Re: [asterisk-users] best format for audio via asterisk...

On Sun, 10 Jun 2007, Matthew Pease wrote: Hi all - We are using voicepulse connect & asterisk together. We'd like to record our own outgoing messages, to be played back people that will be dialing into our voicepulse connect supplied DID. (am I getting this lingo right?) What is the best audi

Re: [asterisk-users] basic asterisk knowledge

On Sun, 10 Jun 2007, Khaled Chehab wrote: I have question concerns asterisk 1-What is difference between G.729 and G.729A? The letter A. http://en.wikipedia.org/wiki/G.729 ... says that G279A uses slightly less CPU to do the compression at the expense of sound quality. Digium appear to sup

RE: [asterisk-users] Bridged PRI calls - processor involvement?

This is the io wait figure from vmstat. If I run a vmstat 2 whilst I'm on a call I can see that the "wa" figure gets very high when the missing audio problem occurs. Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Fredrickson Sent: 08 June

RE: [asterisk-users] Bridged PRI calls - processor involvement?

Hi Steve, No, nothing like that, it has various updated from an 8mb Internet link and that's about it, I feel now that it's more down to disk I/O with the mpt driver than network. Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent:

RE: [asterisk-users] Bridged PRI calls - processor involvement?

On Mon, 11 Jun 2007, Steve Hanselman wrote: This is the io wait figure from vmstat. If I run a vmstat 2 whilst I'm on a call I can see that the "wa" figure gets very high when the missing audio problem occurs. I once looked after a Dell 2850 that exhibited some odd behaviour that I never got

RE: [asterisk-users] Bridged PRI calls - processor involvement?

I checked for BIOS upgrades the other week and there were none. I'm starting to suspect kernel changes as being the reason for this so I guess I'm going to have to remove some of the patchy disk activity to smooth the load and then start researching!!! Steve -Original Message- From: [EM

Re: [asterisk-users] No Audio with Gtalk

Dear Michael, I got the same problem for a long time, but noboday give me some tips. Do you solve it? Best regards, Charles 2007/4/1, Michael Zoller <[EMAIL PROTECTED]>: I configured my * with the instructions found here http://www.voip-info.org/wiki/view/Asterisk+Google+Talk to work with gt

[asterisk-users] MOH Problems.

All, I am using Asterisk 1.4.4 and it is not playing any MOH. I think the underlying problem is the following error: [Jun 11 20:25:37] WARNING[3207]: res_musiconhold.c:424 spawn_mp3: Found no files in '/var/lib/asterisk/moh/asterisk' [Jun 11 20:25:37] WARNING[3207]: res_musiconhold.c:506

Re: [asterisk-users] No Audio with Gtalk

Well I found out what the reason is. When the gtalk client is behind a NAT it will not work (at least for me it doesn't). Citing from : http://www.google.com/support/talk/bin/answer.py?answer=27930&query=udp&topic=&type= gtalk needs either UDP connections to anywhen on any port OR TCP Connectio

Re: [asterisk-users] MOH Problems.

On Monday 11 June 2007, Klaverstyn, David C wrote: > I think the underlying problem is the following error: You are right. You have to put some mp3 files to /var/lib/asterisk/moh/asterisk according to your configuration. regards t. -- knowledgeTools® ... managing complexity. --

[asterisk-users] CDR on transfers of called ZAP channel

Hello list, I have a problem with called ZAP channels making an attended-transfer or blind-xfer. Signalling at the phones is ok, but the CDR of Asterisk is wrong. At the moment there is a bristuffed Asterisk 1.2.18 running with bristuff-0.3.0-PRE-1y-g. Here is my dialplan, I simplified it a bit: [

[asterisk-users] Searchable List Archives?

I'd like to be able to search the list archives when I'm reading someone's message to put what they say in context based on what they've said, and what others have said in conversation with them, in the past. It would help me figure out whether to trust some submitters on some issues, and j

[asterisk-users] Help on text entry. using asterisk.

Hi, please help me in developing and reading "Text" through IVR application using asterisk. can any one help me at highlevel on this, other than using SPANDSP application. Regards K.Rajesh. _ Marriage? Get Detailed Profiles only

[asterisk-users] (no subject)

Hi, please help me in developing and reading "Text" through IVR application using asterisk. can any one help me at highlevel on this, other than using SPANDSP application. Regards K.Rajesh. _ Tried the new MSN Messenger? It’s co

Re: [asterisk-users] Searchable List Archives?

Matthew Rubenstein wrote: > Maybe Digium could upgrade the list SW, or let me do it for them. Or I > could set it up at my website, then import the list archive data and > parse it into my DB for a searchable mirror. Assuming google is indexing the list archives at http://lists.digium.com/piperma

[asterisk-users] change moh during a call?

Hello. Is it possible to change the defined moh sound file within an extension? I have: exten => 18,1,Answer exten => 18,n,Wait(3) exten => 18,n,SetMusicOnHold(durchwahl) exten => 18,n,Dial(SIP/118,15,m) exten => 18,n,Hangup Now i have the situation someone calls and my phone is ringing while m

Re: [asterisk-users] Searchable List Archives?

Per Jessen wrote: > Matthew Rubenstein wrote: > >> Maybe Digium could upgrade the list SW, or let me do it for them. Or >> I could set it up at my website, then import the list archive data >> and parse it into my DB for a searchable mirror. > > Assuming google is indexing the list archives at >

Re: [asterisk-users] Searchable List Archives?

On Mon, Jun 11, 2007 at 08:50:43AM -0400, Matthew Rubenstein wrote: > I'd like to be able to search the list archives when I'm reading > someone's message to put what they say in context based on what they've > said, and what others have said in conversation with them, in the past. > It would

Re: [asterisk-users] How to tell what codec is used for each end of a call MD110->H323->SIP

[EMAIL PROTECTED] wrote: Hi. Calling from Ericsson MD110 via H.323 trunk to an asterisk 1.4.4 I get the call established but no sound heard on either end. What is the best/correct way to try and see what codecs Asterisk is using on each end of the call as it passes through Asterisk fo

[asterisk-users] Different ECs in Asterisk

Does anybody have done some analysis on the different ECs come with Zaptel Driver? If so, can somebody post some summery? Thanks, Yonghua - Need a vacation? Get great deals to amazing places on Yahoo! Travel.

[asterisk-users] Asterisk as an SCCP client

Hi all, Has anyone tried using Asterisk as an SCCP client? My company has just signed up a 2 year agreement with M5 (fools!!) but are having intellectual issues with things like intra office phone calls and voice mail etc. They suddenly realized after M5 was installed that ALL their calls go out

RE: [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yieldinggains at high call volumes

Hi, The system seems to be IO bound for some reason. Reading at the older posts you mentioned that there is no significant disc activity so it could be ethernet i/o and/or interrupts that are causing this (old or insuficient ethernet driver maybe ?) Usually this kind of i/o wait is present on mach

Re: [asterisk-users] CDR on transfers of called ZAP channel

Gunnar-- CDR generation that covers transfers is an "umimplemented" feature in Asterisk, in any version. I have been working on a solution, but unfortunately, my solution is radical enough that I dare not apply it to 1.2 or even 1.4. It will most likely break every working implementation of billi

Re: [asterisk-users] IAX Peers show command

Ronaldo Z. Afonso wrote: Hi all, What does (T) mean on the output of "iax2 show peers"? The following my output. darkstar*CLI> iax2 show peers Name/UsernameHost Mask Port Status ronaldo (Unspecified) (D) 255.255.255.255 0

Re: [asterisk-users] IAX Peers show command

Hi Anthony, It doesn't make sense. This peer is an IAX peer. It was supposed to use UDP. Does Asterisk also use TCP for IAX? Thanks Ronaldo. Anthony Francis wrote: Ronaldo Z. Afonso wrote: Hi all, What does (T) mean on the output of "iax2 show peers"? The following my output. darkstar*CLI>

Re: [asterisk-users] IAX Peers show command

On Mon, 2007-06-11 at 09:59 -0600, Anthony Francis wrote: > T is for TCP, U would be UDP > ___ Actually T stands for TRUNK. IAX2 is always UDP. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +

Re: [asterisk-users] IAX Peers show command

Ronaldo Z. Afonso wrote: Hi all, What does (T) mean on the output of "iax2 show peers"? The following my output. darkstar*CLI> iax2 show peers Name/UsernameHost Mask Port Status ronaldo (Unspecified) (D) 255.255.255.255 0

Re: [asterisk-users] change moh during a call?

Add a r option with that extension so that way when it is called it will ring and then when you put it on hold it will play your moh. You can also set the MOH in sip.conf, but you will still get the same behavior you have now. Thomas Stein wrote: Hello. Is it possible to change the defined m

Re: [asterisk-users] IAX Peers show command

T = Trunking. If it's present then trunking is enabled. Ronaldo wrote: Hi Anthony, It doesn't make sense. This peer is an IAX peer. It was supposed to use UDP. Does Asterisk also use TCP for IAX? Thanks Ronaldo. -- Joshua Colp Software Developer Digium, Inc. __

RE: [asterisk-users] Calls being dropped

Where do I get oej's patch, and how do I install it? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas Brodmann Sent: Tuesday, June 05, 2007 2:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calls being dropped We ha

Re: [asterisk-users] CDR on transfers of called ZAP channel

On Mon, 2007-06-11 at 09:11 -0600, Steve Murphy wrote: > Gunnar-- > > CDR generation that covers transfers is an "umimplemented" feature in > Asterisk, in any version. > > I have been working on a solution, but unfortunately, my solution is > radical enough that I dare not apply it to 1.2 or even

Re: [asterisk-users] Console duplicate output problem

I guess he might mean don't include the -g on the command line? I'm wondering if asterisk is running in the background of the console you're logged in at, so it's dumping messages to the console, AND you've connected with -r? Moj Barton Fisher wrote: Eric "ManxPower" Wieling wrote: This i

Re: [asterisk-users] Console duplicate output problem

On MY distro (Mandrive) you edit /etc/lilo.conf and set the default kernel to "linux-nofb", then rerun lilo. You would have to find out how to disable the framebuffer on YOUR distro. If you use my method chances are your machine won't boot. Mojo with Horan & Company, LLC wrote: I guess he m

[asterisk-users] Grandstream 4104 - Asterisk (Incoming Calls problem)

Hello..I have a Grandstream 4104 (4 FXO) gateway connected to an Asterisk server and a traditional PBX..Asterisk users are able to call the PBX users but PBX users dont have access in Asterisk..Does anyone know if specific configurations in Asterisk and in Grandstream have to be done? Thanks ___

[asterisk-users] Multiple ENUM entries and Asterisk fails to dial

Hi, I use Asterisk 1.2.17 in my site, and now I'm trying to configure ENUM lookup in my server. When someone calls a number that has multiple ENUM entries, randomly Asterisk seems to fail to return a correct answer, and dial by ENUM fails. I've goggled a bit on this, but didn't get any

Re: [asterisk-users] change moh during a call?

"r" should never be used on the Dial line. Anthony Francis wrote: Add a r option with that extension so that way when it is called it will ring and then when you put it on hold it will play your moh. You can also set the MOH in sip.conf, but you will still get the same behavior you have now.

[asterisk-users] CallerID issues

All, I have run into some CallerID issues. It seems to have happened as a result of just moving my config from 1.2.12 to 1.4.4 (although I am not sure of this). Therefore I am sure its just a misconfiguration somewhere, I just don't know where. I have throughout the office either Cisco 79

Re: [asterisk-users] change moh during a call?

On 6/11/07, Thomas Stein <[EMAIL PROTECTED]> wrote: [snip] Now i have the situation someone calls and my phone is ringing while moh (durchwahl) is playing. When i pickup the call and press the hold button during the call moh (durchwahl) is playing as moh. Thats not what i want. Can i define anoth

[asterisk-users] Introduction to AGI programming

I wrote an introduction to AGI programming paper as an exercise to learn more about the process involved. You can find a copy of it here. I welcome any comments or corrections to improve upon it. As I said, it was mainly done to force myself to research the topic so th

Re: [asterisk-users] Polycom sip.cfg / voIpProt.SIP.requestValidation.x.request.y.event

Kenneth Padgett wrote: My company has pretty much standardized on Polycom phones and I am in the beginning phase of writing a GUI for administering/managing polycom provisioning at multiple sites which we intend to release as OS. I've I'd love to be notified when you release the Polycom admin

Re: [asterisk-users] agi with java?

Lenz wrote: Hi Lee, we are a Java shop and our experience with Java has been much the one you say - it does scale pretty well and it is very solid. What I was trying to say is that Java is not very well suited to the classic, Unix-style, fire-up-process-and-let-it-die that goes for CGI/AGI p

Re: [asterisk-users] Introduction to AGI programming

Kyle Sexton wrote: I wrote an introduction to AGI programming paper as an exercise to learn more about the process involved. You can find a copy of it here . I welcome any comments or corrections to improve upon it. As I said, it was mainly done to force myself to

[asterisk-users] AGI "RECORD FILE" for a video message

I have used the AGI call "RECORD FILE" before for an only audio message. I pretty much always used "RECORD FILE filename gsm" before. What paramater do I use for gsm so that if video is present it will record video also and still work for only audio. Will this recorded file (with video) play on

Re: [asterisk-users] Bad Echo between SIP calls

On Jun 8, 2007, at 6:00 PM, Deepak Naidu wrote: Hi,   We have a PRI connection & when its was on test networks we had echo problems withoutside line.  So I bought a TE212P card resolve the echo problem.  Which did to an extent. Its using asterisk 1.2.18 & RHEL4-Update 4. But now

Re: [asterisk-users] Multiple ENUM entries and Asterisk fails to dial

[EMAIL PROTECTED] wrote: > Hi, > > > I use Asterisk 1.2.17 in my site, and now I'm trying to configure ENUM > lookup in my server. > > When someone calls a number that has multiple ENUM entries, randomly > Asterisk seems to fail to return a correct answer, and dial by ENUM fails. > > > I've go

[asterisk-users] sip show registry shows nothing

Hi, In installed a new Trixbox-based system on Friday and initially had some problems with the system registering with our service provider (voip.co.uk), which I put down to the router's configuration. Anyway, the system was finally 'fixed' and worked well until Monday and then just stopped regist

Re: [asterisk-users] Bad Echo between SIP calls

Sounds crazy right? even was I, more over support guy logged in unloaded the zap modules to test them, still an echo. Ya, I was clear saying that we have SIP--- SIP issue ie internal extension echo problem. It seems the echo with SIP--SIP has many factors. I am just curios to eliminate any po

[asterisk-users] which Wifi SIP phones are the good ones

We are planning to buy a wifi SIP phones to work with Asterisk 1.2-18 setup. I would like to get feedback & views regarding Linksys WIP300 WIFI IP Phone or any other wifi phones which has been stable. Thanx for any updates. -- Deepak - The all

RE: [asterisk-users] Bad Echo between SIP calls

The echo cancellation card is for SIP->Zap calls only, no echo cancellation is done in Asterisk for SIP only calls. SIP to SIP, media is just passed through the server untouched (using media flow through, which is the option in sip.conf of canreinvite=no) if you are not handling any translation, e

Re: [asterisk-users] Bad Echo between SIP calls

Once upon a time I used to have a lot of SIP-SIP calls issues, which not always but sometimes included echo problems. There were no zap devices on the server. Googling and struggling to fix it, I found out that it was because of timing issues and ztdummy was not working properly. It had to do some

[asterisk-users] Slightly OT:CSU on Digium cards, and it's requirement

Do the Digium cards have a built in CSU? Is a CSU an FCC requirement? or just a carrier requirement? TIA ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digiu

Re: [asterisk-users] Bad Echo between SIP calls

Hey thanx for sharing your troubleshooting. Ya over days I kind of did some QA. There are SIP--SIP echo's between random phones. We have 75 phones of Polycom 501. I think might be the network or combination of network & polycom creating this. >>Do you have the backup of old setup without

Re: [asterisk-users] Slightly OT:CSU on Digium cards, and it's requirement

Quoting C F <[EMAIL PROTECTED]>: Do the Digium cards have a built in CSU? Is a CSU an FCC requirement? or just a carrier requirement? if you expect things to work you need one regardless of regulations, yes the digium cards have it built in, as do most modern t1 cards. if the "T1" terminat

Re: [asterisk-users] Slightly OT:CSU on Digium cards, and it's requirement

On 6/11/07, Jon Pounder <[EMAIL PROTECTED]> wrote: Quoting C F <[EMAIL PROTECTED]>: > Do the Digium cards have a built in CSU? > Is a CSU an FCC requirement? or just a carrier requirement? if you expect things to work you need one regardless of regulations, yes the digium cards have it built in

Re: [asterisk-users] Slightly OT:CSU on Digium cards, and it's requirement

On Mon, 11 Jun 2007, C F wrote: I disagree with this, I have several T1s that don't use Digium equipment and are directly connecting to T1 cards that DONT have a CSU and work fine. The reason this thing came up was because I was going thru documentation for such a card and it mentioned it's an

[asterisk-users] Crashes with Spandsp, app_rxfax.c, and asterisk 1.4.4

Hi everybody, I have a Fedora Core 4 x86 32 bit install, which I recently upgraded from asterisk 1.2 to the office 1.4.4 tarball. In the process of doing that I had to upgrade some autoconf/automake stuff, but it worked fine, and my new asterisk works fine. Except that anytime

Re: [asterisk-users] change moh during a call?

I think a real simple solution is to use a variable and you can change it when ever you want. So you can do exten => XXX,X,Musiconhold(${VARIABLE}) and then you can create an extension that will change the value of ${VARIABLE} something like: exten => 123,1,Answer exten => 123,2,Background(