Re: [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.

Tilghman Lesher wrote: > On Friday 07 December 2007 20:12:12 Philip Prindeville wrote: > >> Darryl Dunkin wrote: >> >>> You can store most of the configurations in a database which may be more >>> accessable to you. >>> >>> Perl can also parse these configurations quickly enough if you know

[asterisk-users] Playback file and detect a key press

I would like to do the following: Play back a file, and during the playback be able to detect a DTMF tone that may be pressed. I do not want to interrupt the playing of the file, but when the file finishes I would like to be able to tell if a key was pressed and which key it was. Anyway to do th

Re: [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.

On Friday 07 December 2007 20:12:12 Philip Prindeville wrote: > Darryl Dunkin wrote: > > You can store most of the configurations in a database which may be more > > accessable to you. > > > > Perl can also parse these configurations quickly enough if you know how > > to use the input record sepera

Re: [asterisk-users] CDR Function in Hangup Channel

On Thu, 2007-12-06 at 12:54 -0800, Douglas Garstang wrote: > Ok, this is a little crazy... > > billsec and duration are 0, but disposition is ANSWERED. > Huh? > > h => { > NoOp(*** LEG B HANGUP ${CDR(duration)} ${CDR(billsec)} > ${CDR(disposition)}); > &AddCallLeg(${LEGB_SOURCE},$

Re: [asterisk-users] asterisk performance

Your 512k outbound bandwidth will tend to be the defining factor in call quality here. Does your connection only gets used for voip? Or is it shared with other uses? Can you use more compressed codecs? G729 will quadruple you call capacity. What sort of QoS and traffic shaping do you use? Note

Re: [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.

That's sort of my point: that you have to reinvent it, and it's easy to get wrong. Darryl Dunkin wrote: > You can store most of the configurations in a database which may be more > accessable to you. > > Perl can also parse these configurations quickly enough if you know how > to use the input

Re: [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.

You can store most of the configurations in a database which may be more accessable to you. Perl can also parse these configurations quickly enough if you know how to use the input record seperator ($/) properly. The only thing Asterisk will not store which you would probably need is the actual M

[asterisk-users] Using XML for configuration management, single-source-of-truth, etc.

I'm starting work on some provisioning tools to simplify plugging in and configuring hard SIP handsets and conference bridges (maybe eventually MPEG-4 PoE video cameras that speak SIP as well). Issue is that I'd like to glean as much information out of the configuration files... but don't want

Re: [asterisk-users] Print CALLERID in CLI during "pri debug "

Well my project is an experimental project at my university. I need to collect experiment results which could tag every isdn message to the callerid, so it is clear which message belongs to which callerid (as multiple calls could be going on at one time). Thanks Arpit On Dec 7, 2007 5:34 PM, [

Re: [asterisk-users] Print CALLERID in CLI during "pri debug "

What don't you tell us what you are ultimately trying to do. You want the callerid next to the connect message in debug output... why? What will that help you to accomplish? On Dec 7, 2007 4:42 PM, Arpit Mehta <[EMAIL PROTECTED]> wrote: > Ok so the call reference is the 'cr' field (q931.c) and how

Re: [asterisk-users] Print CALLERID in CLI during "pri debug "

Ok so the call reference is the 'cr' field (q931.c) and how do I retrieve the caller id from this call reference ? On Dec 7, 2007 4:29 AM, Richard Revels <[EMAIL PROTECTED]> wrote: > > When the call sets up the 'call reference' is assigned. It will be unique > for the duration of the call and oth

Re: [asterisk-users] Polycom 601 stops ringing

>>> On 12/7/2007 at 2:33 PM, Doug <[EMAIL PROTECTED]> wrote: > At 10:58 12/7/2007, Joe Acquisto wrote: > >I have an odd issue, where a polycom 601 stops ringing, or more > >properly, maybe, stops *being* rung, when a call comes in. Other > >phones/extensions, continue to work fine, they being r

Re: [asterisk-users] Function vs. Application?

On Friday 07 December 2007 14:19:49 Jared Smith wrote: > On Fri, 2007-12-07 at 21:04 +0100, Vincent wrote: > > Out of curiosity, what's the difference between a function and an > > application? > > In a nutshell, an application is something that performs an action on a > channel (such as playing a

Re: [asterisk-users] Function vs. Application?

On Fri, 07 Dec 2007 15:19:49 -0500, Jared Smith <[EMAIL PROTECTED]> wrote: >Hopefully I've explained it in such a way that it's clearer to you know. >If not, let me know and I'll try to be more clear. Nope, good enough for me :-) Thanks. ___ --Bandwidt

[asterisk-users] Problem with the ring timeout in dial command for local extensions

Hi all, I don't know if this is the right list to ask, since I'm using Trixbox version 1.0.0.28, that has asterisk 1.2.17. I'm trying to configure the ring timeout value for my local extensions (when dialing from one to another), and the dial command simply ignores my values... I have one extensio

Re: [asterisk-users] Strange ISDN-problem with incoming calls out of the same city

On Fri, 7 Dec 2007, Stefan Guenther wrote: > Hi, > > >Does this number (you are dialing) has been ported from a different > >Telco? > > > > When you dial from the other city and you get "service not available" > >you may be dialing from a different Telco that either has no route > >aggreement for t

Re: [asterisk-users] "Happy Birthday Asterisk"

Tilghman Lesher wrote: > On Friday 07 December 2007 09:56:56 Bill Andersen wrote: > >> Philip Prindeville wrote: >> >>> So I'd venture to say that by August, the Internet will really be *30* >>> years old. >>> >> As Al Gore was born in 1948, I can see that the Internet could be as ol

Re: [asterisk-users] Function vs. Application?

On Fri, 2007-12-07 at 21:04 +0100, Vincent wrote: > Out of curiosity, what's the difference between a function and an > application? In a nutshell, an application is something that performs an action on a channel (such as playing a sound prompt, gathering DTMF input, putting the call into a call q

Re: [asterisk-users] "Happy Birthday Asterisk"

Bill Andersen wrote: > Philip Prindeville wrote: > >> So I'd venture to say that by August, the Internet will really be *30* >> years old. >> > > As Al Gore was born in 1948, I can see that the Internet could be as old > as 30, but not much more. 35 years ago would put him at 25 years old.

[asterisk-users] Function vs. Application?

Hello Out of curiosity, what's the difference between a function and an application? asterisk*CLI> core show functions asterisk*CLI> core show applications Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-user

Re: [asterisk-users] Polycom 601 stops ringing

At 10:58 12/7/2007, Joe Acquisto wrote: >I have an odd issue, where a polycom 601 stops ringing, or more >properly, maybe, stops *being* rung, when a call comes in. Other >phones/extensions, continue to work fine, they being run at the same time. > >My dial plan works fine (?) seems it will

Re: [asterisk-users] "Happy Birthday Asterisk"

On Friday 07 December 2007 09:56:56 Bill Andersen wrote: > Philip Prindeville wrote: > > So I'd venture to say that by August, the Internet will really be *30* > > years old. > > As Al Gore was born in 1948, I can see that the Internet could be as old > as 30, but not much more. 35 years ago would

Re: [asterisk-users] Strange ISDN-problem with incoming calls out of the same city

Hi, >Does this number (you are dialing) has been ported from a different >Telco? > > When you dial from the other city and you get "service not available" >you may be dialing from a different Telco that either has no route >aggreement for the dialed network, or the number portability data

Re: [asterisk-users] Limit participants in Meetme...

On Friday 07 December 2007 12:04:04 Carlos Chavez wrote: > Is there an easy way to limit the number of participants on a Meetme > room? Lets say we only want 10 people to be able to enter a particular > meetme conference, how can I prevent number 11 from entering this > conference? We will

[asterisk-users] Limit participants in Meetme...

Is there an easy way to limit the number of participants on a Meetme room? Lets say we only want 10 people to be able to enter a particular meetme conference, how can I prevent number 11 from entering this conference? We will not have a pin to enter. -- Telecomunicaciones Abiertas de Mé

Re: [asterisk-users] [Asterisk-users] Show calls in progress

In article <[EMAIL PROTECTED]>, Steve Johnson <[EMAIL PROTECTED]> wrote: > Is there an Asterisk CLI> command to show a list of calls in progress > (for all channels: Zap/SIP/IAX2 etc). > > "Restart when convenient" waits until the system is idle, but is there > an obvious way of seeing what's goin

Re: [asterisk-users] astunicall-1.2.21.0.1 packages and Sangoma A104D - ERROR

Josué, MFC/R2 signaling use pair of frequencies, not letters or numbers. For older packages of spandsp and libmfcr2 the letter E represent the last of this pair of frequencies. Your telco was asking for "F", because for the telco "F" is the last signal of the 15 signals used for MFC/R2. In newer p

[asterisk-users] Polycom 601 stops ringing

I have an odd issue, where a polycom 601 stops ringing, or more properly, maybe, stops *being* rung, when a call comes in. Other phones/extensions, continue to work fine, they being run at the same time. My dial plan works fine (?) seems it will ring properly, right after a reboot. It works

[asterisk-users] [Asterisk-users] Show calls in progress

Is there an Asterisk CLI> command to show a list of calls in progress (for all channels: Zap/SIP/IAX2 etc). "Restart when convenient" waits until the system is idle, but is there an obvious way of seeing what's going on at the moment? Thanks, Steve ___

Re: [asterisk-users] asterisk performance

2007/12/7, C F <[EMAIL PROTECTED]>: > by 3rd call do you mean over the internet? > if the answer is yes, then I wouldn't be surprised. Oh my god! If it is over internet and you get crap quality.. you have to be surprised.. It is depends by Latency (Traffic congestion, Network congestion) and Packe

[asterisk-users] AMQP Support for Asterisk?

Are there any plans to implement AMQP directly in Asterisk or is it best to use a third party bridge like Mule? https://jira.amqp.org/confluence/display/AMQP/Advanced+Message+Queuing+Protocol -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --Bandwidth

[asterisk-users] Open Asterisk Exchange Project

Is there anyone interested in developing an open source Asterisk / MS Exchange solution? Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] Attachment encrypted? click here .

[asterisk-users] Perspective on Asterisk

I’ve put out a project to bid that involves telephony and SMS. These technologies are not in my comfort zone so I’m trying to understand the landscape a little better. Several of the bidders plan to use Asterisk for the backend. From what I understand so far, that would apply to the telephony p

[asterisk-users] Sidetone with Snom 370

Hi all, I'm not getting any sidetone on my Snom 370. I searched the web and the snom wiki, but I don't see any place to enable/adjust it. Callers say I sound great on the other end, but I don't hear myself so it is a little off-putting. Any suggestions would be appreciated. On a related note,

Re: [asterisk-users] astunicall-1.2.21.0.1 packages and Sangoma A104D - ERROR

Hi Steve and Hi Moises, how are you? Greetings! :) would like to thank the you for always helping and to all this community. Steve already helped me some times in 2005, heheheh! I remade all the installation and now I used the following packages: asterisk-1.2.21.1, libpri-1.2.5, zaptel-1.2.19, libs

Re: [asterisk-users] "Happy Birthday Asterisk"

Philip Prindeville wrote: > So I'd venture to say that by August, the Internet will really be *30* > years old. As Al Gore was born in 1948, I can see that the Internet could be as old as 30, but not much more. 35 years ago would put him at 25 years old. And inventing the whole Internet at 25 is

Re: [asterisk-users] asterisk performance

by 3rd call do you mean over the internet? if the answer is yes, then I wouldn't be surprised. another thing what codec are you using? On 12/6/07, jorain <[EMAIL PROTECTED]> wrote: > Hi all, > > We are using > - a dell sc440(Single dual-core intel xeon 3040, 1.86GHz,1066MHz front size > bus 2MB ca

Re: [asterisk-users] PHP AGI script

On Thu, 6 Dec 2007, Nicholas Blasgen wrote: > I've got a very nice PHP AGI script but I want to be able to do some > database cleanup when the user hangs up the phone. I wish everyone would > hang up when they were suposed to, but some people don't. So what does > Asterisk send to an AGI file wh

Re: [asterisk-users] SIP-Realtime and sip reload

Am Donnerstag, den 06.12.2007, 21:06 +0100 schrieb Torbjörn Abrahamsson: > Our current approach is to use the #exec directive, and call a script which > creates static friends by reading information from the DB. We still use the > remote ITSP peers with realtime, as they do not need the OPTIONS. T

Re: [asterisk-users] "Happy Birthday Asterisk"

Philip Prindeville wrote: > [...] There were earlier > experimental versions of IP, but v4 got it right. > and v6 will get it even more right. ;) -- Jason Parker Digium ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asteri

Re: [asterisk-users] Any idea how making Asterisk "transparent"?

On Friday 07 December 2007 04:39:37 Artifex Maximus wrote: > On Dec 7, 2007 11:07 AM, Philipp Kempgen <[EMAIL PROTECTED]> wrote: > > Artifex Maximus wrote: > > > I am using Asterisk as transparent voice recorder for calls (isdn <-> > > > asterisk <-> pbx). Voice recording (therefore voice forwardin

Re: [asterisk-users] PHP AGI script

On Friday 07 December 2007 04:42:34 Andreas Brodmann wrote: > Nicholas Blasgen wrote: > > I've got a very nice PHP AGI script but I want to be able to do some > > database cleanup when the user hangs up the phone. I wish everyone would > > hang up when they were suposed to, but some people don't.

Re: [asterisk-users] Probems receiving 200ok message

No. My asterisk server had two NIC, one for public internet and another to LAN for phones. The problem is when I receive SIP 200 from public internet. Thanks. Fred Em Qui, 2007-12-06 às 21:53 -0500, C F escreveu: > is this machine or the phone behind nat? > > On 12/6/07, Frederico Madeira <[E

Re: [asterisk-users] Cisco power injector with GXP2000 phones

Hi, Am Donnerstag, den 06.12.2007, 11:30 -0500 schrieb Jon Pounder: > Quoting Ricardo Carvalho <[EMAIL PROTECTED]>: > > > I only see one explanation to my problem... > > > > GXP2000 phones only implement PoE mode A of the IEEE 802.3af protocol, and > > the power injector does only PoE mode B of t

Re: [asterisk-users] Any idea how making Asterisk "transparent"?

artifex, if you want call recording transparently, check out orecX.com they have a commercial and an open source SIP call recording package... no zap recording but if you are forwarding to sip exensions, you should be golden! saw them at VON 2007 boston... they have a recorded calls data

Re: [asterisk-users] Any idea how making Asterisk "transparent"?

Maybe a Patton Smartnode or similar would do the trick : ISDN SmartNode PBX | Asterisk I would be very curious to hear opinions from others on this. Regards ___ --Bandwidth and Colocation Provided by http://www.

Re: [asterisk-users] PHP AGI script

Nicholas Blasgen wrote: > I've got a very nice PHP AGI script but I want to be able to do some > database cleanup when the user hangs up the phone. I wish everyone would > hang up when they were suposed to, but some people don't. So what does > Asterisk send to an AGI file when the line has been

Re: [asterisk-users] Any idea how making Asterisk "transparent"?

On Dec 7, 2007 11:07 AM, Philipp Kempgen <[EMAIL PROTECTED]> wrote: > Artifex Maximus wrote: > > > Is Asterisk capable forwarding D-Channel and making Asterisk box > > totally transparent? > > No. Thanks Philipp. Bad news. We need recording calls with using nice functions like time synchro and tar

[asterisk-users] dtmf detection not working on sip trunks using asterisk-1.4.15

Hi all, I am using an asterisk-1.4.13 connected to our carrier via SIP trunk. I use rfc2833 as dtmf detection method. After upgrading to asterisk-1.4.15 our system would not detect dtmf from a caller from PSTN anymore. When investigating the SIP traffic at call initiation I realized that in the S

Re: [asterisk-users] Any idea how making Asterisk "transparent"?

Artifex Maximus wrote: > I am using Asterisk as transparent voice recorder for calls (isdn <-> > asterisk <-> pbx). Voice recording (therefore voice forwarding) is > working great but seems that Asterisk does not route/bridge/forward > D-Channel messages which means PBX cannot get time synchroniza

Re: [asterisk-users] PHP AGI script

Nicholas Blasgen wrote: > I've got a very nice PHP AGI script but I want to be able to do some > database cleanup when the user hangs up the phone. I wish everyone would > hang up when they were suposed to, but some people don't. So what does > Asterisk send to an AGI file when the line has been

Re: [asterisk-users] SIP-Realtime and sip reload

Torbjörn Abrahamsson <[EMAIL PROTECTED]> writes: > Our current approach is to use the #exec directive, and call a script which > creates static friends by reading information from the DB. Brilliant idea! That'll definitely be the replacement for our current realtime system. Thanks! /Benny _

[asterisk-users] Any idea how making Asterisk "transparent"?

Hello! I am using Asterisk as transparent voice recorder for calls (isdn <-> asterisk <-> pbx). Voice recording (therefore voice forwarding) is working great but seems that Asterisk does not route/bridge/forward D-Channel messages which means PBX cannot get time synchronization answer from provide

[asterisk-users] Pickup cmd

Hi all, I have a GXP2000 with BLF configured. I follow the configuration guide to enable the pickup cmd as follow and include it under corresponding content. [BLF_group_pickup] exten => _**1XX,1,Pickup(${EXTEN:2}) exten => _**1XX,n,Hangup The I press the single key to pickup the call to extens

[asterisk-users] asterisk performance

Hi all, We are using - a dell sc440(Single dual-core intel xeon 3040, 1.86GHz,1066MHz front size bus 2MB cache) as the asterisk server - dell 400sc(Intel P4) as a SER server - digium isdn card, TE120P at Asterisk server - Bandwidth: 2Mbps/512kbps All SIP Phones are registered to SER server, and

Re: [asterisk-users] 7960 Won't Register Yet Multiple Attempts?

Port 5060 should be udp as well. Sent from my iPhone On Dec 7, 2007, at 12:14 AM, <[EMAIL PROTECTED]> wrote: Hi List, I've got a 7960 that's behind NAT (nat_enabled: 1 and nat_received_processing: 1) and for whatever reason doesn't seem to register, or at least hold a registration. If both the