Re: [asterisk-users] Two PRI setup questions

On Thursday 01 November 2007 19:31:39 Lutgring, Sam wrote: > 2) Is there a way to see the idle status of a B channel? When AT&T tells > me they don't see the B channels coming up, is there a way that I can see > this in Asterisk??? Ask AT&T to turn off "B channel maintenance protocol" on the PRI

Re: [asterisk-users] zaptel.conf missing

just copy from SRC/zaptel.conf.sample On 11/2/07, kitti jaisong <[EMAIL PROTECTED]> wrote: > Hi all > I have installed zaptel on debian and missing file zaptel.conf in > /etc/zaptel.conf .my system don't have card TDM > please advice what the missing. > > thanks, > kitti > > >

[asterisk-users] zaptel.conf missing

Hi all I have installed zaptel on debian and missing file zaptel.conf in /etc/zaptel.conf .my system don't have card TDM please advice what the missing. thanks, kitti ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- ast

[asterisk-users] __sip_xmit problem

Hi, I got the following warning from CLI when I try to execute the Dial command. It makes the call failed. Anyone can tell me what does it mean and how to solve? -- Executing [EMAIL PROTECTED]:61] Dial("SIP/4009-1f178ba0", "SIP/[EMAIL PROTECTED]|35|L(7200:12)") in new stack --

Re: [asterisk-users] OpenSER for Asterisk Load balance

On 11/2/07, Edgar Guadamuz <[EMAIL PROTECTED]> wrote: > > Hi, guys > > I´ve just seen thta OpenSER can be coupled with Asterisk for load > balance, with the dispatcher module, something like this: > > dispatcher.cfg file > > # group sip addresses of your * units > 1 sip:10.1.2.3:5060 > 1 sip:10.1.

Re: [asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)

> > > What about: > > 1) Message waiting notifications? Especially in a distributed system > with multiple Asterisk servers? Openser with Asterisk real time integration can do this job for you 2) Different codecs for different SIP users/accounts? DTMF modes? I > know SER doesn't deal with the me

Re: [asterisk-users] Get value from linux terminal to dialplan in Asterisk ?

Arpit Mehta wrote: > I wanted to know a simple way in which I could read some file from a console > (say by using system command) and based on that either return true or false > back to dialplan. Is there any built in command in Asterisk for that ? AGI() In the AGI script do something like print

[asterisk-users] Get value from linux terminal to dialplan in Asterisk ?

Hello Asterisk Users, I wanted to know a simple way in which I could read some file from a console (say by using system command) and based on that either return true or false back to dialplan. Is there any built in command in Asterisk for that ? What are the options do I have ? Are there any samp

Re: [asterisk-users] Two PRI setup questions

Yep - as Doug mentioned, give esf framing and national switchtype a try. I have a PRI from AT&T in one of my offices, and use this setup. -erik On 11/1/07, Lutgring, Sam <[EMAIL PROTECTED]> wrote: > > > > I am in the process of implementing a new ISDN pri and have a couple of > questions. This

Re: [asterisk-users] Intercom with Snom phones

At 15:20 11/1/2007, Zaheer Master wrote: >I am currently waiting on a SIP Trunk to be set up for my company. In the >mean time, can I setup my phones on asterisk server so I can try the >intercom function? I'd just like to be able to talk on the local network to >test the sound quality. > >I'

Re: [asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)

> Some caveats (which may be different for OpenSER, so someone else can > chime in): What about: 1) Message waiting notifications? Especially in a distributed system with multiple Asterisk servers? 2) Different codecs for different SIP users/accounts? DTMF modes? I know SER doesn't deal with the

Re: [asterisk-users] Two PRI setup questions

Lutgring, Sam wrote: > > 1) What switchtype should be configured in the zapata.conf file when > AT&T is using CUSTOM? My understanding is that this equates to the > dms100 in Asterisk, is this right? The D channel is coming up just > fine, but AT&T tells me that they cannot see the B channels

[asterisk-users] Two PRI setup questions

I am in the process of implementing a new ISDN pri and have a couple of questions. This is a full 24 channels (23 B and 1 D) delivered over a T1 interface. The interface looks good and is not showing any errors. Any help that you can provide would be greatly appreciated. 1) What switchtype

Re: [asterisk-users] ring group containing external 10-digit numbers

Ryan Stille wrote: > I have a ring group setup that I'd like to ring a bunch of local > extensions, plus a few outside lines. I want recipients to > confirm the call by pressing 1 before they are connected. > But when I add in an external number to this ring group (such > as 5551212#), none

[asterisk-users] OpenSER for Asterisk Load balance

Hi, guys I´ve just seen thta OpenSER can be coupled with Asterisk for load balance, with the dispatcher module, something like this: dispatcher.cfg file # group sip addresses of your * units 1 sip:10.1.2.3:5060 1 sip:10.1.2.4:5060 1 sip:10.1.2.5:5060 the basic openser.cfg should be like: load

[asterisk-users] Polycom Park Button

Hi I have a Polycom 501 phone. I set the park feature to 1 in sip.cfg and the button shows up just fine. However when you press it it does nothing. I have the t and T in the dial string. Is there some trick to getting it to work with asterisk 1.4. Thanks Kelly __

[asterisk-users] ring group containing external 10-digit numbers

I have a ring group setup that I'd like to ring a bunch of local extensions, plus a few outside lines. I want recipients to confirm the call by pressing 1 before they are connected. But when I add in an external number to this ring group (such as 5551212#), none of my internal extensions ring

Re: [asterisk-users] Help: How does one determine the length of an outbound/dialout MESSAGE to be delivered

Costa Dinoteli wrote: > We created a dial plan which performs and outbound dial out and > deliveres a message to a receipient > What call method/option in extensions or anywhere allow us to > determine the length of the message. > IE, what if a 3 minute message is attempted to be delivered and > 4

[asterisk-users] Solved: Outgoing PRI CID?

Quoting "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]>: > Remember Caller*ID Number is either country code + area code + number or > area code + number. You never put a 1 or 0 at the beginning of the > number. CallerID Number also can not have spaces, dashes, or other crud. Darn! This last

Re: [asterisk-users] Outgoing PRI CID?

Quoting "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]>: > ;exten => s,n,Set(CALLERID(num)=${CALLERID(num)}) > > This is the same as saying Set(FRED=${FRED}) I know. I've been playing with setting the CALLERID(num), and this was my last, desperate attempt. > exten => _X.,1,Dial(IAX2/graham/${E

[asterisk-users] Help: How does one determine the length of an outbound/dialout MESSAGE to be delivered

Hi, We created a dial plan which performs and outbound dial out and deliveres a message to a receipient What call method/option in extensions or anywhere allow us to determine the length of the message. IE, what if a 3 minute message is attempted to be delivered and 45seconds into the call drops o

Re: [asterisk-users] PRI debuging shows 'Ext: 0' (Was: Outgoing PRI CID?)

The fact that you are sending "528" as your Caller*ID might be a problem for your carrier. Turbo Fredriksson wrote: > Quoting mail-lists <[EMAIL PROTECTED]>: > >> Turbo Fredriksson wrote: >>> We have now got our new PRI line (10 channels, 100 numbers) connected >>> and everything is working exce

[asterisk-users] PRI debuging shows 'Ext: 0' (Was: Outgoing PRI CID?)

Quoting mail-lists <[EMAIL PROTECTED]>: > Turbo Fredriksson wrote: >> We have now got our new PRI line (10 channels, 100 numbers) connected >> and everything is working except the outgoing caller ID. Whatever >> SIP phone I'm using, the CID that's shown is the very first number... Enabling PRI de

Re: [asterisk-users] Outgoing PRI CID?

It is up to your carrier to permit you to send different Caller*ID. Many carriers that let you send your own Caller*ID number (you can't send Caller*ID Name) but if you send an invalid number, they will drop the call or override the Caller*ID number with the primary number of your PRI. ;exten

Re: [asterisk-users] DST

Turbo Fredriksson wrote: > Quoting "Joe Acquisto" <[EMAIL PROTECTED]>: > > >> My thanks to all. Problem resolved with the assistance. >> > > Would be nice if you posted HOW it was fixed to... I have this exact > same problem at home, but the work phones displays time correctly... > > If

Re: [asterisk-users] DST

>>> On 11/1/2007 at 4:22 PM, Turbo Fredriksson <[EMAIL PROTECTED]> wrote: > Quoting "Joe Acquisto" <[EMAIL PROTECTED]>: > >> My thanks to all. Problem resolved with the assistance. > > Would be nice if you posted HOW it was fixed to... I have this exact > same problem at home, but the work pho

Re: [asterisk-users] AEL2 and Callbacks

Do a 'core show dialplan' and see what the AEL is generating. On 11/1/07, Douglas Garstang <[EMAIL PROTECTED]> wrote: > > >- Original Message > >From: Richard Lyman <[EMAIL PROTECTED]> > >To: Asterisk Users Mailing List - Non-Commercial Discussion < > asterisk-users@lists.digium.com> > >S

Re: [asterisk-users] DST

In case it helps, I've fixed that problem before by making sure my DHCP server gives out the correct time AND offset fields, and updating the Polycom firmware to a recent version :). Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (3

Re: [asterisk-users] DST

Quoting "Joe Acquisto" <[EMAIL PROTECTED]>: > My thanks to all. Problem resolved with the assistance. Would be nice if you posted HOW it was fixed to... I have this exact same problem at home, but the work phones displays time correctly... > joe a. > On 11/1/2007 at 1:43 PM, "Joe Acquisto"

[asterisk-users] Intercom with Snom phones

I am currently waiting on a SIP Trunk to be set up for my company. In the mean time, can I setup my phones on asterisk server so I can try the intercom function? I'd just like to be able to talk on the local network to test the sound quality. I've added this to my extensions.conf: [101] type=fr

Re: [asterisk-users] Outgoing PRI CID?

Quoting "Anciso, Roy" <[EMAIL PROTECTED]>: > I do this to tie extensions to a particular number: > > exten => _9X./_2XXX,1,SET(CALLERID(all)=Manistee ISD<2317231516>) > exten => _9X./_1XXX,1,SET(CALLERID(all)=MISD Tecnology<2317234264>) Tried that but couldn't get it to work. I've tried all the C

Re: [asterisk-users] DST

My thanks to all. Problem resolved with the assistance. joe a. >>> On 11/1/2007 at 1:43 PM, "Joe Acquisto" <[EMAIL PROTECTED]> wrote: > My Polycom phones are displaying time, off by one hour. Seems they are on > the old DST rules. How do I fix this? > > joe a. > > > ___

Re: [asterisk-users] Outgoing PRI CID?

> "mail-lists" == mail-lists <[EMAIL PROTECTED]> writes: mail-lists> I don't know if the same is true for you but we had to mail-lists> call our telco and have them set our callerid settings mail-lists> to 'station level'. Not sure if your telco offers this mail-lists> but the

[asterisk-users] Chanspy attaching to a caller ID entry?

Hi folks, We have a very rare problem with Asterisk 1.2 where Chanspy reports the following: Oct 31 19:53:29 NOTICE[10490] app_chanspy.c: Attaching SIP/105-48807010 to SIP/2015-11b21148 Oct 31 19:53:42 NOTICE[10490] app_chanspy.c: Attaching SIP/105-48807010 to SIP/2066-11d42b80 Oct 31 19:53:4

Re: [asterisk-users] Outgoing PRI CID?

I do this to tie extensions to a particular number: exten => _9X./_2XXX,1,SET(CALLERID(all)=Manistee ISD<2317231516>) exten => _9X./_1XXX,1,SET(CALLERID(all)=MISD Tecnology<2317234264>) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Turbo Fredriksson Sen

[asterisk-users] AsteriskNOW and TDM800P

Hi all I sold new TDM800P card with 8 FXO ports, someone know if can be use this card on AsteriskNOW or trixbox? What can i do for use this card? Thanks. -- RafaelCanchola Product Development Engineer, FonetGlobal Inc. [EMAIL PROTECTED] http://www.fonetglobal.com Ph. + 52 800 022 1

Re: [asterisk-users] Outgoing PRI CID?

Turbo Fredriksson wrote: > We have now got our new PRI line (10 channels, 100 numbers) connected > and everything is working except the outgoing caller ID. Whatever > SIP phone I'm using, the CID that's shown is the very first number... I don't know if the same is true for you but we had to call o

[asterisk-users] Outgoing PRI CID?

We have now got our new PRI line (10 channels, 100 numbers) connected and everything is working except the outgoing caller ID. Whatever SIP phone I'm using, the CID that's shown is the very first number... - s n i p - [default] include => outgoing include => priin [outgoing] exten => _NXX

Re: [asterisk-users] PRI commands missing...

On Thu, Nov 01, 2007 at 12:09:50PM -0500, Matthew Fredrickson wrote: > Arpit Mehta wrote: > > This happens when your pri line is down. This has happened a couple of times > > to me and the pri commands come back when the pri line is up. My guess is > > that your pri line is down. > > No. The pri

Re: [asterisk-users] PRI over T1 calls dropping, cause 100

Matt (& All), Thanks for the info below. You obviously responded to my bug report too (thanks again) - so I'll keep my dialog on the list to benefit other users too. >From your response below and on the bug site, I'm starting to get the picture. I know the firmware on the Nortel is old, so I'm g

Re: [asterisk-users] DST

Joe Acquisto wrote: > My Polycom phones are displaying time, off by one hour. Seems they are on > the old DST rules. How do I fix this? > > joe a. > > If you've got the files centrally managed, you can update the correct tags in sip.cfg to correct the situation. These are the "correct" s

Re: [asterisk-users] DST

Joe Acquisto wrote: > My Polycom phones are displaying time, off by one hour. Seems they are on > the old DST rules. How do I fix this? > > joe a. > > The archives are your friend: http://lists.digium.com/pipermail/asterisk-users/2007-March/181696.html -Dave ___

Re: [asterisk-users] DST

Joe Acquisto wrote: > My Polycom phones are displaying time, off by one hour. Seems they are on > the old DST rules. How do I fix this? http://www.google.com/search?q=polycom+dst+rules gives me some good results. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http

Re: [asterisk-users] DST

Joe Acquisto wrote: > My Polycom phones are displaying time, off by one hour. Seems they are on > the old DST rules. How do I fix this? Update the firmware to a recent release, or copy the DST rules section from the sip.cfg file from a recent release into your older version of sip.cfg. If you a

Re: [asterisk-users] AEL2 and Callbacks

>- Original Message >From: Richard Lyman <[EMAIL PROTECTED]> >To: Asterisk Users Mailing List - Non-Commercial Discussion > >Sent: Thursday, November 1, 2007 8:47:28 AM >Subject: Re: [asterisk-users] AEL2 and Callbacks > >Douglas Garstang wrote: >> I am originating a command via the AMI

Re: [asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)

Ser or openser with asterisk can be possible ram On 11/1/07, Antoine Megalla <[EMAIL PROTECTED]> wrote: > > Hi, > > I have a client who requires an Asterisk system with > 1500 SIP clients. > All clients will have ATAs (mostly Grandstream), so I > think a single > Asterisk server will not be able

[asterisk-users] DST

My Polycom phones are displaying time, off by one hour. Seems they are on the old DST rules. How do I fix this? joe a. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update opti

Re: [asterisk-users] issues with downloads.digium.com

Carlos Chavez wrote: > On Wed, 2007-10-31 at 15:26 +0200, Tzafrir Cohen wrote: >> On a slightly different matter: >> http://asterisk.org/downloads still points to zaptel 1.4.5.1 and libpri >> 1.4.1 . >> > > Yes, I noticed that too and was wondering if it is just because they > have not upda

Re: [asterisk-users] PRI commands missing...

Arpit Mehta wrote: > This happens when your pri line is down. This has happened a couple of times > to me and the pri commands come back when the pri line is up. My guess is > that your pri line is down. No. The pri commands should be there even if the span goes down. The reason why the pri com

Re: [asterisk-users] PRI over T1 calls dropping, cause 100

Michelle Dupuis wrote: > I have a T1 link from asterisk 1.2.23 (also tried with 1.4.13) to a Meridian > Option 61C. Calls either way drop with error "Channel 0/23, span 1 got > hangup, cause 100". Can anyone offer insight into the cause and > solution/workaround? (I tried upgrading to Ast 1.4.1

Re: [asterisk-users] Digium Vs sangoma Hradware

Michelle Dupuis wrote: > And just for confusion, one of the guys I work with swears by Sangoma. (I > have not done a lot of T1 stuff personally...so maybe as your expertise > grows Sangoma becomes a better fit). > > Perhaps I should have voted for Heinz... as for string -- go with fishing line.

[asterisk-users] libpri & tie line vs trunk

We are connecting an asterisk box to a Nortel Option 61 via a T1 with PRI. We have hit a problem we cannot overcome; specifically, the Nortel is asking for the ROSE information element (IE) over the PRI connection. This causes libpri to drop the connection, with cause 100. The Nortel cannot "turn

Re: [asterisk-users] Help

GXP2000 Firmware/sofware version is 1.1.4.18 Jarga -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris 'Xenon' Hanson Sent: Thursday, November 01, 2007 11:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users]

Re: [asterisk-users] Help

Jarga Jallow wrote: > I need help with my grand stream GXP2000 phones they keep freezing > randomly. Any ideas? What firmware revision? Want to buy a used one from me? I'm trying to standardize on Sipura 841s, and I have one GXP2000. > Jarga -- Chris 'Xenon' Hanson | Xenon @ 3D

[asterisk-users] RTP Read too short

Hello, I'm getting the following logs: [Nov 1 10:54:37] WARNING[20878]: rtp.c:1138 ast_rtp_read: RTP Read too short [Nov 1 10:54:39] WARNING[20878]: rtp.c:1138 ast_rtp_read: RTP Read too short [Nov 1 10:54:40] WARNING[20878]: rtp.c:1138 ast_rtp_read: RTP Read too short Anyone know how to corre

Re: [asterisk-users] AEL2 and Callbacks

Douglas Garstang wrote: > I am originating a command via the AMI with this... > > Action: Login > Username: xxx > Secret: yyy > > ACTION: Originate > Async: yes > Timeout: 6 > Exten: callback > Channel: Local/[EMAIL PROTECTED] > Callerid: 849120 > Context: default > ActionID: 849120 > > My LegA

[asterisk-users] Help

I need help with my grand stream GXP2000 phones they keep freezing randomly. Any ideas? Jarga <>___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http:/

Re: [asterisk-users] Connection astrisk to a RAS (portmaster)

Thanks, That's not it. I juste uninstalled wanpipe, redownloaded zaptel (another version (1.4.5.1) to try), re-installed wanpipe, (patching zaptel in the process), recompile wanpipe, re-compiled zaptel, recompiled asterisk, re-installed everything, and still got the same errors... Nicolas -

[asterisk-users] Parking speed

Hi Is it possible to speed up the parking process. We receive a lot of calls during the day and it gets to be painful to wait for the call to be parked and the number to be played back to you. I would like to speed up the whole process including the play back speed. Thanks Kelly__

Re: [asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)

While I can't help with OpenSER (OpenSER and SER are very different these days), the SER config info for SER 0.9.6 can be found in: http://siprouter.onsip.org/doc/gettingstarted/ That will also help you to understand the config methodology. The BASIC meshing Asterisk with SER is pretty straight

Re: [asterisk-users] Druid

Hi Dean, I am using Bicom Systems PBXware . In my opinion there is no asterisk GI that can compare to it. Juan Date: Wed, 31 Oct 2007 19:35:48 -0400 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: [asterisk-users] Druid Is anyone out the

Re: [asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)

You can use freeswitch for this kind of setup its working on asterisk technology asterisk + SER intergration URL http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+OpenSER Antoine Megalla <[EMAIL PROTECTED]> wrote: Hi, I have a client who requires an Asterisk syst

Re: [asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)

This is enough one to setup ser with asterisk http://www.voip-info.org/wiki/view/Asterisk+at+large Antoine Megalla <[EMAIL PROTECTED]> wrote: Hi, I have a client who requires an Asterisk system with 1500 SIP clients. All clients will have ATAs (mostly Grandstream), so I think a single Asterisk

Re: [asterisk-users] Spam Filter News

On Thu, 2007-11-01 at 08:05 +, Thomas Kenyon wrote: > Is there any news on getting the Spam Filter fixed for this mailing list? We've tweaked it quite a bit in the past few weeks, and it seems to be much better. If you're still experiencing issues, please send me a private email and I'll look

Re: [asterisk-users] Connection astrisk to a RAS (portmaster)

On Wed, 2007-10-31 at 20:57 -0400, Nicolas Ross wrote: > I also get sometime : > > == Primary D-Channel on span 2 down > [Oct 31 20:50:53] WARNING[10250]: chan_zap.c:2393 pri_find_dchan: No > D-channels available! Using Primary channel 48 as D-channel anyway! > == Primary D-Channel on span 2

Re: [asterisk-users] Mobile phone codecs ...

On Thu, Nov 01, 2007 at 01:09:24PM +0100, Benny Amorsen wrote: > > "AM" == Anselm Martin Hoffmeister <[EMAIL PROTECTED]> writes: > AM> Maybe the GSM codec is implanted to the "GSM chip" and that one > AM> does alaw, ulaw... > Also, modern handsets like the E90 rarely use the plain GSM codec. >

[asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)

Hi, I have a client who requires an Asterisk system with 1500 SIP clients.All clients will have ATAs (mostly Grandstream), so I think a single Asterisk server will not be able to handle all 1500 registrations, plus typical applications like Voicemail, call forwarding, etc.. and the billing need

[asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)

Hi, I have a client who requires an Asterisk system with 1500 SIP clients. All clients will have ATAs (mostly Grandstream), so I think a single Asterisk server will not be able to handle all 1500 registrations, plus typical applications like Voicemail, call forwarding, etc.. and the billing nee

Re: [asterisk-users] Mobile phone codecs ...

> "AM" == Anselm Martin Hoffmeister <[EMAIL PROTECTED]> writes: AM> Maybe the GSM codec is implanted to the "GSM chip" and that one AM> does alaw, ulaw... Also, modern handsets like the E90 rarely use the plain GSM codec. They use newer codecs such as EFR whenever possible. Asterisk probably

Re: [asterisk-users] SER with Asterisk intergration

satish patel wrote: > machine ?? i have asterisk with 200 SIP device but i voice > qulity and load of asterisk is bit high so i need to implement Why do you think you need SER? You must have something horrible on your set up # asterisk -rx "sip show peers"|tail -n 2 325 sip peer

Re: [asterisk-users] SER with Asterisk intergration

have you configuration file of SER to send request to asterisk i m not much familer with SER i have knowledge of asterik Arun Kumar <[EMAIL PROTECTED]> wrote: just configure SER on another port and use. On 11/1/07, satish patel <[EMAIL PROTECTED] > wrote:Dear all anybody have impleme

Re: [asterisk-users] Druid

Nice - I like where it's heading, finally someone came up with a half decent looking UI for a User Portal (though not perfect) I've consulted to 3 different companies about User Portals and not a single one has implemented anything near what it should look like. I just don't get why Asteris

[asterisk-users] Video Call

Hi.. Iam new with asterisk PBX, and i have read about asterisk video call.: my question: 1. Is imposible to establish system video call (from Phone with GPRS/3G enabled to Computer Running Softphone like X-Lite) over Asterisk Gateway.. 2. If posible what requirement (Hardware and Software on my A

Re: [asterisk-users] Call Failed

Robert La Ferla wrote: > After so many rings when the party does not answer, my SIP phone says > Call Failed. Why doesn't it just keep ringing? > > Here's the dial plan rule: > > exten => _NX,1,Dial(SIP/[EMAIL PROTECTED],,r) > exten => _NX,n,Hangup() > Not that it's the caus

[asterisk-users] Autodialing

hi,all I want make Autodialer in c++ using Asterisk Mangager Interfase; how to syncronize originate action i.e. at a time one call made and this time asterisk wait for some second to generate new call. thnks in advance. Bhrugu Mehta (SAI INFO SYSTEM) __

Re: [asterisk-users] Druid

Alex Epshteyn wrote: > Dean, > > > > If you are looking for a non-restricting and extensible Asterisk GUI > please look at Thirdlane http://www.thirdlane.com > . If you are comfortable installing OS, > Webmin and Asterisk, I would suggest installing PBX Manager GUI

Re: [asterisk-users] SER with Asterisk intergration

just configure SER on another port and use. On 11/1/07, satish patel <[EMAIL PROTECTED]> wrote: > > Dear all > > anybody have implement SER with Asterisk in single machine ?? i > have asterisk with 200 SIP device but i voice qulity and load of asterisk is > bit high so i need to implemen

Re: [asterisk-users] SparkLan WVPR-100 Wireless Handset.

Thomas Kenyon wrote: > Is there any news on getting the Spam Filter fixed for this mailing list? > It just figures that this should get through. Anyway, back to my original Post. Has anyone here used a SparkLan WVPR-100 Wireless Handset? http://www.sparklan.com/product_details.php?prod_id=4

[asterisk-users] Spam Filter News

Is there any news on getting the Spam Filter fixed for this mailing list? ___ --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/list

[asterisk-users] SER with Asterisk intergration

Dear all anybody have implement SER with Asterisk in single machine ?? i have asterisk with 200 SIP device but i voice qulity and load of asterisk is bit high so i need to implement SER for SIP registra and asterisk for feature Rgerads PGP Signature-- Satish Patel mobile:- +9