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Hey all, I'm trying to create redial on busy for my users, but haven't the foggiest on how to make asterisk route depending on the status code returned over SIP (483, Busy Here?). . . anyone know how to do this? Thanks Sherwood McGowan -------------- next part -------------- An HTML attachment was scrubbed... URL:

I want to see if any of my fellow Asterisk-Users list members have dealt with these guys. I'm a admin for a VOIP provider, and have encountered a few PBX customers that want consulting/support for the IAreaNet provided Asterisk pbxs. These guys are selling AAH servers to the public, and are (at least the techs that I talked to trying to get our service working) not knowledgeable at all.

Gents, this concerns a CVS-HEAD downloaded today. I configured my system as I usually do, including using allowguest=yes to attempt to correct the following problem, but to no avail. When any call comes in from an external server I get this: Oct 1715:36:43 NOTICE[4040]: chan_sip.c:10774 handle_request_register: Failed to authenticate user "+16143691415" /(this is the number making the

Hi, I know that SIP is using port 5060 for session initiation, but which port does it use for audio ? is it dynamically assigned ? Thanks, Adrien -- Adrien Laurent - CIO www.modulis.ca 514-284-2020 ext 202 adrien@modulis.ca

Guys, Is there any work going on to have multiple CLI connections, each getting different outputs? I'd love one user to be able to connect to the server and start (for example) a SIP Debug on a peer, and another to be watching the standard verbose output, etc... I've done some cursory looking online, but found nothing really. Sherwood McGowan -------------- next part -------------- An

Does anyone know if AbsoluteTimeout is working completely? As far as I can see on my systems, I'm still getting occasional hung SIP channels, even though there should be nothing over my setting... cheers Sherwood -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050804/31a0dac0/attachment.htm

Has anyone been able to fix the issue of too many Asterisk remote connections on the CLI? I'm using monitoring scripts that use asterisk -rx 'command' and some of them hang (that doesn't make sense either). The hangs cause open connections to build up (still not that large) until the scripts die, but can cause crashes. thanks Sherwood -------------- next part -------------- An

since delete is a reserved word, what do you name a column in your voicemail options table to allow setting of the delete option for realtime voicemail? Anyone? Sherwood McGowan ViaTalk Level 2 Support VOIP System Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL:

I haven't been able to find an answer....and got no response whatsoever to my previous questions concerning it. Has anyone found a fix for the remote connections to the CLI causing crashes? Also, is there a known limit? I have a huge need for using asterisk -rx in scripts, which seems is kinda why the -x option as added anyway... Anyone? Sherwood McGowan -------------- next part

Sorry, sent with wrong account....read below _____ From: Sherwood McGowan [mailto:sherwood@viatalk.com] Sent: Tuesday, August 23, 2005 8:34 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: SIP DEADLOCK Anyone using a CVS-HEAD pulled later than 8/13? We're runnign a downloaded CVS-HEAD from 8/13/2005 and getting SIP Deadlocks like crazy.....

Anyone have a good tool(s) to use for simulating a bunch of calls? Benchmarking or stress testing? I only need SIP protocol, and do appreciate any replies...I realize I could google it, but I am looking for opinions as well. Sherwood McGowan -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm looking for some information on how the CDR gets the data for the source and destination on the records. My current system sets the callerid to private, via SetCallerID(Private<>) in the dialplan. Unfortunately, this means there's records in my CDR that have no source on them, and as such I'm unable to bill... Any ideas? Sherwood McGowan -------------- next part

When using the Comedian mail system, we've set up a remote access trunk to dump to the VoicemailMain. Problem is, 90% of the time, the system won't recognize the key tones from the PSTN. anyone else have this problem and end up solving it? We're using Real Time setup, but the problem can't be there because it's not even reading the username/password from the caller.

Since I can't seem to get anything figured out for the Comedian system, are there any other systems out there that we can hook asterisk into? Sherwood McGowan ViaTalk Level 2 Support VOIP System Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050912/32ae9d25/attachment.htm

Is WaitExten working in current CVS-HEAD? I'm attempting to use it in some new dialplan code but not getting anything.... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050912/132b82c8/attachment.htm

I'm looking to upgrade my unit, and would like to not have to wait on our company's suppliers to get back to me on it. Thanks in advance for any help -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050919/347b3af8/attachment.htm

I have a need to use cdr_custom and would like to know if anyone has gotten it to work with a mysql cdr backend, and any examples if possible -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050922/bca01301/attachment.htm

I know it's been touched on before, but no answers have been found to the best of my knowledge. I'm using a SIP only setup, with a sip provider giving PSTN and would like to see if anyone has an idea for creating redial busy using ${DIALSTATUS} and possibly MeetMe? I figure something like this, but want to get feedback 1. Get callers last dialed number, if international number, do not

Hello all! I always prefer to get referrals from fellow professionals, and this is such a request. I'm looking for the following: 1. Colocation providers in the chicago area to store a small server for the purpose of setting up a VOIP service (including pstn connection via Digium cards) for between 100-10,000 users. Obviously value is a big part, but reliability and network speed are also

Has anyone else had problems with users being able to press key tones during a voice prompt? I have a few users complaining that some systems will not recognize key presses during them. using current CVS-HEAD, linksys PAP2 UA's, rfc2833 dtmf mode. Thanks Sherwood McGowan -------------- next part -------------- An HTML attachment was scrubbed... URL: