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I am building a new Asterisk server here at the office, and I'm wondering if there are any downsides to creating my dialplan with AEL. It seems more intuitive (to me), but I'm not sure if there are any pitfalls I need to be aware of first. We use this for internal extensions, 8 pots lines, and our answering service which gets about 500 incoming calls a day down our T1. Also, one more

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: http://lists.digium.com/pipermail/asterisk-users/attachments/20050906/cc8e0a4e/attachment.htm

Hi! Guess I am doing something totally wrong here: Some smart person could maybe plz tell me what.

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 lists.digium.com> > Cc: Bill Michaelson<bill at cosi.com&g...

In my Asterisk CLI I get Ring/Off-hook in strange state 6 when i make a call into the system, the system claims to answer the call, and do the things in the dial plan, but I just hear ringing on the phone I'm calling in from. I am using a Sangoma A200 4 Port Analog card. my wanrouter version: WANPIPE Release: 3.3.6 asterisk -V: PBXtra Core fon_o_1.2.17 Any ideas? Daniel Lockard

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: http://lists.digium.com/pipermail/asterisk-users/attachments/20050902/545c2460/attachment.htm

...ritance with dialplan command Originate > (Naomi Rosenberg) > 3. Re: CRC Zaptel.conf (Shaun Ruffell) > 4. Re: Variable inheritance with dialplan command Originate > (Jim Dickenson) > 5. Re: Variable inheritance with dialplan command Originate > (Sherwood McGowan) > 6. Re: Variable inheritance with dialplan command Originate > (Sherwood McGowan) > 7. Re: IAX2/0.0.29.199 (satish patel) > 8. Re: asterisk login to voicemail (vip killa) > 9. Re: asterisk login to voicemail (satish patel) > 10. Re: IAX2/0.0.29.199 (Paul Bela...

I recently configured a SIP peer which i must specify my fromuser as my ten digit "DID". I send calls to this peer, but whenever Asterisk sends an options message, the fromuser is "asterisk". Is this a bug? Or is there some other config I must make ? register = 2155551941:123456 at 10.0.138.226/2155551941~600 [peer](!) type=peer context=inbound qualify=yes

...le=${recfile}) ; The log output: -- Executing [s at menu.main:1] Set("SIP/9999-00000000", "____recfile=12974953060") in new stack -- Executing [s at menu.main:2] Set("SIP/9999-00000000", "____logfile=") in new stack Anybody have thoughts? Thanks, S McGowan -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110212/0e41cdce/attachment.htm>

I need help translating extensions.conf to AEL: [default] exten => _X.,1,Set(DID=${EXTEN:6}) exten => _X.,n,Goto(continue,1) exten => _1X.,1,Set(DID=${EXTEN:7}) exten => _1X.,n,Goto(continue,1) exten => continue,1,Noop(${DID}) exten => continue,n,Set(GROUP(IAX)=incoming) exten => continue,n,GotoIf($[${MATH(${GROUP_COUNT(incoming at IAX)}+${GROUP_COUNT(outgoing at

...er own professional competition...... that's all...nothing super rude, but I had to get that one out there.... I get annoyed when I answer about 12-13 questions (all in separate emails, mind you) from someone, and then I never get even find out if I was successful in helping them -- Sherwood McGowan Telecommunications and VOIP Consultant -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110510/c7d4c206/attachment.htm>

...ow something that I'm missing for the above mentioned devices? They're all the 2 line version of the ATA and/or router configs (wireless and wired) Thank you all in advance for your thoughts and comments. I apologize in advance if I missed something that was publicly available. Sherwood McGowan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050930/85c8dfff/attachment.htm

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: http://lists.digium.com/pipermail/asterisk-users/attachments/20050826/cecbe1e9/attachment.htm

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: http://lists.digium.com/pipermail/asterisk-users/attachments/20050906/e1d49fc1/attachment.htm

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

Dear All After various failed attempts at using Wine I have installed this from the Mandriva 2006 DVD. I have version ?Wine 20050725?. I have installed Paint Shop Pro as follows: As a user I typed: ?Winecfg? and when in the configuration programme I went to the Application tab > Add Application button > browse to the installation executable: ?/mnt/cdrom/autorun.exe? > Open button

I am trying to setup one time caller id block on my system(activated when an incoming call matches *811XXXXXXXXXX), and I have had little to no luck. Could you take a look at my context/macro definition and help me figure out what I am missing? Here is my context for my dialplan: include=default plancomment=user-default

I was browsing apps and looking for games that run in Wine as Gold or better. The issue is that I see many apps listed as gold that require recompiling Wine with patches to get them to work. From my understanding, any application that requires code changes should be listed as garbage and tied to a patch/bug to be able to operate. That and any application that ranks higher should run under a

Gentlemen, I have googled, searched the mailing list archives, and even spoke on the IRC channel, but have not found an answer to the following problem. I am attempting to retrieve multiple columns in an ODBC query using ARRAY per the solutions offered by many individuals. My dialplan code is as follows: exten => _.,n,Set(ARRAY(var1,var2,var3)=${ODBC_LOOKUP(${KEYVAL})}) exten =>

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hey all, First of all, hello again! Been a while since I've posted to the list, but I've been here lurking and watching ;-) Anyway, I wanted to pose a general question to the list to see if it turns up new suggestions for everyone/me. What is your preferred editor when coding in the dialplan? This is mainly aimed at those of you who write