[Asterisk-Users] Asterisk featdmf signalling.

DEBUG[18897]: DTMF digit: 6 on Zap/73-1 Jan 6 11:00:21 DEBUG[18897]: DTMF digit: 1 on Zap/73-1 Jan 6 11:00:21 DEBUG[18897]: DTMF digit: 0 on Zap/73-1 Jan 6 11:00:21 DEBUG[18897]: DTMF digit: 6 on Zap/73-1 Jan 6 11:00:21 DEBUG[18897]: DTMF digit: # on Zap/73-1 Regards Michael Baird

Re: [Asterisk-Users] Asterisk featdmf signalling.

On Mon, 2006-01-09 at 14:20 -0600, Dave Weis wrote: On Mon, 9 Jan 2006, Michael Baird wrote: I've recently started PIC'ing some calls into a asterisk box across a feature group D trunk from Verizon. Everything seems to work ok, except for some reason Asterisk doesn't grab the full caller ID

[Asterisk-Users] Asterisk MWI and Realtime

this work properly, short of going back to a flat file for voicemail.conf? Regards Michael Baird ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit

Re: [Asterisk-Users] Asterisk MWI and Realtime

On Fri, 2005-08-05 at 14:06 -0500, Matthew Boehm wrote: Michael Baird wrote: I'm testing my asterisk system and the realtime backend. My Asterisk build is rather aged, 03/18/2005 CVS. I have successfully moved Sip peers and Voicemail boxes to the realtime database backend and this works

Re: [Asterisk-Users] Asterisk MWI and Realtime

to verify asterisk is working properly, but with your input, I think the ATA is the issue now. Regards Michael Baird turned on there. I do get the Sip Notify when a message is left, just no stutter tone when picking up the phone. I will also refresh my build as well, hopefully no big changes

[Asterisk-Users] Asterisk and Max TNT

. Regards Michael Baird ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] VOIP Gateways Asterisk

to use a TNT with asterisk, and it works properly. I used Asterisk to talk to them via SIP, didn't try mgcp but it should work fine. Regards Michael Baird Hi all, I was just wondering if someone could help me with info on VOIP Gateways. We are planning to do an * install in an apartment building

Re: [Asterisk-Users] FGD Support

Add me to the list, of needing this functionality (MF w/FGD). I'm exactly in the same position as Jason. Regards Michael Baird Has Digium gotten back with you on the quote or if they can/will do it? Thank you, Jason Miller Eminent Network Technologies Inc. d/b/a Interlinc.net Phone

[Asterisk-Users] Asterisk Max TNT

it egresses if the SIP INVITE messages are not enclosed in brackets, can anyone confirm this? If so has someone tried 11.0.2 to see if it works with asterisk properly. Regards -- Michael Baird [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk

[Asterisk-Users] MF Trunk Signaling

me some pointers, that would be really helpful. I haven't been able to get the test T1 out of the yellow state yet, so I'm pretty sure my zaptel.conf is wrong. Regards Michael Baird. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

Re: [Asterisk-Users] MF Trunk Signaling

James, d4 ami, I'm going to recheck my jumper cable then. I have this T1 connected to another device, the other 3 T1's on the asterisk box continue to work fine as well (Max TNT, and it takes calls fine AMI/D4). Regards Michael Baird MF is problem number 2. Your yellow light will most likely

Re: [Asterisk-Users] Adit 600 as VoIP router (MGCP) and Asterisk

the MGCP option for my Adit 600 to test with, the Occams are a bit flaky and this makes me nervous and they only support g.711. I wish I could find a similar product with SIP instead of MGCP, Asterisk MGCP support has kind of fallen behind the other protocols. Regards Michael Baird

[Asterisk-Users] Realtime Engine

I'm going to be testing the new realtime stuff further in the next few days, and just wanted some clarification on a couple of things before I start on it. I believe I can store any config file in a external config such as mgcp.conf for example, by adding it to extconfig.conf with the below

RE: [Asterisk-Users] VoIP-to-TDM processing on-card?

(there is one/two of these fellows on every mailing list), don't let him ruin your day, this list is quite helpful and many guys will give you a good answer without the extra attitude. Regards Michael Baird ___ Asterisk-Users mailing list Asterisk

RE: [Asterisk-Users] SS7 and Asterisk solution

Same here, interested in the details of a SS7/Asterisk solution. Regards MIKE Steve, I also would be very interested in getting those details. We would very much like to move forward with SS7, please feel free to contact me off list. Cheers, Ben Merrills Griffin Internet

Re: [Asterisk-Users] MySQL Realtime Driver

to reload mgcp when u make a change. Matthew - Original Message - From: Michael Baird [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, January 12, 2005 2:25 PM Subject: Re: [Asterisk-Users] MySQL Realtime

Re: [Asterisk-Users] PRI concentrator

Heh, yea a SWITCH. My Lucent TNT's will do it also, however nothing will be as cheap as multiple asterisk boxes terminating the PRI's from the PSTN I think. Regards MIKE Hey gang, We currently have a class 3 switch (CSX) that..well..it sucks. It does terrible CDR writes, doesn't support LCR,

[Asterisk-Users] RDNIS

with the same results for both. I'm trying to use this for Voicemail. I'm using CVS-HEAD-10/16/04, any advise or pointers would be appreciated. I get the called from fine, and it puts the forwarded to number as the called to, the RDNIS value is blank. Regards Michael Baird

Re: [Asterisk-Users] RDNIS

Thanks, this helps a ton, we get our PRI's from a CLEC we work with, this will help them provide the info I require. Regards Michael Baird Michael, I have asked a very similar question over the past week and gotton no answer... Seems like no one here knows much about it...or just doesn't want

Re: [Asterisk-Users] Wonderful Success with PAP2-NA

the rest of this year. Press release at http://www.voip-info.org/tiki-index.php?page=Linksys Regards Michael Baird [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE

[Asterisk-Users] res_odbc app_realtime

reload_config: Unable to load config sip.conf, SIP disabled I tried setting up voicemail.conf with similar errors, both voicemail and sip work fine when accessed from flatfiles. Regards Michael Baird ___ Asterisk-Users mailing list [EMAIL PROTECTED] http

RE: [Asterisk-Users] res_odbc app_realtime

. I'm using todays asterisk CVS. Is there a better way for me to debug what's happening, asterisk - doesn't show me anything when a sip call comes in, other then it can't find the extension. SQL Error SQLError | enter: szErrorMsg: bfffd530 SQLError Regards Michael Baird Also your database

Re: [Asterisk-Users] SIP peers in MySQL Database

Matthew, will the new method for dynamic storage of sip peers be based on res_config? Or will it be some new method, is there anything I can look at to start researching it yet? Regards Michael Baird You have obviously never posted to any kind of mailing list before. Sometimes you may have

[Asterisk-Users] SS7 capability

We are looking into turning up SS7, my Lucent TNT's support it by using a SS7 Gateway, are there any open source products that will serve this purpose. I've looked at openss7.org a little bit, it looks kind of stagnant, and it doesn't appear asterisk has any of the functionality I'm looking for.

Re: Anti-Ex Girl Friend logic (was Re: [Asterisk-Users] Set context based on CID...)

I do it through AGI, I send the call to an external perl script, check the called-from-id against a mysql database, then send the call back to a context based on a ruleset I use, call-approved/call-not-approved/no digits received. Each context having a different voice message, so that the caller

Re: [Asterisk-Users] AGI accountcode.

Wonderful, that's what I was looking for. Regards MIKE I've setup cdr_mysql and am using AGI to authenticate users based on the called-from # (callerid), use the AGI perl module. Looking at the info stored in the caller detail, I see a field called accountcode, is it possible for me to set

[Asterisk-Users] Extension handling.

I've designed a voice menu, someone calls a certain extension, and I send them to another context via a goto, and play a background message. After playing this message can I provide them dialtone from asterisk again, in order to dial out? Regards MIKE

[Asterisk-Users] Sound Quality.

I've been using asterisk for a while, only for dialout from a SIP client over a PRI - PSTN, this works great. Now I have a need to also dialin to asterisk over the PRI/TDM, I've been testing by creating an extension, and essentially playing back a recording on that extension. If I access the

Re: [Asterisk-Users] G.711 Codec

I use g711 ulaw, works fine, g711 is uncompressed (64k) as far as I'm aware. Regards MIKE Has anyone done anything with Asterisk using the G.711 codec? Also, is there a uncompressed option so that you could assign a single port to be unconpressed audio? Thanks! Stu

Re: [Asterisk-Users] MeetMe PIN functionality

Hrm, interesting, is it possible in asterisk to authenticate VoIP users based on called-from number, if so, could someone post sample syntax (especially if I can auth from a list, mysql/flatfile it doesn't matter, for a universal extension, this is outbound only). Regards MIKE On Thu,

Re: [Asterisk-Users] Asterisk Website Theme

How about a web interface module for asterisk itself, a webmin module would be wonderful. Regards MIKE On Fri, 2003-03-21 at 10:24, Mark Spencer wrote: Dear Asterisk Community, Due to the loss of our dear CVS and database server, the fact that the old asterisk web site was pretty lame

[Asterisk-Users] MGCP Config

Hey guys, the occam people are coming in Monday with boxes for us to test, they are controlled via MGCP, I've never used MGCP with asterisk (or ever) and haven't been able to find much info about them on the list. My guestimate config is as follows, let me know if I'm on the right track.

Re: R: [Asterisk-Users] Cisico ATA licence

Well at least we don't have Janet Reno sending tanks in on civilians anymore. Regards MIKE On Thu, 2003-03-06 at 11:59, Steven Critchfield wrote: On Thu, 2003-03-06 at 10:44, Florian Overkamp wrote: At 10:20 6-3-2003 -0600, you wrote: On Thu, 2003-03-06 at 04:25, Matteo Brancaleoni wrote: