[asterisk-users] ChanSpy * and 1234# not working

Hi, I am using ChanSpy with Asterisk 1.2.12.1. My extensions.conf has the following lines for ChanSpy exten => 1234,1,ChanSpy(Agent) exten => 1234,2,Hangup When I dial 1234 I can listen to one agent talking, but nothing happens if I press * or another agent number followed by #. Also archives

Re: [asterisk-users] any possibility of Vonage Integration

The providers have in there minds that A residential will use less line time than business will use. Like it was said I guess they don't have teenage kids There is more usage on my residential line than there is on my business line. I Put 1800 Mins on the cellular and about 1000 on the VOIP (

Re: [asterisk-users] any possibility of Vonage Integration

On 12/7/06, Paul <[EMAIL PROTECTED]> wrote: Some things are clear and some things not so clear. I couldn't find anything where specific limits on minutes in or out are stated. I think they try to limit the number of accounts cancelled strictly for high minutes. Accumulate enough of those and a s

Re: [asterisk-users] Error compiling Eicon Diva from source

On Thu, 7 Dec 2006, Matt Arnilo S. Baluyos (Mailing Lists) wrote: > Hello everyone, > > I'm having some errors while compiling the source code for an Eicon > Diva BRI card. > > I am using Fedora Core 6 with a compiled from source Linux kernel > 2.6.19. I have downloaded the Eicon Diva source RPM

Re: [asterisk-users] any possibility of Vonage Integration

Lacy Moore - Aspendora wrote: > On 12/6/06, *Paul* <[EMAIL PROTECTED] > wrote: > > Time Bandit wrote: > > > The TV ads promote it as unlimited. If there are real cases where > residential subscribers did not get unlimited residential service for > the adve

Re: [asterisk-users] problem with asterisk-1.4+sip communicator

hi, i'm using same network for asterisk server ,PBX agent and aslo users ...like an office have 20 PCs.. and my present sip.conf nat=no.is i need to set it as yes or any suggestions.. regards, thiru On 12/7/06, ram <[EMAIL PROTECTED]> wrote: On 12/6/06, Thirumal Saminathan <[EMAIL PROTECTED]

Re: [asterisk-users] any possibility of Vonage Integration

On 12/6/06, Paul <[EMAIL PROTECTED]> wrote: Time Bandit wrote: The TV ads promote it as unlimited. If there are real cases where residential subscribers did not get unlimited residential service for the advertised price, why aren't any state attorney generals going after vonage? Vonage clea

Re: [asterisk-users] any possibility of Vonage Integration

Time Bandit wrote: >> $25/month will buy you close to 50 hours of urban SIP termination, >> it's down to half a cent in some of the big cities. Are you >> going to average 50 hours on the phone each month? Some people >> do, but most don't. (Otherwise Vonage could not even pretend it is >> g

Re: [asterisk-users] 200+ analog phones connected to FXS modules

Hi guys, We are using AudioCodes, but still looking an alternative a cheaper one for our expansion. We are currenly running 4x24-ports FXS VoIP Gateway with 2 Asterisk Server each server has Dual-Port Card interfaced with E1 PRI ISDN to PSTN and E1 MFC/R2 to PABX. Our E1 ISDNs to PSTN are alre

Re: [asterisk-users] Switching from FreeBSD to Linux - which distro?

On 12/6/06, John Novack <[EMAIL PROTECTED]> wrote: Go get the ISO's, and remember to INSTALL EVERYTHING, then you won't run into some "gotcha" down the road where there is some missing file that needs to be put who knows where. Wow! Are you sure about that?

RE: [asterisk-users] Is there any Asterisk controllable thermostat?

I wouldn't IVR this. I'd do it with extensions. You have areas, targets, and actions assigned to an extension digit. So if lights are appliance type 1, the bedroom is location 1, and the action to turn the target on is 1, so you dial extension 111 and perform an action based on the extension. Th

Re: [asterisk-users] any possibility of Vonage Integration

I found the link for Vonage Integration with Asterisk http://www.vonage-business-plus.com/ Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll F

Re: [asterisk-users] Is there any Asterisk controllable thermostat?

Doug Crompton wrote: and it works great. Now I have one more way to control X10 devices. I can even call my VM on the way home and turn on my lights or whatever before I get home. Doug I've started to play with writing some code using the Java FastAGI interface to connect to my home automatio

Re: [asterisk-users] Govarion.

Govarion is SLOW. Slow to respond, slow to ship. Send E-mails every day to both support and [EMAIL PROTECTED] Last response I received from Ben was timestamped at 2 AM, EST! Their card seems to work OK, but I had problems with their tor2 card, bought off eBay at a "make offer" price. Port #1 wou

Re: [asterisk-users] centos 4.4 + asterisk

No problem. I was just enquiring before starting to install. Varun On Wed, 2006-12-06 at 12:12 +0200, Dovid B wrote: > No problems here. What problems are you getting ? > > - Original Message - > From: "varun" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discu

RE: [asterisk-users] MWI across multiple servers

Been working fine for us so far. -Original Message- From: Andrew Joakimsen [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 06, 2006 6:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI across multiple servers How well would NFS work in

[asterisk-users] Echo problem with TDM440P and ADSL Line

Hello, I'm a newbie user of Asterisk, i'm sucessfully install it it's great but, i get some problems with echo in a adsl line. My system is a TDM440P with 3 FXO Ports and 1 FXS. Asterisk 1.2.13 Zaptel 1.2.11 Line 1 - Analog Line 2 - Analog with ADSL It's installed with two analog lines one of t

Re: [asterisk-users] MWI across multiple servers

How well would NFS work in this situation? On 12/6/06, Porier, Jeremy M. <[EMAIL PROTECTED]> wrote: We are about to deploy six Asterisk servers across the state with SIP phones at each site registering to their "local" server. However, we are centralizing voicemail at our main campus to enable

[asterisk-users] RE: 0002475: [patch] Allow app_directory to work with REALTIME

I tried the patch. The config.c.patch was ok, the app_directory patch failed: lab1:/usr/src/asterisk-1.2.9.1/apps# more app_directory.c.rej *** *** 426,437 ast_log(LOG_WARNING, "directory requires an argument (context[,dialcontext])\n"); return -1

[asterisk-users] 0002475: [patch] Allow app_directory to work with REALTIME

Hi All, I'm running 1.2.9.1 stable. I'm wondering has this patch been applied to stable release or is it still only in CVS. Will this file patch apply correctly to 1.2.9.1 stable? Which file do I patch? I'm guessing app_directory_realtime_1.6.1.patch

[asterisk-users] Error compiling Eicon Diva from source

Hello everyone, I'm having some errors while compiling the source code for an Eicon Diva BRI card. I am using Fedora Core 6 with a compiled from source Linux kernel 2.6.19. I have downloaded the Eicon Diva source RPM (divas4linux_EICON-106.12-1.i386.rpm) and installed it already. When I go to /

Re: [asterisk-users] Switching from FreeBSD to Linux - which distro?

Phil Finkler wrote: Does there seem to be a popular Linux distro folks use specifically for Asterisk? I’d like to move off of FreeBSD but I’m not too familiar with Linux distros. In particular, I’m looking for a free, stable, well supported distro that has a friendly community. Any advice

Re: [asterisk-users] Setting outgoing caller id on a zap channel for one sip extension only

Asterisk supports whats called context, using a context just for that phone you can set a different callerid, then use a default context for all the other phones. On 12/6/06, Ron McCarthy <[EMAIL PROTECTED]> wrote: Hi List, Ive got one extension/login that when they call out from that it needs

Re: [asterisk-users] any possibility of Vonage Integration

$25/month will buy you close to 50 hours of urban SIP termination, it's down to half a cent in some of the big cities. Are you going to average 50 hours on the phone each month? Some people do, but most don't. (Otherwise Vonage could not even pretend it is going to make money.) You don't ha

Re: [asterisk-users] Switching from FreeBSD to Linux - which distro?

SuSE works well On 12/5/06, Phil Finkler <[EMAIL PROTECTED]> wrote: Does there seem to be a popular Linux distro folks use specifically for Asterisk? I'd like to move off of FreeBSD but I'm not too familiar with Linux distros. In particular, I'm looking for a free, stable, well supported

RE: [asterisk-users] Auto dialing: .call file vs. manager interface

> The manager interface expects "Exten" NOT "Extension" argument header. Well honk my hooter! I had been using 'Extension' but since I always used the 's' extension I never noticed anything goofy until I tried a numeric extension. Thanks for the heads up. -MC ___

Re: [asterisk-users] Setting outgoing caller id on a zap channel for one sip extension only

I use the GoToIf: If the SIP phone is Extension 2501 and dials out (I am using the norstar 9 to dial out convention). BTW. ${PSTNOUT} is a global variable for "ZAP/G2". exten => _9X.,1,GoToIf($["${CALLERIDNUM}" = "2501"]?2:3) exten => _9X.,2,Set(CALLERID(num)=9195551212) exten => _9X.,3,NoOp(${C

[asterisk-users] Avoided initial deadlock asterisk v 1.2.12.1 SIP clients IAX2 termination.

Periodically (as in sometimes several times a day and sometimes never) I get A channel.c: voided initial deadlock for '0x82*', 10 retries! The * figure is different each time. When this happens an active call (in or out) is dropped. The setup is as follows: handset --SIP--> Asterisk 1.

[asterisk-users] Govarion.

Hello All, Someone have the govarion phone number? I am sending e-mails to Mr. Ben and he never repply me. I buyed a card last week and after that never more give a repply. Sorry for this off-topic. Jefferson ___ --Bandwidth and Colocation provided by

Re: [asterisk-users] Setting outgoing caller id on a zap channel for one sip extension only

The way I would try to solved this would be to have a different context for just that use. His outbound calls would set a personal caller id, and then make the outbound call. Everyone else would use the group context. Other than that, possibly a good Macro might take care of that. Rob Ron McCart

Re: [asterisk-users] Setting outgoing caller id on a zap channel for one sip extension only

Yeah, Bascailly lets say extension 2 places a outbound call, it needs to show that persons private DID name and number, and anyone else gets the global callerid name/number. I guess you do this via a if statement, im trying but having a hell of a time getting it to work! On 12/6/06, Rob Scha

Re: [asterisk-users] Switching from FreeBSD to Linux - which distro?

What do you think about Slackware? Slackware is nice if you want to use a full distro because they use a pretty much vanilla kernel. Additional benefits is very few services are started by default. I use CentOS on almost all general server setups for small businesses and find that ther

Re: [asterisk-users] any possibility of Vonage Integration

On Wed, Dec 06, 2006 at 08:13:00AM -0500, Paul wrote: > Also, I should have mentioned that many of these providers advertise > "business" plans on their website. How can anyone honestly advertise > phone, fax, email hosting, web hosting, etc. to the business community > without 24/7 support? I li

Re: [asterisk-users] Setting outgoing caller id on a zap channel for one sip extension only

Ron, By source #, i assume you mean you have something like a SIP phone on the network with the extension like 4455, and you want that to have a different caller id when you make outgoing calls, then the rest of the phones on your network (the rest would show a global company number). Based on wh

Re: [asterisk-users] problem with asterisk-1.4+sip communicator

On 12/6/06, Thirumal Saminathan <[EMAIL PROTECTED]> wrote: Hi all, Thanks for your reply, I'm using sip communicator(in java that is intergrated with one ERP ) and asterisk is interfaced with this. i'm able to make calls between pingtel and Voip user, and also i can able to make call from Sip co

Re: [asterisk-users] TE110P Out fine / In Fail

On 12/6/06, Klaverstyn, David C <[EMAIL PROTECTED]> wrote: I have just installed Asterisk wit a TE110P card. I have configured 30 channels which seems to be recognised by staff and zap show channels. I can make outbound calls with exceptional call quality but inbound (receiving) calls the c

Re: [asterisk-users] Problem loading unicall

http://soft-switch.org/downloads/spandsp/spandsp-0.0.3pre22.tgz http://soft-switch.org/downloads/unicall/unicall-0.0.3pre9/ http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.13.tar.gz -- Humberto Figuera - Using Linux 2.6.17 Usuario GNU/Linux 369709 Caracas - Venezuela GPG Key Fingerprint

Re: [asterisk-users] Rejecting a Call

Ray Jackson wrote: The problem is I do not want to answer the call but reject it during call setup with the appropriate 401 or 403 type response - similar to the way Busy() sends back a '486 Busy Here'. The problem is this causes the other end to Hunt to the next endpoint, whereas I want the o

Re: [asterisk-users] Setting outgoing caller id on a zap channel for one sip extension only

Hi Rob, Well see that would work great if I knew the numbers they would be calling, but all I know is the source number/phone, i have no clue who they will be calling. Any ideals now? I wish it was that easy! Thanks! On 12/6/06, Rob Schall <[EMAIL PROTECTED]> wrote: Ron, I believe you would

Re: [asterisk-users] Setting outgoing caller id on a zap channel for one sip extension only

Ron, I believe you would just want to edit your extensions.conf file so that the extension you want separate has its own rule set. exten => 4567,1,Set(CALLERID(all)=000-000-) exten => 4567,n,Dial(SIP/4567) all other calls would just fit in like: exten => _4.,1,Set(CALLERID(all)=111-000-

Re: [asterisk-users] Rejecting a Call

The problem is I do not want to answer the call but reject it during call setup with the appropriate 401 or 403 type response - similar to the way Busy() sends back a '486 Busy Here'. The problem is this causes the other end to Hunt to the next endpoint, whereas I want the other end to give up

RE: [asterisk-users] any possibility of Vonage Integration

On Wed, 2006-12-06 at 13:03 -0500, Vijay Gandhi wrote: > must say very nice & deep calcutaion Thank you. Did you test it for errors? There's also a factor of 6/6 (or whatever) billing vs Vonage $25/75 flat, which can save in generic bills. It might even save an average of abou

[asterisk-users] Setting outgoing caller id on a zap channel for one sip extension only

Hi List, Ive got one extension/login that when they call out from that it needs to show a different name/number, and then the rest of the phone will have a default one. Whats the best way to do this? I know it can be done, just cant figure out how! Ive looked around and seem to see no docs on it.

Re: [asterisk-users] Was Vonage, now Tech support heaven

At 05:43 AM 12/6/2006, you wrote: So going with anyone over someone else because of 24/7 support, you need to find out what kind of support you really get after hours. If you are just going to get someone that can take your call, tell you "why yes, that is a problem, here is your ticket number, t

[asterisk-users] (REPOST DUE TO NO ANSWER) translate.c:88 powerof: Powerof 0: No power?? / translate.c:133 ast_translator_build_path: No translator

Dec 2 17:45:19 WARNING[64722]: translate.c:88 powerof: Powerof 0: No power?? Dec 2 17:45:19 WARNING[64722]: translate.c:88 powerof: Powerof 0: No power?? Dec 2 17:45:19 WARNING[64722]: translate.c:133 ast_translator_build_path: No translator path from gsm to unknown Dec 2 17:45:19 WARNING[6472

RE: [asterisk-users] FW: G.726 on Asterisk 1.4.0

I downloaded this two days ago from digium ftp. It reports to be 1.4.0-beta2. I also have the g726nonstandard=yes on the sip.conf file. Now, do I need to modify or not the rtp.c that is on /main directory? I already checked, I don't have audio on g726, with both ports on g729, and with both por

RE: [asterisk-users] any possibility of Vonage Integration

must say very nice & deep calcutaion Regards Vijay Gandhi -Original Message- From: Matthew Rubenstein [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 06, 2006 12:29 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] any possibility of Vonage Integration On Wed

Re: [asterisk-users] FW: G.726 on Asterisk 1.4.0

Carlos Alperin wrote: Ok, With everything restore on rtp.c, still I have no audio however the call is not destroyed immediately as before. I'm going to put a second Granstream box, and findout if between two boxes this happen too. I cannot believe that we cannot do 2 g726 on the same box at o

Re: [asterisk-users] problem with asterisk - calls where both sidescannot hear each other

Singer, Assuming that you have no issues with firewalls in the path regarding the rtp ports, or hardware/firmware problems, take a look at this patch: http://www.sineapps.com/news.php?rssid=1019 Please take note if * does not receive rtp packets for any reason, it does not send either. Jorge

Re: [asterisk-users] MWI across multiple servers

I've posted the instructions and scripts on my blog for everyone to grab. This way I'm not sending random files to random people :) http://asterisk.mdaniel.net/?p=14 Let me know if I need to change anything. On Wed, 2006-12-06 at 10:12 -0700, David Thomas wrote: > Aaron, > > Could you please s

Re: [asterisk-users] any possibility of Vonage Integration

On Wed, 2006-12-06 at 05:41 -0700, [EMAIL PROTECTED] wrote: > Date: Wed, 6 Dec 2006 12:21:12 +0200 > From: "Dovid B" <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] any possibility of Vonage Integration > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Message-ID: <[E

Re: [asterisk-users] MaxRetries:10 - Problems Dialout Call files grabbing end of RetryTime

Hi List I use calls file with WaitTime and RetryTime I need to understand when last retry is reached, this because some other work have to start. Is there any variable containig it in the Channels? Regards Doug Lytle wrote: > Marco Mouta wrote: >> Hi all, >> >> I'm working with dialout call

Re: [asterisk-users] MWI across multiple servers

Aaron, Could you please send me the scripts as well. Thanks! David ___ --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] bristuff error: "received SETUP message for callthat is not a new call"

On Mon, Dec 04, 2006 at 08:13:43AM +0100, Koopmann, Jan-Peter wrote: > On Monday, November 27, 2006 10:23 AM Louis-David Mitterrand wrote: > > > Hello, > > > > With the following setup: > > > > - asterisk 1.2.13, > > - zaptel 1.2.10 > > - bristuff 0.3.0-PRE-1v > > - quadbri card, > > Have you

[asterisk-users] Detecting no answers and/or disconnected numbers

Hi, Using call files, is there a way to identify no answered calls from disconnected numbers (no longer in service). Both return the same value and so far I can not find a way to know one from the other. Thank you, Andre Courchesne ___ --Bandwidth

RE: [asterisk-users] MWI across multiple servers

Aaron, Yeah, could you please send me that script. Thanks, Jeremy Porier Senior Director of IST Colorado Christian University [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel Sent: Wednesday, December 06, 2006 9:44 AM To:

Re: [asterisk-users] MWI across multiple servers

We've got a setup similar to that. Depending on how you want to set it up, just use externnotify and a script that touches the msg.txt file in the user's vm directory on the other boxes. We're using ssh, you may choose to use a different method. It's an immediate MWI notification, and seems to w

[asterisk-users] Dec 6 09:47:52 NOTICE[3263]: chan_iax2.c:1619 iax2_destroy: Avoiding IAX destroy deadlock

Hello, I am getting this message on my asterisk CLI. Dec 6 11:30:24 WARNING[3233]: chan_iax2.c:6516 socket_read: Received trunked frame before first full voice frame after that this message "Dec 6 09:47:52 NOTICE[3263]: chan_iax2.c:1619 iax2_destroy: Avoiding IAX destroy deadlock" "Dec 6 09:4

RE : [asterisk-users] Re: RE : Re: Recommendation for FXO

Hi Marty, I have checked/played FXS ports behind Asterisk with success and checking now a new firmware for FXO one stage (normaly two stages). All this gateways have the same manager unit and parameters suite, looking like Cisco models. It's normaly easy to use if you have trained for Cisco (only

[asterisk-users] MWI across multiple servers

We are about to deploy six Asterisk servers across the state with SIP phones at each site registering to their "local" server. However, we are centralizing voicemail at our main campus to enable the transfer of voicemails between users regardless of site. It also simplifies our backup procedures

[asterisk-users] FW: G.726 on Asterisk 1.4.0

This is what I found today googling on the Web, also I post it now in order to save time to others: The G726-32 codec: * It has been determined that previous versions of Asterisk used the wrong codeword packing order for G726-32 data. This version supports both available packing orders, and can

Re: [asterisk-users] Agent autologoff dynamic queue members - Brain aches please help

Have you tried putting a Local channel for the dynamic agent and, after the dial() tries for 20 seconds, you perform a dynamic agent logoff. Not sure if this will cause deadlocks, removing a dynnamic agent while he's being called, but maybe worth trying. Just my two (euro)cents, l. On

[asterisk-users] Error in codec string '=audio 5004 RTP/SAVP 3'

Hello, I have a problem with a grandstream IP Phone. The SIP autentication is OK, but when try to call someone I get the message --> WARNING[14281] chan_sip.c: Error in codec string '=audio 5004 RTP/SAVP 3' I tried to change the CODECs (ulaw, alaw, GSM, etc), the result is always the same. Tr

RE: [asterisk-users] problem with asterisk - calls where both sidescannot hear each other

I have bindaddr=0.0.0.0 in my sip.conf; what my major problem is that it only happens 5-8% of the time.. On Wed, 2006-12-06 at 09:56 -0500, Ed Nuñez wrote: > If you use both the public and private interfaces for VoIP in the Asterisk > Server, make sure you don't specify one of them for the bindi

Re: [asterisk-users] Rejecting a Call

On Wed, Dec 06, 2006 at 10:25:29AM -0400, Arlen Nascimento wrote: > Maybe you want something like > > [undesireable] > exten => _x,1,Playback(forbidden) This requires Asterisk to pass audio to the other party. -- Tzafrir Cohen icq#16849755jabber:[EM

RE: [asterisk-users] problem with asterisk - calls where both sidescannot hear each other

Singer I would be interested to see the rest of your configuration pertaining to how you are recording the calls. I am having trouble with this part. Are you using monitor or MixMonitor from extensions.conf of are you using the queues.conf or agents.conf monitor ? Ed Nuñez IT/Telecom Engineer

RE: [asterisk-users] problem with asterisk - calls where both sidescannot hear each other

If you use both the public and private interfaces for VoIP in the Asterisk Server, make sure you don't specify one of them for the binding in sip.conf Example bindaddr=0.0.0.0 will allow SIP traffic on any of your interfaces. Ed Nuñez IT/Telecom Engineer 4037 Metric Drive Winter Park, FL

Re: [asterisk-users] any possibility of Vonage Integration

- Original Message - From: "Paul" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, December 06, 2006 3:13 PM Subject: Re: [asterisk-users] any possibility of Vonage Integration Paul wrote: Dovid B wrote: Vonage has 24/7 s

Re: [asterisk-users] any possibility of Vonage Integration

- Original Message - From: "Paul" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, December 06, 2006 2:43 PM Subject: Re: [asterisk-users] any possibility of Vonage Integration Dovid B wrote: Vonage has 24/7 support. When my DI

Re: [asterisk-users] Same issue, different way to ask.

On Wed, Dec 06, 2006 at 09:36:28AM -0500, Carlos Alperin wrote: > Since nobody answer my previous question You ask a different question in a different thread with a totally non descriptive subject. Follow-up on your original message in the original thread, please. -- Tzafrir Coh

RE: [asterisk-users] CAS DID 2way

Then the only way it is going to be to use an Asterisk box with a PRI card. If you choose a dual one, you can install a Channel Bank on the second PRI and feed the PBX with up to 23 Channels. I don't know how many channels do you need on the PBX. Regards, Carlos -Original Message- From:

RE: [asterisk-users] Rejecting a Call

In the old asterisk use to have an exapmle of fiance calls, so if the system knows the caller id of the caller and match, you allways get busy.. Try to get the example, but this is nothing to do with the kind of device SIP or IAX or MGCP. Regars, Carlos -Original Message- From: [EMAIL

[asterisk-users] Same issue, different way to ask.

Since nobody answer my previous question (It looks like g.726 is a bad word). I have this scenario: One box with Asterisk 1.4.0 beta 2 IAX to anothers Asterisk working properly. As an ATA I have only one Grandstream HT496. Two lines on the ATA 727 & 726. >From outside I can call any of

[asterisk-users] CAS DID 2way

Greetings, I have a customer with an old PBX which cannot accept a PRI. Has anyone tried/tested connecting a CAS T1 to provide 2way DID trunks to a pbx? Either directly to an * server or a gateway? Thanks Jerry ___ --Bandwidth and Colocation provid

Re: [asterisk-users] Rejecting a Call

Maybe you want something like [undesireable] exten => _x,1,Playback(forbidden) exten => _x,n,Hangup Regards On 12/6/06, Ray Jackson <[EMAIL PROTECTED]> wrote: All, Is there a way of rejecting a call using SIP in the Asterisk Dialplan? Essentially, I want to look at the called number

Re: [asterisk-users] Can not hear called party

Hi, I'm about to leave for work, but I will send you a email from work. We're having similar problems with our setup, and we're been working on it for two weeks now. I would like to share notes with you to see if we can find anything. thanks, Singer Wang On Wed, 2006-12-06 at 14:10 +0100, Thomas

[asterisk-users] iax/sip registering and real-time

Hi all, I configure an Asterisk with real-time sip and iax users/peers. I want to do an action when users register/unregister to asterisk, how to do that ? Thanks Greg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users m

[asterisk-users] MWI/realtime/openSer in 1.4

Hi, can somebody clear up the situation with SIP voicemail SUBSCRIBE and realtime SIP peers in 1.4 for me. From a lot of sketchy information about the new chan_sip in 1.4 I know that it implements rfc compliant SUBSCRIBE behaviour (with the subscribemwi=yes option?), but what about realtime peer

Re: [asterisk-users] for all Asterisk Users

Hi Uugan, Because of the numerous additions and changes that Trixbox makes to the system, you're better off posting this question to the forums on Trixbox.org. As I recall, there is a forum dedicated to H.323 there. Cheers, Alex On 12/6/06, Uuganbayar.B <[EMAIL PROTECTED]> wrote: I have inst

Re: [asterisk-users] Asterisk freezes when DNS not working: a BUG??

Hi Bob, thanks for reply. The problem is all PBX are not in the same LAN and every customer wants his/her own DNS. I think I'll use /etc/hosts but the problem still remain: Asterisk shouldn't freeze during reloadthe registration should be located in another process but I think that such a c

Re: [asterisk-users] Re: Nokia E60 problems

2006/12/4, Martin Joseph <[EMAIL PROTECTED]>: On 2006-12-04 04:10:36 -0800, "Giedrius Augys" <[EMAIL PROTECTED]> said: > > > Hi, > I am testing Nokia E60 with Asterisk. And I noticed that if another side > is busy, nokia is still calling (I hear alerting), it do not show that > another side i

[asterisk-users] Agent autologoff dynamic queue members - Brain aches please help

Hi list, Using Asterisk 1.2.10 I am getting seriously confused by Queues and Agents. So far I configured my queue and agents, had my agents login using agentcallback. Call enters queue agent gets a call, if agent doesn't answer after 20 seconds a flag is set in AstDB (thanks to: Leo A

Re: [asterisk-users] any possibility of Vonage Integration

On Dec 6, 2006, at 8:13 AM, Paul wrote: Also, I should have mentioned that many of these providers advertise "business" plans on their website. How can anyone honestly advertise phone, fax, email hosting, web hosting, etc. to the business community without 24/7 support? People should also keep

RE: [asterisk-users] Ping

ping Arjan Kroon From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marnus van Niekerk Sent: woensdag 6 december 2006 14:15 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Ping Sorry to do this but I sent a

Re: [asterisk-users] G.729E

It's a 11.8kbps codec that's supposed to improve the quality for non-voice signal etc. And, according to this http://www.voiceage.com/prodg729.php, it's more CPU intensive than G.729A. It's listed at 25MIPS vs 10MIPS for G.729A. Also needs more RAM compared to G.729A. I don't think it qualifies as

[asterisk-users] Ping

Sorry to do this but I sent a couple of posts and I do not see them or any replies to them. Could someone reply to this please. Tx Marnus -- "Opportunity is missed by most people because it is dressed in overalls and looks like work." Thomas Alva Edison - Inventor of 1093 patents, includin

Re: [asterisk-users] any possibility of Vonage Integration

Paul wrote: >Dovid B wrote: > > > >> >> >> >> >>>Vonage has 24/7 support. When my DID is out I don't want to wait until >>>Monday morning. >>> >>> >> >> >>Yup. They sure do. Support thay you can barely understand. Plus I >>would go with a provider that has no issues (of course every co

[asterisk-users] Can not hear called party

Hello, We have a problem on a recent asterisk install with Polycom 30x phones; Sometimes (can not reproduce or find the logic of the problem after one week one analysis), the called party (even incoming or outgoing call) can not hear the calling party, as other flow works (caller hears called)

Re: [asterisk-users] any possibility of Vonage Integration

Dovid B wrote: > > >> Vonage has 24/7 support. When my DID is out I don't want to wait until >> Monday morning. > > > > Yup. They sure do. Support thay you can barely understand. Plus I > would go with a provider that has no issues (of course every company > has some down time). Been with one DI

Re: [asterisk-users] G.729E

Michael Iedema wrote: Greetings list, Does anyone have any information (providers' support) about G.729E? Voip-info.org came up empty, the implementers guide from the ITU wants my credit card and the rest of the pages I found simply made a few comparisons between it and iLBC. From what I unders

Re: [asterisk-users] Sipura phone does not ring

Sorry for my delay in answer you. S0 is how pstn line is identified into spa3000. It means that an incoming call from S0 will be forwarded to "[EMAIL PROTECTED]" yor must configure in sip.conf an account for the sip pstn line and a context , in extensions.conf the same context with a pattern or nu

[asterisk-users] G.729E

Greetings list, Does anyone have any information (providers' support) about G.729E? Voip-info.org came up empty, the implementers guide from the ITU wants my credit card and the rest of the pages I found simply made a few comparisons between it and iLBC. From what I understand, the codec is supp

Re: [asterisk-users] Attended Transfer

I had problems with featuredigittimeout . It was too short and betwen digit and digit was happened a timeout. modify to featuredigittimeout = 1000 2006/12/5, Arlen Nascimento <[EMAIL PROTECTED]>: Dear List, I've been working with Asterisk CVS-v1-0-09 and i'm trying to enable attended transfer

Re: [asterisk-users] Asterisk freezes when DNS not working: a BUG??

Giorgio, You could set up a caching name server in your local network, use it as your primary DNS server and your ISP's as a secondary. This would cache your ITSP's address(es) locally limiting your reliance on your ISP. Bob... On Wed, 2006-12-06 at 10:43 +0100, Giorgio Incantalupo wrote: > Hi,

[asterisk-users] Problems with bridging data calls over Wildcard TE405P

Hi, I recently installed an Asterisk server at a customer's here in Germany, using a Wildcard TE405P - Ports 1 and 2 (group 1) are connected to the phone system (Deutsche Telekom), whereas ports 3 and 4 (group 2) connect to the existing HiPath 3500 PABX. Currently, the dialplan is doing nothing

Re: [asterisk-users] regcontext, NoOp extension vanishes when extension reload and doesn't come back

On 15:59, Tue 05 Dec 06, Watkins, Bradley wrote: > You should put all of the 2 thru n priorities in a separate context and > then include the regcontext into that. > > For example: > > Let's say regcontext = registrations > > And you have a SIP peer: > > [1234] > type=peer > ... > regexten=1234

Re: [asterisk-users] Asterisk freezes when DNS not working: a BUG??

Just do a lookup for the domain name and resolve it to the IP address Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * *

Re: [asterisk-users] any possibility of Vonage Integration

I dont believe that vonage lets you control that. - Original Message - From: Vijay Gandhi To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Tuesday, December 05, 2006 11:11 PM Subject: RE: [asterisk-users] any possibility of Vonage Integration Thanks for a

Re: [asterisk-users] any possibility of Vonage Integration

Vonage has 24/7 support. When my DID is out I don't want to wait until Monday morning. Yup. They sure do. Support thay you can barely understand. Plus I would go with a provider that has no issues (of course every company has some down time). Been with one DID provider for 2 1/2 years and h

Re: [asterisk-users] any possibility of Vonage Integration

Don't be fooled by the flat rates of the locked-box providers. The real rates are so low these days most people pay less paying per minute than paying a Vonage style flat rate. In addition people report if you start making really heavy usage of your Vonage flat rate so that they are losing mone

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