[asterisk-users] How to notify that a message is waiting using RingTones with a Dahdi-connected analog phone ?

Hi, I'm wondering how I could notify to a dumb analog phone that a voicemail message is waiting. My goal would be to change the tone that is heard just before user starts to dial. Any idea on that ? Regards ___ -- Bandwidth and Colocation Provided by h

Re: [asterisk-users] asterisk-GUI

On Mon, Jul 06, 2009 at 11:19:49AM +0900, Tseveendorj wrote: > Hello, > > I've installed Asterisk 1.4.21.2 with Asterisk-GUI on Ubuntu 9.04 but > Asterisk. > > I couldn't see asterisk-GUI web interface when I accessed to > http://IPADDRESS:8088/asterisk/static/config/index.html. > But after cre

[asterisk-users] SIP registry fails during night

Every morning I check my SIP registry to the SIP-provider. And I must conclude that during the night somewhere registry has failed. asterisk*CLI> sip show registry HostUsername Refresh State Reg.Time 85.119.188.3:5060 092779077 105

Re: [asterisk-users] What is the best way to share extension state

writes: > 3) Hook into the AMI and parse the ExtensionStatusEvent which I think gives > me what I want. This is the "traditional" way, I believe. The challenge with AMI is that it is becoming a high-bandwidth channel. If you're only interested in one event type, you will spend quite a bit of CP

Re: [asterisk-users] SIP registry fails during night

On 6 Jul 2009, at 09:37, jonas kellens wrote: > What could be failing ?? Is this a NAT issue of some kind ? Could it > be that my firewall at one point blocks things off ??? > If it has something to do with NAT or firewall, why does a simple > 'sip reload' gets me registered again ?! Don't us

Re: [asterisk-users] SIP registry fails during night

Thanks for your reply, Steve. My firewall is a 3-NIC pc with Endian Community installed. Jonas. On Mon, 2009-07-06 at 09:49 +0100, Steve Howes wrote: > On 6 Jul 2009, at 09:37, jonas kellens wrote: > > What could be failing ?? Is this a NAT issue of some kind ? Could it > > be that my firewal

Re: [asterisk-users] SIP registry fails during night

Hi, Hmm. Not used that. Just out of interest, do you have 'qualify' on the peer in Asterisk? Could it be that its all a bit quiet and your router is breaking the stateful firewall stuff? Steve On 6 Jul 2009, at 10:05, jonas kellens wrote: > Thanks for your reply, Steve. > > My firewall is a

Re: [asterisk-users] Asterisk capacity

Hello, This is the configuration of my server got from PHP system info: *System Vital:* Kernel Version 2.6.18-6-amd64 (SMP) Distro Name Debian 4.0 *Hardware Information:* Processors 4 Model Intel(R) Xeon(R) CPU E5420 @ 2.50GHz CPU Speed 2.49 GHz Cache Size 6.00 MB System Bogomips 19954.78

[asterisk-users] asterisk and mISDN on Solaris

Hi, I read that installing asterisk on Solaris is supported. Does anyone of you actually have experiences with that? And especially, does anyone of you have experiences in runnning asterisk with misdn unter Solaris? Thanks and regards, Christophorus ___

Re: [asterisk-users] SIP registry fails during night

Hi Steve, I have qualify for all the peers that are defined, and so also for the peer I have defined for my SIP-provider. What you see below is such qualify : [Jul 6 11:38:26] Reliably Transmitting (NAT) to 85.119.188.3:5060: OPTIONS sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.2:5060;bra

[asterisk-users] Monitor

Hi All am using trixbox with call queues..I've set setinterfacevars=yes in queues.conf and following is dial plan : [test] exten => s,1,Answer() exten => s,2,Set(FILE_NAME=${CALLERID(num)}-${MEMBERINTERFACE}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}) exten => s,3,Monitor(wav,${FILE_NAME},m) exten =>

[asterisk-users] Iax trunk quality

Hi, I try to find a solution for this problem : [Jul 3 09:30:38] WARNING[3756]: chan_iax2.c:7312 socket_process: Received trunked frame before first full voice frame [Jul 3 09:30:38] WARNING[3757]: chan_iax2.c:7312 socket_process: Received trunked frame before first full voice frame [Jul

Re: [asterisk-users] Monitor

if you haven't exectued the queue cmd you cannot know who will took that call. You cannot know this before the agent took it because there are many agents who can do it. You can know it via cdr or manager interface, but only when the call is tooked or finished. On Mon, Jul 06, 2009 at 03:23:29PM

Re: [asterisk-users] Grandstream 2010 and blinky lights

Thanks for the info. We've managed to achieve or goal using 1.4 and a few hacks. 1) When the agent logs in / logs out, we rewrite the part of the dialplan for the hints and reload the dialplan 10 seconds after the *last* login / logout 2) For the MWI, we give each phone a "fake" voicemail (let'

Re: [asterisk-users] Grandstream 2010 and blinky lights

The quick answer is 'no'. It is not currently possible to monitor 'hints' for Agents - as an Agent never actually dials out (the device does). Even exten => 1234,hint,Agent/1234 won't work - as the 'core show hints' will show the agent as 'notinuse' when they can be. There are ways around it (I

Re: [asterisk-users] SIP registry fails during night

Registration timed out... This time a 'sip reload' doesn't help (why would it?). Registration uses no NAT, and a SIP option uses NAT... Verbosity is at least 25 [Jul 6 11:58:09] -- Remote UNIX connection [Jul 6 11:58:39] NOTICE[30045]: chan_sip.c:7683 sip_reg_timeout:-- Registration for

Re: [asterisk-users] Asterisk capacity

It can make 9977.39 Bogocalls of course! On Mon, Jul 6, 2009 at 5:17 AM, abdelkader wrote: > Hello, > > This is the configuration of my server got from PHP system info: > > System Vital: > Kernel Version 2.6.18-6-amd64 (SMP) > Distro Name  Debian 4.0 > > Hardware Information: > Processors 4 > Mode

Re: [asterisk-users] What is the best way to share extension state

Benny Amorsen schrieb: > writes: > >> 3) Hook into the AMI and parse the ExtensionStatusEvent which I think gives >> me what I want. > The challenge with AMI is that it is becoming a high-bandwidth channel. > If you're only interested in one event type, you will spend quite a bit > of CPU time j

Re: [asterisk-users] Asterisk capacity

You can handle 600 SIP sessions and about 400 calls doing transcoding ( passing RTP ) On Mon, Jul 6, 2009 at 1:26 PM, Steve Totaro wrote: > It can make 9977.39 Bogocalls of course! > > On Mon, Jul 6, 2009 at 5:17 AM, abdelkader > wrote: > > Hello, > > > > This is the configuration of my server g

Re: [asterisk-users] Asterisk capacity

Steve Totaro schrieb: > It can make 9977.39 Bogocalls of course! Mind to share the formula? Wait. Got it. Bogomips/2. Why on earth isn't that documented?! ;) > On Mon, Jul 6, 2009 at 5:17 AM, abdelkader wrote: >> Kernel Version 2.6.18-6-amd64 (SMP) >> Distro Name Debian 4.0 >> >> Hardware Infor

Re: [asterisk-users] What is the best way to share extension state

2009/7/6 > Greetings. > > I wonder what is the best way in your opinion to share real-time extension > state with applications outside of asterisk? What do you exactly mean by "applications" ? Do you mean a single server application or several instances of client applications ? > > > ... > >

Re: [asterisk-users] Queues recording & CDR

Hello, Just a correction, Asternic Call Center Stats is not from asteriskguru. Asteriskguru has its own statistic program that is not open source, but free to use. Asternic was written by me (not asteriskguru) and has an open source version and a commercial one. Best regards, -- Nicolás Gudiño B

Re: [asterisk-users] SIP registry fails during night

Hi! > Every morning I check my SIP registry to the SIP-provider. And I must > conclude that during the night somewhere registry has failed. > > I must do a 'sip reload' to get registered again. Can you ALWAYS solve this with a SIP RELOAD, or is it sometimes necessary to restart Asterisk? Anywa

Re: [asterisk-users] How to notify that a message is waiting using RingTones with a Dahdi-connected analog phone ?

Olivier wrote: > Hi, > > I'm wondering how I could notify to a dumb analog phone that a voicemail > message is waiting. > My goal would be to change the tone that is heard just before user starts to > dial. > > Any idea on that ? Yea, it's called stutter dial tone. For DAHDI channels just specif

Re: [asterisk-users] Problem configuring TDM400

jonas kellens wrote: > On Fri, 2009-07-03 at 11:58 +0100, Mike wrote: > >> tempest:~# lspci >> 00:00.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN >> interface > > I don't think this is you TDM-card... > > This is mine : > > 04:05.0 Ethernet controller: Digium, Inc. TDM400P (

[asterisk-users] Asterisk & Jabber : WARNING: res_jabber.c aji_recv_loop: JABBER: socket read error

I have installed gnutls and gnutls-devel from RedHat repositories [r...@asterisk asterisk]# yum install gnutls gnutls-devel I have installed iksemel with gnutls support : [r...@asterisk asterisk]# cd /usr/src/iksemel-1.3/ [r...@asterisk asterisk]# ./configure --with-gnutls --prefix=/usr [r...@ast

Re: [asterisk-users] Problem configuring TDM400

Mike wrote: > Folks, > > I have a Xen Asterisk VM with a TDM400 card. When I try to run > dahdi_cfg, I get: > > tempest:~# dahdi_cfg -vvv > DAHDI Tools Version - 2.2.0 > > DAHDI Version: 2.2.0 > Echo Canceller(s): > Configuration > == > > > Channel map: > > Channel 01: F

[asterisk-users] false answer on zaptel

Hi, I have an x100p zaptel card with asterisk 1.4. I'm using the system for outgoing calls. My problem is that Answer() is falsely returning while the call is still ringing and was not really answered yet. I've been digging google, wikis but have not found what might be causing this. SIP works fi

[asterisk-users] migrate from zaptel to dahdi

Over the weekend I tried to migrate a system from asterisk 1.4.25, libpri 1.4.1 ZAPTEL 1.4.12.1 to asterisk 1.4.25, libpri 1.4.7 (not 1.4.1) and DAHDI 2.2.0 I removed all old zaptel by: mv /etc/zaptel.conf /tmp mv /etc/asterisk/zapata.conf /tmp rm /etc/init.d/zaptel

Re: [asterisk-users] SIP IP-Trunk to be authenticated based on username and password, not IP address

And these mistery appear with Asterisk to Asterisk and does not appear between Asterisk to other products or from any IP Phone to Asterisk? How? Just because call came from Asterisk and was sent to Asterisk it is going to suffer this? While if it was originated from IP Phone then no problem? An

Re: [asterisk-users] migrate from zaptel to dahdi

On Jul 6, 2009, at 10:00 AM, Jerry Geis wrote: > Over the weekend I tried to migrate a system from > asterisk 1.4.25, libpri 1.4.1 ZAPTEL 1.4.12.1 > > to > asterisk 1.4.25, libpri 1.4.7 (not 1.4.1) and DAHDI 2.2.0 > > I removed all old zaptel by: >mv /etc/zaptel.conf /tmp >mv /etc

Re: [asterisk-users] migrate from zaptel to dahdi

Having gone through similar pains myself, I highly recommend going the the SVN asterisk 1.4 branch. I have had far fewer headaches following this path. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Niles Inga

Re: [asterisk-users] migrate from zaptel to dahdi

> > Jerry, > Check the dahdichanname setting in asterisk.conf. I had the same issue > myself - Niles > > Niles, I did a grep -i dahdichanname /etc/asterisk/* and no results. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital

Re: [asterisk-users] DTMF is not working occasionally over IAX Trunk

Hi all, Did some more digging in. I changed the trunk from IAX to SIP and still there are not much difference. So I guess it's not an IAX problem. I have enabled DTMF logging and captured the DTMF logs for two servers. (A: where E1 card is connected asterisk-1.4.25, dahdi-linux-2.1.0.4) and B (v1.

Re: [asterisk-users] Asterisk & Jabber : WARNING: res_jabber.c aji_recv_loop: JABBER: socket read error

Try client instead of component. Make sure that you selected the component in the menu select as well I can assure you that it works, and that it works well. We use it ;) Julian jonas kellens wrote: > I have installed gnutls and gnutls-devel from RedHat repositories > [r...@asterisk asterisk]# y

[asterisk-users] Small site survivability

We are currently moving away from a wide-spread Cisco CallManager deployment to Asterisk. For many of our small sites we have the routers configured for what Cisco calls SRST so if we have a WAN failure, the router acts as a SCCP registrar. We are converting to SIP, and from what I can tell Cisco

Re: [asterisk-users] Music on Hold

Julien Claassen wrote: > Hello! >I've configured Music on Hold in asterisk, the only, most certainly, > stupid > problem I have is, which DTMFs to send to activate and deactivate it. >If I use the cli, I can establish a call with originate. With the "misdn > send digit" command I can sen

Re: [asterisk-users] false answer on zaptel

Botond Botyanszki wrote: > Hi, > > I have an x100p zaptel card with asterisk 1.4. I'm using the system for > outgoing calls. > My problem is that Answer() is falsely returning while the call is still > ringing and was not really answered yet. I've been digging google, wikis > but have not found wh

Re: [asterisk-users] Dial cmd help

On Monday 06 July 2009 12:15:03 am Joseph L. Casale wrote: > >exten => s,n,ExecIf($["${ARG1}" = "1${ARG1:1}" > > ]?Set(Dialnum=${ARG1:1}):Set(Dialnum=${ARG1})) > > Much simpler Dhaval, thanks! Even simpler: exten => s,n,Set(Dialnum=${IF($["${ARG1:0:1}"="1"]?${ARG1:1}:${ARG1})}) -- Tilghman

Re: [asterisk-users] Small site survivability

Audiocodes supports SRST on their mediapack analog gateways. Cory J. Andrews Director New Market Initiatives Sayers Media Group VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE candr...@sayersmedia.com

Re: [asterisk-users] How to notify that a message is waiting using RingTones with a Dahdi-connected analog phone ?

On Monday 06 July 2009 08:18:37 am Dave Fullerton wrote: > Olivier wrote: > > Hi, > > > > I'm wondering how I could notify to a dumb analog phone that a voicemail > > message is waiting. > > My goal would be to change the tone that is heard just before user starts > > to dial. > > > > Any idea on t

[asterisk-users] Variable using AMI

Hello, if I do a variable assignation using AMI interface, that variable will be visible only for the current AMI instance or will be readable for all AMI instances?. I will login using the same user, concurrently. A program will write a global variable using the same name and if asterisk don't ha

Re: [asterisk-users] chan_mobile help.

Thanks for your response. I gave loads of info in my original mail, surely someone can help without jumping distro? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options vis

Re: [asterisk-users] dahdi_dummy configure

In article <20090703192211.gq25...@xorcom.com>, Tzafrir Cohen wrote: > On Fri, Jul 03, 2009 at 02:38:46PM -0400, Jerry Geis wrote: > > Description Alarms IRQ > > bpviol CRC4 > > DAHDI_DUMMY/1 (source: Linux26) 1UNCONFIGUR 0 > > 0

Re: [asterisk-users] chan_mobile help.

Try upgrading your bluez library. You can also try a parallel installation with the last revision of chan_mobile. Use the same phone always to discard any phone issues. On Mon, Jul 6, 2009 at 11:43 AM, Razza wrote: > Thanks for your response. I gave loads of info in my original mail, surely > s

Re: [asterisk-users] Asterisk capacity

Un-top-posting... On Mon, 6 Jul 2009, abdelkader wrote: > I need to know how many calls I can handle with my Asterisk. > Steve Totaro schrieb: >> It can make 9977.39 Bogocalls of course! On Mon, 6 Jul 2009, Philipp Kempgen wrote: > Mind to share the formula? Wait. Got it. Bogomips/2. Why on e

Re: [asterisk-users] Source for OpenVox cards?

In article , Timothy Legge wrote: > > I am looking for a source for an OpenVox card. Has anyone purchased through > http://www.voiplink.com or do you normally use another vendor or OpenVox.cn > directly? > > thanks > > Tim I have used voipon.co.uk, but I don't know whether that's useful to yo

Re: [asterisk-users] Small site survivability

On Mon, 6 Jul 2009, Jonathan Thurman wrote: We are currently moving away from a wide-spread Cisco CallManager deployment to Asterisk. For many of our small sites we have the routers configured for what Cisco calls SRST so if we have a WAN failure, the router acts as a SCCP registrar. We are co

[asterisk-users] Listed agents in queue not ringing

Hi All, I am having a problem when we call inbound the ivr picks up > send caller to the queue> but does not forward the calls to the listed agents, however if we use the call groups instead of queues it rings to the listed agents in group Here are the default settings include=DID_suhaib_tim

Re: [asterisk-users] Variable using AMI

On Mon, Jul 6, 2009 at 11:37 AM, Carlos Ruiz Diaz wrote: > Hello, > > if I do a variable assignation using AMI interface, that variable will be > visible only for the current AMI instance or will be readable for all AMI > instances?. I will login using the same user, concurrently. A program will >

Re: [asterisk-users] Music on Hold

Thanks Brent! I'll have a look there in features.conf. Warm regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: =

Re: [asterisk-users] chan_mobile help.

I'm running centos, so tried a yum upgrade but nothing was marked for upgrade. I've reinstalled bluez-libs.i386 0:3.7-1.1. I've tried a different dongle, but still get the same message. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.co

Re: [asterisk-users] migrate from zaptel to dahdi

On Mon, Jul 06, 2009 at 10:00:12AM -0400, Jerry Geis wrote: > After upgrading incoming calls seemed to work just fine. > Outgoing calls gave me an error 99 What is the outout of: cat /proc/dahdi/* Can you provide a trace of such a failed call? -- Tzafrir Cohen icq#16849755

Re: [asterisk-users] Asterisk & Jabber : WARNING: res_jabber.c aji_recv_loop: JABBER: socket read error

On Mon, 2009-07-06 at 16:18 +0100, Julian Lyndon-Smith wrote: > I can assure you that it works, and that it works well. We use it ;) My jabber.conf : [general] debug=yes ;;Turn on debugging by default. autoprune=no;;Auto remove users fro

Re: [asterisk-users] Variable using AMI

Maybe you could use the Asterisk Database. In 1.4 you can do it with DBGet and DBPut: http://www.voip-info.org/wiki/view/Asterisk+cmd+DBget http://www.voip-info.org/wiki/view/Asterisk+cmd+DBput In 1.6 use DB() function. Regards, Juan David Backeberg wrote: On Mon, Jul 6, 2009 at 11:37 AM, Carl

Re: [asterisk-users] Dial cmd help

>Even simpler: >exten => s,n,Set(Dialnum=${IF($["${ARG1:0:1}"="1"]?${ARG1:1}:${ARG1})}) Thanks Tilghman, I am making a note of this as well! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UN

Re: [asterisk-users] Asterisk & Jabber : WARNING: res_jabber.c aji_recv_loop: JABBER: socket read error

usetls=no Julian jonas kellens wrote: > On Mon, 2009-07-06 at 16:18 +0100, Julian Lyndon-Smith wrote: >> I can assure you that it works, and that it works well. We use it ;) > > My jabber.conf : > > [general] > debug=yes ;;Turn on debugging by default. > autoprune=

[asterisk-users] Get channel string

Hello, When I attempt to make a call using AMI interface with originate action I successfully specify all of the needed parameters but when I try to control the flow of the call I am unable to identify each call because asterisk uses some kind of unique identification appended to the channel stri

Re: [asterisk-users] Variable using AMI

Thank you! I did not know the existence of DB command. The command allows me to store KVPs but I have to use the same variable name every time so every process that starts the AMI instance will override the values making it unusable for what I want to achieve. It was really useful anyways. :)

Re: [asterisk-users] Get channel string

Carlos Ruiz Diaz schrieb: > When I attempt to make a call using AMI interface with originate action I > successfully specify all of the needed parameters but when I try to control > the flow of the call I am unable to identify each call because asterisk > uses some kind of unique identification ap

Re: [asterisk-users] Get channel string

No :( . Response gave me an empty unique-Id. Apparently it is generated on the fly once the resources are allocated or something else. I don't have any channel information in the response. On Mon, Jul 6, 2009 at 3:22 PM, Philipp Kempgen wrote: > Carlos Ruiz Diaz schrieb: > > When I attempt to m

Re: [asterisk-users] Variable using AMI

Well, I do not understand very well what you are trying to do, but I'll give you some advice: If you want a variable only for the AGI you call, you just have to declare that variable on the AGI. If you would like to make visible that variable as long as the call is active and for each call, ev

Re: [asterisk-users] Variable using AMI

I am sorry for my bad English. Apparently I'm explain myself wrongly but you got the point. I tried GetVar as AMI action but I have to specify a channel string. Of course I have the channel string, I parametrized it but Asterisk adds another string to the original channel and I can't obtain the va

Re: [asterisk-users] Some IAX calls do not disconnect.

Ah, and you are using iax trunking - which depends on the realtime clock. I'm no expert on virtualization, but I think I read that the usb based zaptel clock was a better choice in a virtualized system. T. On 6 Jul 2009, at 06:44, Rajkumar S wrote: Hi, The servers B & C are running in a

[asterisk-users] Bug or Not?

Hi gang, When I try to park a call using blind-transfer (#1), the caller hears the lot instead of the transferring party. Attended transfer and blind transfer from the phone buttons (Polycom 501) work fine, so this isn't a showstopper, just a "WHY??". Thanks for you attention.

Re: [asterisk-users] What is the best way to share extension state

http://bugs.digium.com/view.php?id=14595 has a patch to add a new class, bridge, so you get less events in AMI. This is for 1.6.0.x. It will give you an idea of what needs to be changed in order to make the call class of messages more granular. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http

[asterisk-users] Voicemail attachments not working

Today I discovered that voicemail attachments are not working on our latest asterisk server (version 1.4.24.1). I have two other asterisk servers that I maintain but I didn¹t do the configuration on these so this is my first time that I have done the voicemail.conf. I get an email but there is n

Re: [asterisk-users] Some IAX calls do not disconnect.

Just use SIP and solve all your problems. On Mon, Jul 6, 2009 at 5:00 PM, Tim Panton wrote: > Ah, and you are using iax trunking - which depends on the realtime clock. > > I'm no expert on virtualization, but I think I read that the usb based > zaptel clock > was a better choice in a virtualized s

Re: [asterisk-users] Bug or Not?

'One touch park' was designed to work around this issue. PaulH Danny Nicholas wrote: > > Hi gang, > > When I try to park a call using blind-transfer (#1), the caller hears > the lot instead of the transferring party. Attended transfer and blind > transfer from the phone buttons (Polycom 501) wo

Re: [asterisk-users] Some IAX calls do not disconnect.

- "Steve Totaro" wrote: > Just use SIP and solve all your problems. I seem to be noticing a common element to your posts about IAX. :-) I've been successfully using IAX in a large scale environment with no problems... yet. Can you shed some light on the reasoning behind your obvious disli

Re: [asterisk-users] Some IAX calls do not disconnect.

On Mon, Jul 6, 2009 at 9:41 PM, Tim Nelson wrote: > - "Steve Totaro" wrote: >> Just use SIP and solve all your problems. > > I seem to be noticing a common element to your posts about IAX. :-) > > I've been successfully using IAX in a large scale environment with no > problems... yet. Can you

Re: [asterisk-users] Some IAX calls do not disconnect.

On Tue, Jul 7, 2009 at 12:05 AM, Steve Totaro wrote: > On Mon, Jul 6, 2009 at 9:41 PM, Tim Nelson wrote: >> - "Steve Totaro" wrote: >>> Just use SIP and solve all your problems. >> >> I seem to be noticing a common element to your posts about IAX. :-) >> >> I've been successfully using IAX in

Re: [asterisk-users] Dail in modem

Umm.. they call that a BBS. :-) Sounds like the perfect application for FidoNet/Opus... ABBAS SHAKEEL wrote: > Sorry i replied late bcz i have to do some other work > I have a new required functionality. that is > Develop a Client server application that will communicate using a > normal modem

Re: [asterisk-users] How to notify that a message is waiting using RingTones with a Dahdi-connected analog phone ?

2009/7/6 Tilghman Lesher > On Monday 06 July 2009 08:18:37 am Dave Fullerton wrote: > > Olivier wrote: > > > Hi, > > > > > > I'm wondering how I could notify to a dumb analog phone that a > voicemail > > > message is waiting. > > > My goal would be to change the tone that is heard just before use

[asterisk-users] How to debug "Nothing to pick up" ?

Hi, General pickup doesn't seem to work here while directed pickup do. -- SIP/7530-08338f80 is ringing == Using SIP RTP CoS mark 5 [Jul 7 08:20:03] NOTICE[2299]: chan_sip.c:18383 handle_request_invite: Nothing to pick up for d61a727f746a9304 Upgrading debug level to 5, doesn't improve co