameter and sends the fax at 9600.

Hello. I'm having a strange problem with the IAX2 channel and IAXmodem and I was hoping to get some light from someone in these lists. On my logs and on the console I'm getting a bunch of lines with: [Oct 23 14:26:18] NOTICE[4417] chan_iax2.c: Peer 'XXX' is now UNREACHABLE! Time: 3 [Oct 23 14:26:28] NOTICE[4413] chan_iax2.c: Peer 'XXX' is now REACHABLE!

Hello list, debugging SIP, I found many empty lines like: <--- SIP read from UDP://XXX.XXX.XXX.XXX:5060 ---> <-------------> The IP address above corresponds to one of my accounts, which is behind a firewall. Is that normal, maybe some firewall that tries to keep a port open, or is my firewall cleaning the SIP Packet? Thanks in

Hello list, I'm having troubles implementing the ${CDR(duration)} & ${CDR(billsec)} variables in this scenario: PEER CALLS OUT -> CALL GOES TO PEER'S DEFAULT OUTGOING CONTEXT -> THE CALL IS SENT TO A MACRO AND GOES IN HANGUP -> THE CALL RETURNS TO EXTENSION h OF PEER'S DEFAULT OUTGOING CONTEXT (here I'm trying to print the variable)

Hello, I need a hand in choosing a small ATA, even with one FXS port, that should do only fax with T38. I've tried Grandstream (ht286 model) but the faxes go out without ECM, even if the Fax machine has ECM enabled. Is there anyone that can recommend an ATA that might do the trick? Thanks, Alex -------------- next part -------------- An HTML

Hello list! I'm having a strange problem with the VoIP Gateway that I'm using to go on the PSTN: if the number on the other end is busy or unavailable I hear an initial RING, generated by Asterisk from what I'm seeing and after that the line goes down with busy signal: Here is the scenario: Softphone *ASTERISK

Hello list! I've run into a strange problem today and I was hoping that someone here has seen this before and maybe can give me a hand: I'm using asterisk 1.6.0.22 in this config: (A)PATTON ISDN ->(B) ASTERISK -> (C)PATTON PRI -> PSTN -> (D)OTHER PBX Strange Problem: USER A calls makes a call to a PBX over the PSTN and ends into an IVR. When the user makes a selection and