similar to: asterisk-users Digest, Vol 81, Issue 27

Hey Guys, Whenever i calling any extension i am getting following WARNING messages do you have any idea they coming from where? -Satish shirley*CLI> == Using SIP RTP CoS mark 5 -- Executing [7623 at from-sip:1] Macro("SIP/7527-00000008", "stdexten,7623,sip/7623&sip/7624") in new stack -- Executing [s at macro-stdexten:1] Dial("SIP/7527-00000008",

Hi, I would have thought that when spawning a channel using the Originate() dialplan command, variables prefixed with two underscores would be preserved. However this does not work in the following case. Dialplan code: [intern] exten => 200,1,Set(__myvar="foo") exten => 200,n,Originate(Local/123 at test_orig,exten,dummy) [test_orig] exten => 123,1,NoOp(${myvar}) exten =>

Hi, I'm trying to set up remote voicemail pickup. I've created the following dialplan, but when I press *, I am not sent to voicemailmain. The unavailable message just continues to play as normal. exten => 2345551111,1,Set(MAILBOXID=1) exten => 2345551111,n,Set(MAILBOXCONTEXT=company3) exten => 2345551111,n,Voicemail(${MAILBOXID}@${MAILBOXCONTEXT},u) exten =>

On 04/05/2011 03:06 PM, asterisk-users-request at lists.digium.com wrote: > Message: 12 > Date: Tue, 5 Apr 2011 13:36:21 -0500 > From: Sherwood McGowan<sherwood.mcgowan at gmail.com> > Subject: Re: [asterisk-users] Iptables configuration to handle brute, > force registrations? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at

Where this revers IP comes from ? == Using SIP RTP CoS mark 5 -- Executing [7623 at from-sip:1] Macro("SIP/7527-0000006b", "stdexten,7623,SIP/7623") in new stack -- Executing [s at macro-stdexten:1] ChanIsAvail("SIP/7527-0000006b", "SIP/7623&IAX2/7623,20,t") in new stack -- Hungup 'IAX2/0.0.29.199:4569-5255' -- Executing [s at

I listened to all the demos you showed. My ear discerns a little muffling and minor "slushiness" in the GSM files you sent, along with a much more narrow bandwidth, mainly on the high end side, and Allison either has a mild whistling s or slushy s sound in her voice or the producer didn't properly compress it to "de-ess" the recording. Or, I could just be rather tired.

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:

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:

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:

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

I'm currently using the Linksys PAP2, and since there's a shortage I'm looking for different devices. I'm mainly looking at the Sipura SPA sets since they are the base of the pap2. Anyone else have experience using them, and which one? Thanks Sherwood McGowan -------------- next part -------------- An HTML attachment was scrubbed... URL:

I run a bunch of the Linksys ATA's.. I always use rfc2833 for DTMF. works very well and have never had a problem with it. ..o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of

Guys, I've been poking around trying to find a good answer for this via voip-info, google, etc... Haven't found anything that helps, so maybe you mates could. A lot of my customers are using Linksys UAs (router/ATA PAP2) and some using Sipura SPA-2002s. Every once in a while, the customer will get one-way audio. I've read that this is commonly caused by the outgoing RTP port not

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.....

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

Hello All, I came to know from a friend that using LLVM insfrastucture one can convert C++ programs to C. I needed this for my cross-compiler because we don't have support for C++ compilation in our cross-compiler. I tried following: http://llvm.org/docs/FAQ.html#translatecxx While I can generate .bc its llc which gives error. Then I also tried "clang" as oppose to llvm-g++. Here

Hello everyone, I successfully set up the mail_crypt plugin using folder keys, and require user's key to be encrypted with a password using mail_crypt_require_encrypted_user_key = yes. As I'm trying to streamline the process of creating a user, and want to develop an application in PHP to help me in that process, I'm very interested in the doveadm HTTP API. Although the

please send me the ways to change asterisk port from 5060 to 5061 i need to configure it because we are already using 5060 port in router then we cant use it again we have to configure other sip server so please suggest me a way.......................... On 4/10/11, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing

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