Re: [asterisk-users] SIP user registration and Asterisk Realtime
You still have to enter that information into the sip.conf file. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ast guy Sent: Sunday, 10 February 2008 12:11 a.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] SIP user registration and Asterisk Realtime Hi, I have installed asterisk real time and sip buddies information is being stored in DB. Now I have a question, Asterisk Realtime Server -A Third party SIP server-B Question: Is there any configuration in * RT that it can register with defined sip user on Server-B I was only able to find sip users information in DB not about user registration on other server. -ag No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.20.0/1268 - Release Date: 9/02/2008 11:54 a.m. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.20.2/1270 - Release Date: 10/02/2008 12:21 p.m. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk as a softswitch
Yes you could, but asterisk was designed to be a PBX. I would not use it as soft switch due its limitations. It really depends on how much traffic you are going to be passing. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, 24 August 2007 1:11 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk as a softswitch Mark Quitoriano wrote: Can i use asterisk as a softswitch? This thread has been discussed over and over. Search the archives, there are more thoughts and opinions there than you probably have time or desire to read. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.484 / Virus Database: 269.12.4/969 - Release Date: 23/08/2007 4:04 p.m. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.484 / Virus Database: 269.12.4/969 - Release Date: 23/08/2007 4:04 p.m. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to make this easier
exten = _*1XX,1,Set(CALLERID(all) = Nursery ${EXTEN:1})
exten = _*1XX,2,Dial(SIP/400)
Tom Vile wrote:
I have a need for a dialplan that call for the ability for people to
dial *1XX and it send a call
to extension 400 with the calleridname of Nursery and the calleridnum
of the *1XX number that
was put in minus the *. Now I know how to do it individually but I
now there must be an easier
way to simply the code.
Any help would be appreciated.
Tom Vile
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Re: [asterisk-users] Asterisk Load balancing
jk wrote: Hello, we are building an asterisk cluster. Here is what we are trying. Four Asterisk Servers AT1, AT2, AT3, AT4. Two service providers (SIP accounts). One for call origination (CO) and one for call termination. (CT) 1. Say some one dials a number 18xx xxx , CO forwards the call to our asterisk server, and the call goes to IVR, user select the option and call go to agent. 2. User wants to call out side, Asterisk server use CT trunk and terminate the call to the origination. Now my concern is that only one Asterisk can register with CO and CT at one time. Right? I want to do the load balancing so that OC server send call to one of the Asterisk server based on the load. How the user registered on AT1 can make call if AT4 is registered at CT. I have heard about Dundi and SER, but I am not if that is the right way to go. I get the idea how a user can register at different servers. But I am not getting how a CO knows where to send the call. Can anyone give me some lead to get around this load balancing issue? Thank you, -JK ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You are going to need to a gateway of some sort. that handles the registers and then will accept a call from the cluster then send it out the ITSP that you are registered to. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Copyright issues with libcurl and OpenSSL
Does anyone know why the g729 codec module sold by diguim does not display the OpenSSL copyright information. Do they have an agreement with OpenSSL to not display the Copyright Information that is required ny their license when distributed as part of a binary that uses OpenSSL. The registration program uses libcurl and openssl (both statically linked) to register the g.729 codec (and they said soon other products). This too does not display the required information about other peoples open source code. If they don't have already signed agreements with the developer of libCurl and OpenSSL they are distrabuting libCURL and OpenSSL with giving the correct copyright information with register program and then they are making money off the OpenSSL libs with they sell the g729 codec with out giving correct copyright information. I just want to confirm they have agreement not to show copyright information for the products, if not it could cause legal issues for people who work with Asterisk when they distribute it to their clients/customers. I have enclosed relivent snippites of the licenses in question and links to the full licenses. libCurl: http://curl.haxx.se/docs/copyright.html COPYRIGHT AND PERMISSION NOTICE Copyright (c) 1996 - 2006, Daniel Stenberg, [EMAIL PROTECTED]. All rights reserved. Permission to use, copy, modify, and distribute this software for any purpose with or without fee is hereby granted, provided that the above copyright notice and this permission notice appear in all copies. THE SOFTWARE IS PROVIDED AS IS, WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT OF THIRD PARTY RIGHTS. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. Except as contained in this notice, the name of a copyright holder shall not be used in advertising or otherwise to promote the sale, use or other dealings in this Software without prior written authorization of the copyright holder. OpenSSL: http://www.openssl.org/source/license.html /* * Copyright (c) 1998-2006 The OpenSSL Project. All rights reserved. * * Redistribution and use in source and binary forms, with or without * modification, are permitted provided that the following conditions * are met: * * 1. Redistributions of source code must retain the above copyright *notice, this list of conditions and the following disclaimer. * * 2. Redistributions in binary form must reproduce the above copyright *notice, this list of conditions and the following disclaimer in *the documentation and/or other materials provided with the *distribution. * * 3. All advertising materials mentioning features or use of this *software must display the following acknowledgment: *This product includes software developed by the OpenSSL Project *for use in the OpenSSL Toolkit. (http://www.openssl.org/) * * 4. The names OpenSSL Toolkit and OpenSSL Project must not be used to *endorse or promote products derived from this software without *prior written permission. For written permission, please contact *[EMAIL PROTECTED] * * 5. Products derived from this software may not be called OpenSSL *nor may OpenSSL appear in their names without prior written *permission of the OpenSSL Project. * * 6. Redistributions of any form whatsoever must retain the following *acknowledgment: *This product includes software developed by the OpenSSL Project *for use in the OpenSSL Toolkit (http://www.openssl.org/) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Copyright issues with libcurl and OpenSSL
but in the register program it is staticly linked. Matt Riddell (IT) wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 James Jones wrote: Does anyone know why the g729 codec module sold by diguim does not display the OpenSSL copyright information. Do they have an agreement with OpenSSL to not display the Copyright Information that is required ny their license when distributed as part of a binary that uses OpenSSL. The registration program uses libcurl and openssl (both statically linked) to register the g.729 codec (and they said soon other products). This too does not display the required information about other peoples open source code. Don't know about libcurl, but have you read the LICENCE file in the Asterisk directory? Specific permission is also granted to link Asterisk with OpenSSL and OpenH323. I assume if they have specific permission, then they've sorted something with the OpenSSL people. Best place to ask this though is: http://licensing.digium.com/main_page.php - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFCIe/S6d5vy0jeVcRAiT0AJ9LNMoQsqK+CkVoLbktqy/sQu+hAACffXnV 9AwB6HnOof0IO3FEtOPyhq4= =hjhw -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_zap.so stopped working after upgrading CentOS
Zeeshan Zakaria wrote: Yes, I did rebuild zaptel after upgrading the OS. And ended up rebuilding everything, i.e. libpri, zaptel and asterisk, doing make clean on all of them. But still the problem persists. I have to load CentOS using the old kernel to keep the things working. As for spinlock, that error is in its version 4, but I am running ver 3. So what should I do to build zaptel for the new kernel? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I would try that. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Moving to New Zealand
Is there anyon here currently in New Zealand that use asterisk, I need to help getting voice and internet services. I will be moving in a week. Any help would be great. Please use the details below to get ahold of me. Thanks in advance. James Jones Signate, LLC [EMAIL PROTECTED] 415.442.4012 (office) 413.771.1402 (office) 413.977.6482 (mobile) 413.667.3105 (fax) 665 Third Street Suite 100 San Francisco, CA 94107-190 Asterisk Services and Training ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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-Dev] Re: [Asterisk-Users] Help Solving Asterisk Lockups
If this issue exists doesn't it mean that asterisk is unstable anyway? On Sat, 2005-08-27 at 16:29 -0400, Marc Olivier Chouinard wrote: I have repeatedly mention this issues, and I keep getting laugh at from Mark... So I do not think donation to digium will fix the core problem. Digium want to sell the product like it is rightnow, and have no plan to do masive change to fix any core problems. They think that if they start redesign this, it will bring back asterisk to be unstable again. Marc O. James Jones wrote: I know of good way to solve this problem. I have been authorize by my company to try to a group of people and businesses to give donations to get Digium to fix this issue. We will start the pot at $200. Are there any takers? On Sat, 2005-08-27 at 10:08 -0400, [EMAIL PROTECTED] wrote: So the only thing we have in common is the remote monitoring ... Are you using: 1) Realtime (and if so, with mysql, odbc, etc?) 2) Logging CDR records? (and if so, how) This post looks like it could pertain to the same problem: http://lists.digium.com/pipermail/asterisk-dev/2005-August/014797.html .. but I don't think it has been resolved. Eric Julian [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Now I'm worried - we have exactly the same problem, but were going to upgrade to 1.2. Now it seems as if CVS-HEAD has the same issue. We have a TE405P, with 80 cisco7960 phones connected to a isdn30 pri. The same issues ocurr - Busy on inbound calls, cannot place outbound, nothing in the logs. Are you (as we are) 1) running with queues and agents We are *not* using queues or agents. 2) reloading the config (reload from the cli) I have used restart now from the cli to bring the system back when it freezes. Honestly I'm not sure that I've tried a plain reload. I'll see if that brings it back next time it dies. 3) monitoring the system by connecting to the manager cli ? We have an application (similar to the Flash Operator Panel) that connects to the manager API (via port 5038, not the CLI) and is used by our receptionist to monitor extensions and transfer calls. I intend to slowly start stripping the system down. Next time it crashes I will change the logging from mysql to csv only. This bug makes it sound like an mysql glitch can cause the system to hang: http://bugs.digium.com/view.php?id=4953 We are looking for all possible solutions to this. Me too! Eric ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto: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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto: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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto: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-use http://lists.digium.com/mailman/listinfo/asterisk-usersrs http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Help Solving Asterisk Lockups
I know of good way to solve this problem. I have been authorize by my company to try to a group of people and businesses to give donations to get Digium to fix this issue. We will start the pot at $200. Are there any takers? On Sat, 2005-08-27 at 10:08 -0400, [EMAIL PROTECTED] wrote: So the only thing we have in common is the remote monitoring ... Are you using: 1) Realtime (and if so, with mysql, odbc, etc?) 2) Logging CDR records? (and if so, how) This post looks like it could pertain to the same problem: http://lists.digium.com/pipermail/asterisk-dev/2005-August/014797.html .. but I don't think it has been resolved. Eric Julian [EMAIL PROTECTED] wrote: Now I'm worried - we have exactly the same problem, but were going to upgrade to 1.2. Now it seems as if CVS-HEAD has the same issue. We have a TE405P, with 80 cisco7960 phones connected to a isdn30 pri. The same issues ocurr - Busy on inbound calls, cannot place outbound, nothing in the logs. Are you (as we are) 1) running with queues and agents We are *not* using queues or agents. 2) reloading the config (reload from the cli) I have used restart now from the cli to bring the system back when it freezes. Honestly I'm not sure that I've tried a plain reload. I'll see if that brings it back next time it dies. 3) monitoring the system by connecting to the manager cli ? We have an application (similar to the Flash Operator Panel) that connects to the manager API (via port 5038, not the CLI) and is used by our receptionist to monitor extensions and transfer calls. I intend to slowly start stripping the system down. Next time it crashes I will change the logging from mysql to csv only. This bug makes it sound like an mysql glitch can cause the system to hang: http://bugs.digium.com/view.php?id=4953 We are looking for all possible solutions to this. Me too! Eric ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Broadvoice BYOD Plans - 3-way and Call Waiting
if you have anyone questions about your service you can contact us at the support 978-418-7300 James Jones Broadvoice Technical Support From: [EMAIL PROTECTED] on behalf of Ben Wern Sent: Sat 8/28/2004 4:34 PM To: Asterisk Users Subject: [Asterisk-Users] Broadvoice BYOD Plans - 3-way and Call Waiting Can anyone who is using Asterisk with Broadvoice tell of their experiences with 3-way calling and call waiting? I can't get Broadvoice to respond to my question, but I understand that there is a per minute fee (3.9 c/minute?) if you go over your use allowances. My question is, how are 3 way and call waiting calls handled? Because Asterisk would just handle them as two different channels/calls -- does Broadvoice allow BYOD customers to have two active lines and then start charging for a third? If so, does anyone have any configuration examples of limiting the number of sessions to a single provider? Ben Wern ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Broadvoice problem
you are correct. if you don't use sip.broadvoice.com it mess up you sip uri so the server will reject it. Also you should enable srvlookup it will help things run better. James Jones Broadvoice Technical Support From: [EMAIL PROTECTED] on behalf of Ed Brady Sent: Sat 8/28/2004 1:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Broadvoice problem Marty Mastera wrote: I had the same problem. To fix it, I had to do two things First: I had to update to CVS head, this was as per broadvoice support. Second: After updating, I had to change my sip.conf. Originally my sip.conf used hard coded ip addresses for broadvoice's IP servers, so I had to change the following lines as such: register = [mynumber]:[EMAIL PROTECTED] to read register = [mynumber]:[EMAIL PROTECTED] Ed, Weird things...I took your advice but executed it in stages...just like you, I was registering with 147.135.8.129, hardcoded ip. My CVS-HEAD is 7/14/04. The only thing I changed so far is to replace the 147.135.8.129 with sip.broadvoice.com. I didn't update from CVS, I also don't have SRV lookups enabled (yet anyway). It now registers and I can receive inbound calls. Does it make sense that BV may have implemented a change that would allow registrations from a FQDN but not from a hardcoded ip? Just a thought Marty ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Marty, Yeah, I agree it is pretty weird that Broadvoice would have made this change. When I called support they said that they had made some changes to coverup up some kind of security loop hole, however I am not clear how this would relate to this FQDN change. If nothing else, it caused me to (finally) update my system. BTW, does the latest CVS code have better support for SRV lookups? Ed ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Broadvoice User hung up on voicemail
don't quote me on this but I believe the earlier assumtion is correction. I think you need to have RTP going bothways otherwise the call will disconnect. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Shaw Sent: Friday, August 13, 2004 12:40 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Broadvoice User hung up on voicemail Maybe it has to do with RTP timing? * has to have sound coming in both directions in order to sync RTP... I notice when I'm leaving messages on my * Voicemail that I hear sharp clicks, the same thing happens using the record application... They sound like frame slips... Maybe when timing gets off a certain amount * just hangs up the call? -Chris - Original Message - From: Kevin [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 13, 2004 9:32 AM Subject: RE: [Asterisk-Users] Broadvoice User hung up on voicemail I place a call through Broadvoice to a phone and put it on mute(no noise) and it didn't get disconnected. -Original Message- From: Chris Shaw [mailto:[EMAIL PROTECTED] Sent: Friday, August 13, 2004 12:23 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Broadvoice User hung up on voicemail Ok, here's an update... I tried also just using the Record() application and the same thing seems to happen... It almost seems as if BroadVoice needs to have sound coming in both directions or it will disconnect... James if you're listening is this true? Is there some way we can get around this? I guess we could make the voicemail app introduce some noise maybe? -Chris - Original Message - From: Chris Shaw [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 13, 2004 8:46 AM Subject: Re: [Asterisk-Users] Broadvoice User hung up on voicemail If anyone else is having this problem (especially someone not using BV) please post, it'll help track down the problem. - Original Message - From: Chris Shaw [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 13, 2004 8:14 AM Subject: Re: [Asterisk-Users] Broadvoice User hung up on voicemail Not yet... I think it may have something to do with *'s voicemail silence detection... I tried turning down the sensitivity (so that it's more sensitive) to see if that helps, I'll let you know what I find... I suspect it's the voicemail app which is having this problem and not BV, otherwise * would hang up after 30 seconds or so and that's just not the case... I've kept my IVR running for like 5 minutes playing an MP3 and it worked just great! -Chris - Original Message - From: Kevin [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, August 12, 2004 1:30 PM Subject: RE: [Asterisk-Users] Broadvoice User hung up on voicemail Were you able to resolve this issue? -Original Message- From: Chris Shaw [mailto:[EMAIL PROTECTED] Sent: Thursday, August 12, 2004 2:08 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Broadvoice User hung up on voicemail Yes I've noticed this as well... Also some sharp clicks during recording, possible frame slips... - Original Message - From: Kevin [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, August 12, 2004 11:00 AM Subject: [Asterisk-Users] Broadvoice User hung up on voicemail After a call is sent to voicemail on an inbound connection from Broadvoice, the call is hung up in the middle of recording a voice mail after about 30 or so seconds. I get an error User hung up. If I answer the call and not have it go to voicemail, the call will stay connected. This only seems to happen on the Broadvoice connection and voicemail. Is anyone experiencing this issue? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
[Asterisk-Users] New Zealand DIDs
Does anyone know where i can get DIDs in New Zealand. I am look for area code 06. -James --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] broadvoice/asterisk
i have found you need to have the insecure=yes or insecure=very in your broadvoice context to get it to work. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Shaw Sent: Wednesday, July 28, 2004 1:20 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] broadvoice/asterisk Is it still working now? I haven't been able to get it to work since last night... I all of a sudden lost registration and it refuses to register... all of their servers are UNREACHABLE... I know they're working though, I used X-Lite and it worked just fine... I do have SRV records turned on, but instead of using insecure=very, I created contexts for each of their machines... That worked up until last night when EVERYTHING stopped working... James if you're listening... heeep :) -Chris - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 28, 2004 8:31 AM Subject: RE: [Asterisk-Users] broadvoice/asterisk I've got srvlookup=yes, insecure=yes, and an entry in /etc/hosts for 147.135.8.128. Registration is fine, however if an incoming call (from broadvoice) arrives from 147.135.8.129, the call fails. So I added a sip.conf entry like: [sip-broadvoice] type=user context=from-broadvoice deny=0.0.0.0/0.0.0.0 permit=147.135.8.129/255.255.255.0 permit=147.135.0.129/255.255.255.0 which seems to correct the problems with broadvoice calls arriving from different broadvoice servers. Anyone see an issue with this approach, or, is there a better way to handle this? Rich Also make sure that you have insecure=yes in your friend/peer section of you sip.conf file. Sorry forgot to mention. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of James Jones Sent: Tuesday, July 27, 2004 3:20 AM To: Asterisk User (E-mail) Subject: [Asterisk-Users] broadvoice/asterisk Ok we have found a better solution. Put everthing back the way it was and make sure that you have this line in your general section of you sip.conf file: srvlookup=yes We have added a SRV entry in the correct place now. So everyrthing should go the correct servers. -james --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] broadvoice/asterisk
Ok we have found a better solution. Put everthing back the way it was and make sure that you have this line in your general section of you sip.conf file: srvlookup=yes We have added a SRV entry in the correct place now. So everyrthing should go the correct servers. -james --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] broadvoice/asterisk
you can also thank jeff pyle for this one. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of James Jones Sent: Tuesday, July 27, 2004 3:20 AM To: Asterisk User (E-mail) Subject: [Asterisk-Users] broadvoice/asterisk Ok we have found a better solution. Put everthing back the way it was and make sure that you have this line in your general section of you sip.conf file: srvlookup=yes We have added a SRV entry in the correct place now. So everyrthing should go the correct servers. -james --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] broadvoice/asterisk
no problem, I am advid asterisk user. I will do as much as I can help from my end. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Brian Roy Sent: Tuesday, July 27, 2004 9:03 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] broadvoice/asterisk James, Sorry, was reading your reply about Jeff and misquoted the name. Anyway, thanks again! -Brian On Tue, 27 Jul 2004 07:50:24 -0500, Brian Roy [EMAIL PROTECTED] wrote: Jeff, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Broadvoice problems again Attn: James
not sure I know is pinging does not work. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Wolfgang S. Rupprecht Sent: Monday, July 26, 2004 5:00 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Broadvoice problems again Attn: James [EMAIL PROTECTED] (James Jones) writes: you can not ping that address because ICMP is turned off. Do you mean *all* ICMP is turned off or just icmp-echo-request / icmp-echo-reply? -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] broadvoice/asterisk
Also make sure that you have insecure=yes in your friend/peer section of you sip.conf file. Sorry forgot to mention. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of James Jones Sent: Tuesday, July 27, 2004 3:20 AM To: Asterisk User (E-mail) Subject: [Asterisk-Users] broadvoice/asterisk Ok we have found a better solution. Put everthing back the way it was and make sure that you have this line in your general section of you sip.conf file: srvlookup=yes We have added a SRV entry in the correct place now. So everyrthing should go the correct servers. -james --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Feature question
Does asterisk support outbound proxies? -James --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Broadvoice problems again
Broadvoice is not down. He is how to get it working: edit your /etc/hosts file to include the following. sip.broadvoice.com 147.135.8.128 That should fix it. We are allowing asterisk to connect only the that server for time being, Due to the fact that asterisk does not support outbound proxies. This only a temporary fix. Please, if anyone knows how to get asterisk to use outbound proxies please post it here. -James. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Shaw Sent: Monday, July 26, 2004 1:17 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Broadvoice problems again - Original Message - From: Charlie Hedlin [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 26, 2004 9:59 AM Subject: Re: [Asterisk-Users] Broadvoice problems again Broadvoice isn't down, it works fine, just not with Asterisk, which for most of us (myself included) means they might as well be down, but I believe we all went to their service knowing this wasn't truely supported. I am starting to see why so many providers don't want to go down this path, it has the tremendous potential to make them look bad. BroadVoice IS down, it is NOT working... I have tried it like 2 minutes ago just using my GrandStream and also with X-Lite and both are not working... it refuses to register... and when I check the logs in X-Lite, it's showing 404 Not Found which means that the proxy is either not running or not receiving connections... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Broadvoice problems again
with the fix. insecure=very will not work. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jeff Pyle Sent: Monday, July 26, 2004 2:42 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice problems again What worked for me: Adding to /etc/hosts: sip.broadvoice.com 147.135.0.128 Changing in BV friend/peer in sip.conf: insecure=very You may also choose to add deny/allow ACLs since BV won't authenticate against us anymore. I agree -- When I went to BroadVoice, I told Peter outright that I know Asterisk isn't officially supported, and once I got the system to register, I'll leave them alone. Today I had a small billing issue to take care of, so I just added oh, by the way, any word on when Asterisk will be back up? -- I got an instant email-response and an hour later, one of my lines is working already. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Broadvoice problems again Attn: James
you can not ping that address because ICMP is turned off. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Deon Rodden Sent: Monday, July 26, 2004 2:22 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Broadvoice problems again Attn: James Greetings, C:\ping 147.135.8.129 Pinging 147.135.8.129 with 32 bytes of data: Request timed out. Request timed out. Request timed out. Request timed out. Ping statistics for 147.135.8.129: Packets: Sent = 4, Received = 0, Lost = 4 (100% loss), Approximate round trip times in milli-seconds: Minimum = 0ms, Maximum = 0ms, Average = 0ms C:\ping 147.135.0.129 Pinging 147.135.0.129 with 32 bytes of data: Reply from 147.135.3.6: Destination host unreachable. Reply from 147.135.3.6: Destination host unreachable. Reply from 147.135.3.6: Destination host unreachable. Reply from 147.135.3.6: Destination host unreachable. Ping statistics for 147.135.0.129: Packets: Sent = 4, Received = 4, Lost = 0 (0% loss), Approximate round trip times in milli-seconds: Minimum = 0ms, Maximum = 0ms, Average = 0ms C:\ The first one is probably a firewall, but the 147.135.0.129 indicates a larger problem. P.S - Can phones that do not support outbound proxy also register at 147.135.8.128 (not just Asterisk?) - Original Message - From: Wolfgang S. Rupprecht [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 26, 2004 1:57 PM Subject: Re: [Asterisk-Users] Broadvoice problems again [EMAIL PROTECTED] (Olle E. Johansson) writes: The easiest first-level hack would be to randomly choose on of the SRV records provided they have the same weight. One of the other posts mentioned their ATA that simply registered with all the addresses. I don't think it would be a big or difficult change to have asterisk register with all the addresses also. I'm not sure what the right thing for outgoing is, or if it is even possible to have asterisk try all the sip servers in parallel, and then blow off the ones that are late in replying. That sounds like a much more involved hack. (I'll try to hack the registration issue here and post some GPL-ed patches if I get it working.) -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Broadvoice problems again
will work on it -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Shaw Sent: Monday, July 26, 2004 1:54 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Broadvoice problems again - Original Message - From: James Jones [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 26, 2004 10:35 AM Subject: RE: [Asterisk-Users] Broadvoice problems again Broadvoice is not down. He is how to get it working: edit your /etc/hosts file to include the following. sip.broadvoice.com 147.135.8.128 That should fix it. We are allowing asterisk to connect only the that server for time being, Due to the fact that asterisk does not support outbound proxies. This only a temporary fix. Please, if anyone knows how to get asterisk to use outbound proxies please post it here. -James. ok did that, registration works now and outbound seems to work but DTMF is not being received by * anymore... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Parking call problem
I been having a issue with call parking. I can park calls from internal extensions. But call from the outside can not be parked. When I recieve call from the outside I press the # key and nothing happens. Does any one have any thoughts? P.S. I am allowing the to be transferable. James --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.709 / Virus Database: 465 - Release Date: 6/22/2004
[Asterisk-Users] no audio with sip
I can make call in
to the asterisk server listen to voice mail, and do the echo test. When make a
call I get no audio inbound or outbound. When making incoming call I can leave a
valid voice message, but when then extentions pick up again no audio inbound or
outbound.I am using Xten liteand Broadvoice. Below are the messages
from console when call is made and my sip.conf. Any
thoughts.
console
info:
-- Executing Dial("SIP/xlite2-725e", "SIP/[EMAIL PROTECTED]") in new
stack -- Called [EMAIL PROTECTED]
-- SIP/sip99413-6d20 is ringing -- SIP/sip99413-6d20
answered SIP/xlite2-725e -- Attempting native bridge of
SIP/xlite2-725e and SIP/sip99413-6d20 -- Attempting native
bridge of SIP/xlite2-725e and SIP/sip99413-6d20
sip.conf:
[general]disallow=allallow=ulawport=5060
; Port to bind
tobindaddr=0.0.0.0
; Address to bind SIP channel
toexternip=24.218.94.95localnet=192.168.2.0localmask=255.255.255.0context=default
; Default context for incoming
callsmaxexpirey=180defaultexpirey=160canreinvite=notos=reliabilitysrvlookup=yes
register =
4137711401:[EMAIL PROTECTED]/99413
[sip99413]secret=passwordusername=4137711401host=sip.broadvoice.comtype=friendnat=yescanreinvite=nodtmfmode=inbandfromuser=4137711401callerid=4137711401context=incomingfromdomain=sip.broadvoice.comqualify=yesdisallow=allallow=ulaw
[xlite2]type=friendusername=xlite2secret=passwordcallerid="outcast"
5678host=dynamicnat=yes
; X-Lite is behind a NAT
routercanreinvite=no
; Typically set to NO if behind
NATdisallow=allallow=ulaw
---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.700 / Virus Database: 457 - Release Date: 6/6/2004
RE: [Asterisk-Users] Broadvoice and DTMF
Does this work for every. If so I will add it to our knowledge base, so let me know. -James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jay Milk Sent: Monday, June 14, 2004 3:32 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF It's official, Greg figured it out. And you know what, it all makes sense now: The scope for the dtmfmode setting is the section. Since the [broadvoice] section is needed for outgoing calls only, the [general] section -- the one containing the register directives would have to be where you define the dtmfmode for incoming connection. How about -- [general] dtmfmode=inband register = usera:[EMAIL PROTECTED] dtmfmode=rfc2833 register = userb:[EMAIL PROTECTED] Would that work? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marty Mastera Sent: Sunday, June 13, 2004 5:39 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Greg, Per your suggestion, I added dtmfmode=inband to the general section of my sip.confthe other items you mentioned were already in sync with what I had. With that one change inbound DTMF to * IVR works! I will continue to play with it to flesh out it's reliability, but I was successfully able to navigate my IVR and log on to * VM. Thanks for the suggestion, I will followup with any interesting developments from my testing. Marty -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Hill Sent: Sunday, June 13, 2004 4:23 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF On Sat, 12 Jun 2004, Jay Milk wrote: Makes me think that the problem isn't with Broadvoice at all, but rather with Asterisk's DTMF recognition. I'm running CVS Head from late April. I'm running CVS-HEAD-06/06/04. I've spent a couple hours tinkering and taking notes on the dtmf issue this morning. I tried various combinations of rfc2833 and inband in my dtmfmode= statements in sip.conf and with each combination tried dialling out (xten softphone - * - BV - cell phone voicemail) and calling in (cell phone - BV - * - IVR) to test DTMF functionality (or brokenness). During each call, I used show channel in the CLI to see how * really thought the channel was configured. I think I finally came up with a setup where DTMF works. I'm hoping maybe some of you who have been struggling with this issue also will give it a try and tell the rest of us if it works in your config. sip.conf: [general] ... dtmfmode=inband [broadvoice] type=peer ... dtmfmode=inband [xtenphone] type=peer ... dtmfmode=rfc2833 I don't have any allow/disallow statements for any codecs, although I'm thinking about bringing those into my puzzle soon.. Anyway, with sip.conf set up as I described above, I placed a call: xten - * - BV - cell phone and the DTMF was passed through so that I could interact with the voicemail system. In this call, * indicated: xten - * channel: codec=GSM, dtmfmode=rfc2833 * - BV channel: codec=ULAW, dtmfmode=inband When I tried to set the xtenphone to use dtmfmode=inband (in sip.conf), * filled my console with Unable to process inband DTMF on 2 frames and I couldn't capture any info on the channel setup through the CLI. Removing the dtmfmode=inband statement under [general] didn't affect results. So then I placed another call: cell phone - BV - * I set up extensions.conf so that the incoming call from BV would go into an IVR I built for controlling xmms. I was able to enter the extension numbers to control the system. I don't have voicemail set up on this * box, so I couldn't test a call to that app. In this call * indicated: BV - * channel: codec=ULAW, dtmfmode=inband Removing the dtmfmode=inband statement under [general] DID affect results! With this statement commented out, * indicates: BV - * channel: codec=ULAW, dtmfmode=rfc2833 In this setup, DTMF broke and I couldn't control my xmms. So.. Jay, Michael, others.. if you try this config, let us know what results you find! Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.700 / Virus Database: 457 -
RE: [Asterisk-Users] Broadvoice and DTMF
Does anyone know how Asterisk detects the DTMF tone on a SIP connection. Is it through the audio, or is it through the SIP Info? James Jones Broadvoice Technical Support From: [EMAIL PROTECTED] on behalf of Jay Milk Sent: Sat 6/12/2004 4:52 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Makes me think that the problem isn't with Broadvoice at all, but rather with Asterisk's DTMF recognition. I'm running CVS Head from late April. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marty Mastera Sent: Saturday, June 12, 2004 2:11 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Jay: I hope this input is in some way helpful, if only to confirm your findings... I don't have a Sipura to play with, but I do have a X100P...I used my cell phone to call my BV number, then answered the call from my 7960 and transferred the call out the X100P to my cell providers voicemail access number...once the system answered, I was able to send DTMF to log on to my cell phone voicemail. As with before though, no inbound DTMF working to IVR in asterisk... This worked: Cell Phone -- BV -- SIP -- * -- 7960 --Transfer-- Zap -- Cell phone VM retrieval number Marty -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk Sent: Friday, June 11, 2004 6:19 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF I spoke too soon. However, it's getting even weirder now. Works: Landline --POTS- BV --SIP- * --SIP- Sipura --CABLE- Zap --PCI- * -- VoiceMailMain Doesn't work: Landline --POTS- BV --SIP- * -- VoiceMailMain I don't have any other FXS devices to test this with. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Broadvoice and DTMF
has anyone tried info? James Jones Broadvoice Technical Support From: [EMAIL PROTECTED] on behalf of usedcanon Sent: Sat 6/12/2004 9:45 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Depends how you have it configured. To the best of my knowledge asterisk supports, inband, info and RFC2833 Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of James Jones Sent: 12 June 2004 14:26 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Does anyone know how Asterisk detects the DTMF tone on a SIP connection. Is it through the audio, or is it through the SIP Info? James Jones Broadvoice Technical Support From: [EMAIL PROTECTED] on behalf of Jay Milk Sent: Sat 6/12/2004 4:52 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Makes me think that the problem isn't with Broadvoice at all, but rather with Asterisk's DTMF recognition. I'm running CVS Head from late April. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marty Mastera Sent: Saturday, June 12, 2004 2:11 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Jay: I hope this input is in some way helpful, if only to confirm your findings... I don't have a Sipura to play with, but I do have a X100P...I used my cell phone to call my BV number, then answered the call from my 7960 and transferred the call out the X100P to my cell providers voicemail access number...once the system answered, I was able to send DTMF to log on to my cell phone voicemail. As with before though, no inbound DTMF working to IVR in asterisk... This worked: Cell Phone -- BV -- SIP -- * -- 7960 --Transfer-- Zap -- Cell phone VM retrieval number Marty -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk Sent: Friday, June 11, 2004 6:19 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF I spoke too soon. However, it's getting even weirder now. Works: Landline --POTS- BV --SIP- * --SIP- Sipura --CABLE- Zap --PCI- * -- VoiceMailMain Doesn't work: Landline --POTS- BV --SIP- * -- VoiceMailMain I don't have any other FXS devices to test this with. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Broadvoice and DTMF
are you doing internal codec translation internally (i.e. g.711u to g.729)? James Jones Broadvoice Technical Support From: [EMAIL PROTECTED] on behalf of Michael Swan Sent: Sat 6/12/2004 12:23 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Hi, Yes, I have tried all three: inband, rfc2833 and info. No luck with any. Michael Swan Neon Software, Inc. At 10:31 AM 6/12/2004 -0400, you wrote: has anyone tried info? James Jones Broadvoice Technical Support From: [EMAIL PROTECTED] on behalf of usedcanon Sent: Sat 6/12/2004 9:45 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Depends how you have it configured. To the best of my knowledge asterisk supports, inband, info and RFC2833 Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of James Jones Sent: 12 June 2004 14:26 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Does anyone know how Asterisk detects the DTMF tone on a SIP connection. Is it through the audio, or is it through the SIP Info? James Jones Broadvoice Technical Support From: [EMAIL PROTECTED] on behalf of Jay Milk Sent: Sat 6/12/2004 4:52 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Makes me think that the problem isn't with Broadvoice at all, but rather with Asterisk's DTMF recognition. I'm running CVS Head from late April. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marty Mastera Sent: Saturday, June 12, 2004 2:11 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Jay: I hope this input is in some way helpful, if only to confirm your findings... I don't have a Sipura to play with, but I do have a X100P...I used my cell phone to call my BV number, then answered the call from my 7960 and transferred the call out the X100P to my cell providers voicemail access number...once the system answered, I was able to send DTMF to log on to my cell phone voicemail. As with before though, no inbound DTMF working to IVR in asterisk... This worked: Cell Phone -- BV -- SIP -- * -- 7960 --Transfer-- Zap -- Cell phone VM retrieval number Marty -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk Sent: Friday, June 11, 2004 6:19 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF I spoke too soon. However, it's getting even weirder now. Works: Landline --POTS- BV --SIP- * --SIP- Sipura --CABLE- Zap --PCI- * -- VoiceMailMain Doesn't work: Landline --POTS- BV --SIP- * -- VoiceMailMain I don't have any other FXS devices to test this with. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Broadvoice and DTMF
I understand there has been some issues sending DTMF tone through Broadvoice. Can some provide me with symptoms? --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.700 / Virus Database: 457 - Release Date: 6/6/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Cisco Auto Provisioning
Do you just need a default config file for sip and which cisco device? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Senad Jordanovic Sent: Friday, June 11, 2004 10:17 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco Auto Provisioning Hi, Anyone knows where can one get ptag.dat for sip. I do not fancy waiting weeks to get it from Cisco! Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.700 / Virus Database: 457 - Release Date: 6/6/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.700 / Virus Database: 457 - Release Date: 6/6/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Cisco Auto Provisioning
I believe that is correct. But I have not done much work with provisioning the 186. Sorry. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Senad Jordanovic Sent: Friday, June 11, 2004 11:27 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco Auto Provisioning James Jones wrote: Do you just need a default config file for sip and which Cisco device? Sure, I have default config file for Cisco ATA 186. However, the task is to create config files for each device and place them in TPTP root directory and apparently one needs cfgmfg and pdat files in order to create these config files. Is that your understanding as well? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.700 / Virus Database: 457 - Release Date: 6/6/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.700 / Virus Database: 457 - Release Date: 6/6/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Broadvoice and DTMF
I have noticed, tell if doesn't mean anything, but I able to call my BV number and hear myself pressing the touch tones. Which means the audio for the DTMF tone is being passed to BV line. Of coarse I could be wrong I am fairly new to this technology. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Greg Hill Sent: Friday, June 11, 2004 1:23 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Broadvoice and DTMF On Fri, 11 Jun 2004, Michael Swan wrote: At 09:06 AM 6/11/2004 -0400, you wrote: I understand there has been some issues sending DTMF tone through Broadvoice. Can some provide me with symptoms? BroadVoice seems to not pass DTMF to Asterisk on incoming calls. It does send DTMF on outgoing calls. I've tried all three dtmfmode values in sip.conf leaving the setting at inband which seems to work for outgoing. I think it started working for me last night (using dtmfmode=inband). I just set up a callme arrangement where I run a script on the server and it places a call to the number I specify. When I answer that call, I can dial an extension number and * connects me there. So that's technically an outbound call, but the dtmf is being sent inbound to my * box. I also found dtmf to be working outbound on an outbound call (I placed a call to my cell phone's voicemail and was able to log in). Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.700 / Virus Database: 457 - Release Date: 6/6/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.700 / Virus Database: 457 - Release Date: 6/6/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Broadvoice and DTMF
inband should be used when processing DTMF from a BV line. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jay Milk Sent: Friday, June 11, 2004 2:33 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Hello James, I got three BV lines running, thanks to Peter's help. I'm pretty sure this USED to work at least once, but it isn't working currently. Outbound calls on BV -- DTMF works FINE. Inbound calls from landline, other VOIP line or cell phone: - DTMF are audible (which would indicate inband processing), but they're not received by asterisk, such that you can't control voicemail or other functions. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Jones Sent: Friday, June 11, 2004 8:06 AM To: Asterisk User (E-mail) Subject: [Asterisk-Users] Broadvoice and DTMF I understand there has been some issues sending DTMF tone through Broadvoice. Can some provide me with symptoms? --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.700 / Virus Database: 457 - Release Date: 6/6/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.700 / Virus Database: 457 - Release Date: 6/6/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.700 / Virus Database: 457 - Release Date: 6/6/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Broadvoice conf
Can anyone send default configuration files to me forconnect to broadvoice. --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.700 / Virus Database: 457 - Release Date: 6/6/2004
[Asterisk-Users] Introduction
Greeting to all, Hello my is James Jones. I am one of the new Tech Support people at Broadvoice. I have signed up for the Asterisk mailing list to better understand some of our customer's need and to learn more about Asterisk and what it can do. I can help answer any general questions about our services. I have been a Linux user for more than 8 years now and I understand for the most part how the open source community works. I would to help by providing help when I can to the group. I would like to note any information you may receive through me from this list or my VoIP forum (http://www.outcast.ws) is not the official policy or fully supported by Broadvoice, INC. You can receive official support by contacting us at [EMAIL PROTECTED] or by calling 978-418-7300. I do this help with the progress of a tool which I find interesting and useful to the VoIP. -James --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.700 / Virus Database: 457 - Release Date: 6/6/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
