[Asterisk-Users] Asterisk increased memory

Not being a "c" developer, perhaps I am totally wrong, and therefore beg for your understanding ... ;) Is it normal for the * executable to be increasing in memory size ? I've noticed when using "top" that the * executable starts life like PID USER PRI NI SIZE RSS SHARE STAT %CPU %MEM

[Asterisk-Users] Echo cancelling with Adit 600

Do anyone have experience with echo cancelling on Adit 600? My Adit 600 consist of 5*8 FXS cards and 1 CMG Router using MGCP to Asterisk. I've turned on Echo Cancelling with 64ms as longest delay (that's maximum). But there still are great echo with delay when dialing through the telco (through an

RE: [Asterisk-Users] Line Noise UPDATE - If you've got line noise, read this

Ok, well I managed to fix the IRQ conflicts...well, sorta. I have two X100P cards in the system. One now has it's own interrupt and the other is sharing one with the soundcard. I tested outbound calls on both cards, still have the damn static. I am so sick of this. Is anyone else using X100P cards

[Asterisk-Users] How to attended/supervisor transfer

Hi all I don't know how to do an attended call transfer in asterisk. Iam using asterisk1.0.7 and oh323 client ( gnomemeeting ). I have to do any dialplan for that. can anyone help me to know. Thanks Varadhan ___ Asterisk-Users mailing list Ast

[Asterisk-Users] dialling problem with astcc

when a call comes on zap astcc.agi script launch and ask caller about his card number,and when the caller is dialing his card number(56170) sometimes astcc take it by missing a number as (5670) or doubled number as (556170) i dont know whats the problem is it from zap or is it from astcc.agi scri

Re: [Asterisk-Users] One touch voicemail on Cisco 7940/60

On Thu, 2005-04-21 at 21:36, Ron Wellsted wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Morris, Simon wrote: > > Hello, > > > > I'd like to program my Cisco phones to authenticate themselves to > > voicemail upon hitting the right button on my 7940/60's > > > > Ideally the voicema

[Asterisk-Users] Spandsp 0.0.2Pre15 with bristuff-0.2.0-RC8 Problem - Hangup

Hi ! After succesfully setting up a Server with an E1 Card -Asterisk CVS and Spandsp-0.0.2Pre10, I am having a problem getting the combination of bristuff-0.2.0-RC8.tar.gz and Spandsp-0.0.2pre15 to work on another machine. I have a trust HFC card I want to use. The problem described was identical

[Asterisk-Users] Re: TE110p - universal voltage?

In article <[EMAIL PROTECTED]>, Craig Guy <[EMAIL PROTECTED]> wrote: > Can anyone with a TE110p confirm that it will fit and work in both a 3.3 and > 5 volt pci slot? From photos it looks to be a universal card but the digium > literature makes no mention of voltage requirements. I can cofirm tha

Re: [Asterisk-Users] Provisioning lines 5 and 6 via TFTP

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Thu, 21 Apr 2005, Robert Goodyear wrote: Has anyone experienced a problem provisioning lines 5 and 6 of a Cisco 7960 via a SIPx.CNF over TFTP? What I'm experiencing is that regardless of the linex_... entries in the CNF file, lines 5 and 6 sho

Re: [Asterisk-Users] Provisioning lines 5 and 6 via TFTP

Can you please post your .cnf files? On 4/21/05, Robert Goodyear <[EMAIL PROTECTED]> wrote: > Has anyone experienced a problem provisioning lines 5 and 6 of a Cisco > 7960 via a SIPx.CNF over TFTP? > > What I'm experiencing is that regardless of the linex_... entries in > the CNF file, lines

Re: [Asterisk-Users] mpg123 won't compile, arch x86_64

try compling using 'make linux-devl' otherwise play around with formatmp3 after doing a cvs checkout asterisk-addons On 4/21/05, Michael Welter <[EMAIL PROTECTED]> wrote: > mpg123 won't compile on my Opteron system. Doesn't seem to like the > pushl and popl assembly instructions, ie.e, "pushl %e

[Asterisk-Users] X100P delayed ring on incoming calls?

I have setup an asterisk box with 3off X100P cards and hooked them up to the PSTN. So far so good, everything does what it is supposed to do for the msot part. Incoming calls seem to ring three or four times before asterisk then skips to do what it is supposed to do. If the caller drops the ca

Re: [Asterisk-Users] Adit 3104 - user experiences?

>From what I was able to gather on CACs web site the 3104 only supports 24 analog ports. On 4/20/05, Peter Hoppe <[EMAIL PROTECTED]> wrote: > Hello, > > I am looking for a solution to connect about 40 analog telephones to an > Asterisk pbx. Initially I wanted to use an Adit 600 channel bank, but

Re: [Asterisk-Users] X100P delayed ring on incoming calls?

On Fri, 2005-04-22 at 10:22 +0100, bam wrote: > I have setup an asterisk box with 3off X100P cards and hooked them up > to the PSTN. So far so good, everything does what it is supposed to do > for the msot part. > > Incoming calls seem to ring three or four times before asterisk then > skips to do

Re: [Asterisk-Users] X100P delayed ring on incoming calls?

On Friday 22 April 2005 10:45, Dave Cotton wrote: > On Fri, 2005-04-22 at 10:22 +0100, bam wrote: > > Incoming calls seem to ring three or four times before asterisk then > > skips to do what it is supposed to do. If the caller drops the call > > before the extensions have started ringing asterisk

Re: [Asterisk-Users] X100P delayed ring on incoming calls?

3-4 rings seems kind of long, I usually see 1.5-2 (enough to grab the caller ID here in the States). The only way I know of to speed it up is to turn off all of the features like distinctive ring detection, caller ID, etc. -- depending on your usage, that may help some. I haven't confirmed that th

Re: [Asterisk-Users] X100P delayed ring on incoming calls?

On Fri, 2005-04-22 at 10:45, Dave Cotton wrote: On Fri, 2005-04-22 at 10:22 +0100, bam wrote: > I have setup an asterisk box with 3off X100P cards and hooked them up > to the PSTN. So far so good, everything does what it is supposed to do > for the most part. > > Incoming calls seem to ring t

Re: [Asterisk-Users] One touch voicemail on Cisco 7940/60

On Thu, 2005-04-21 at 21:36 +0100, Ron Wellsted wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Morris, Simon wrote: > > Hello, > > > > I'd like to program my Cisco phones to authenticate themselves to > > voicemail upon hitting the right button on my 7940/60's > > > > Ideally the voi

[Asterisk-Users] DTFM tones almost completly muted.

Hello everyone, I am trying to receive DTMF commands on asterisk from PSTN calls terminated at my asterisk box. I have tried to terminate the PSTN calls with both SIP and IAX using sigate.co.uk and voipuser as the PSTN terminator. When I listen to tones sent from the PSTN side (e.g. continuous

[Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P?

Hello, I've been asked to build a couple of Gateway servers for a client w/ TE405P hardware, and have been looking around at various 1U options. I've been looking at SuperMicro and Tyan barbones boxes as possible platforms, but then was directed to Dell's SC1425 by a friend. Short story

[Asterisk-Users] IAX2 Error

Anyone has any idea what does this error means when executing an IAX2 call? Apr 22 11:50:19 WARNING[9124]: Received mini frame before first full voice frame The called party can hear but the calling, no. Is this a fine tunning into iax.conf? Thanks, Robson ___

Re: [Asterisk-Users] livevoip callerid

I don't think it's correct to put dashes in the CIDNum. MARK. Paul Fielding wrote: Hmmm... I still can't get name, though number works. Perhaps I'm missing something? context livevoip in iax.conf that hooks me to livevoip dial 9 in front of long distance number to dial livevoip instead of regul

Re: [Asterisk-Users] Echo cancelling with Adit 600

On Fri, 22 Apr 2005, Daniel Nyström wrote: > Do anyone have experience with echo cancelling on Adit 600? > My Adit 600 consist of 5*8 FXS cards and 1 CMG Router using MGCP to Asterisk. > I've turned on Echo Cancelling with 64ms as longest delay (that's maximum). > > But there still are great echo

Re: [Asterisk-Users] DTFM tones almost completly muted.

On 22/04/05, Ian Hailey <[EMAIL PROTECTED]> wrote: > Hello everyone, > > I am trying to receive DTMF commands on asterisk from PSTN calls > terminated at my asterisk box. I have tried to terminate the PSTN calls > with both SIP and IAX using sigate.co.uk and voipuser as the PSTN > terminator. When

Re: [Asterisk-Users] Line Noise UPDATE - If you've got line noise, read this

On Fri, Apr 22, 2005 at 02:40:10AM -0500, Paul said: > > Ok, well I managed to fix the IRQ conflicts...well, sorta. I have two X100P > cards in the system. One now has it's own interrupt and the other is sharing > one with the soundcard. I tested outbound calls on both cards, still have > the damn

Re: [Asterisk-Users] asterisk@home 0.9 zap problems

> > -- Executing Dial("SIP/3001-e13a", "ZAP/1/65869804") in new stack > > This is what's wrong I think. The line is missing the 'g' for the trunk > group. On all of my [EMAIL PROTECTED] boxes the cli shows > >-- Executing Dial("SIP/227-a4dd", "ZAP/g0/3428463") in new stack It depends how y

Re: [Asterisk-Users] X100P delayed ring on incoming calls?

Joseph Gutowski <[EMAIL PROTECTED]> wrote: [...] > Either way, the best I've ever managed on the X100P's was 1 ring > before Asterisk picks up and starts doing its thing. Well, when you think about it, it's hardly going to pick up after zero rings, is it? :) -- Beer is proof that God loves us an

Re: [Asterisk-Users] X100P delayed ring on incoming calls?

On Friday 22 April 2005 12:07, Peter Corlett wrote: > Joseph Gutowski <[EMAIL PROTECTED]> wrote: > [...] > > > Either way, the best I've ever managed on the X100P's was 1 ring > > before Asterisk picks up and starts doing its thing. > > Well, when you think about it, it's hardly going to pick up af

Re: [Asterisk-Users] X100P delayed ring on incoming calls?

> 3-4 rings seems kind of long, I usually see 1.5-2 (enough to grab the > caller ID here in the States). > > The only way I know of to speed it up is to turn off all of the > features like distinctive ring detection, caller ID, etc. -- depending > on your usage, that may help some. I haven't conf

Re: [Asterisk-Users] X100P delayed ring on incoming calls?

Joseph Gutowski wrote: The only way I know of to speed it up is to turn off all of the features like distinctive ring detection, caller ID, etc. -- depending on your usage, that may help some. I haven't confirmed that this actually does anything myself, but it seems logical that Asterisk could pick

Re: [Asterisk-Users] X100P delayed ring on incoming calls?

Gavin Hamill <[EMAIL PROTECTED]> wrote: > On Friday 22 April 2005 12:07, Peter Corlett wrote: [...] > In the UK it's entirely possible - the CallerID info comes through > as encoded data before the first ring has taken place :) > Polarity change, a burst of V23 data, then the normal rings A g

Re: [Asterisk-Users] Asterisk Restart after crash

There is a bug with safe_asterisk and FC2, you must edit the script to remove 'daemon' from the the startup command and then it will auto restart. Craig - Original Message - From: "David Phelan" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Fr

Re: [Asterisk-Users] Line Noise UPDATE - If you've got line noise, read this

I have three in one machine, and 4 customers that have 2 in each of their machines. The only problem I've ever had is momentary echo when a call first begins, but that is to be expected until the line trains. - Original Message - From: "Paul" <[EMAIL PROTECTED]> To: "'Asterisk Users Mai

Re: [Asterisk-Users] Re: Email to Fax

You could try http://www.inter7.com/?page=astfax - I haven't used it yet myself but it looks like it'll work. Craig - Original Message - From: "Anton Krall" <[EMAIL PROTECTED]> To: "'Justin Newman'" <[EMAIL PROTECTED]>; "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: F

RE: [Asterisk-Users] Error in starting asterisk

> channel=1-15,17-30 > > Apr 22 06:02:04 WARNING[18915]: parse error: No category context for line 10 > of zapata.conf > Apr 22 06:02:04 ERROR[18915]: Unable to load config zapata.conf Just a couple of guesses here... I'm not so sure the "line 10" is counting correctly (or you're not counting th

Re: [Asterisk-Users] Line Noise UPDATE - If you've got line noise, read this

Paul wrote: Ok, well I managed to fix the IRQ conflicts...well, sorta. I have two X100P cards in the system. One now has it's own interrupt and the other is sharing one with the soundcard. I tested outbound calls on both cards, still have the damn static. I am so sick of this. Is anyone else using

RE: [Asterisk-Users] Asterisk Restart after crash

Could you give some more information on where to remove 'daemon' and the effects? Since all our productionservers running FC2 I'm a bit concerned. > There is a bug with safe_asterisk and FC2, you must edit the script to > remove 'daemon' from the the startup command and then it will auto restart.

RE: [Asterisk-Users] Asterisk Restart after crash

OK how do you know if it's running in safe_asterisk mode? I am running [EMAIL PROTECTED] Does that run in safe mode by default? What file do you look at to see how asterisk starts up? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Phelan Sent: Fri

Re: [Asterisk-Users] IAX2 Error

On April 22, 2005 08:33 am, Robson Ribeiro wrote: > Apr 22 11:50:19 WARNING[9124]: Received mini frame before first full voice > frame It means nothing, only that the first two packets out of the gate arrived in the opposite order that they were sent. Simple explanation: Asterisk sends small

[Asterisk-Users] Asterisk transcoding

I would like to learn more over the transcoding function with asterisk. How exactly works asterisk, in order to transcoding. Where I can get exactly informations? If asterisk transcodes, for example ilbc to gsm, as I can see which (ilbc) rtp-packet becomes which (gsm) rtp-packet? would be ver

Re: [Asterisk-Users] Line Noise UPDATE - If you've got line noise, read this

On April 22, 2005 03:40 am, Paul wrote: > Ok, well I managed to fix the IRQ conflicts...well, sorta. I have two X100P > cards in the system. One now has it's own interrupt and the other is > sharing one with the soundcard. I tested outbound calls on both cards, > still have the damn static. I am so

[Asterisk-Users] callto: URL (URI) tag for dialing

I see that there seems to be a 'callto' URL/URI for dialling a phone number... ie - on my web site's "Contact Page" - I have added the code... +27 12 807-0590 There should be some generic way for Mozilla (firefox - etc) to somehow turn a click on such a link into persuading Asterisk to dial the nu

Re: [Asterisk-Users] Re: chan_unicall.c compile error

Fabio, check if you have libtiff and libtiff-devel installed. and also you have to patch the asterisk source code first.ç I dont know if you did that... regards, Hector. On 4/21/05, Fabio Vasco <[EMAIL PROTECTED]> wrote: > Hector, > > This is my Linux Fedora Core 3 version info > > [EMAIL

[Asterisk-Users] Mysql using Sip and voicemail

I am currently running asterisk 1.0.7 and decided to try using MySQL to hold some of my voicemail and sip configuration. As a note - MySQL is already holding my CDR info. I followed the directions in on voip-info.org to copy files, modify Makefiles, recompile, and change the conf files accordi

Re: [Asterisk-Users] using * for Internet call waiting

The * box would sit in a CO connected via PRIs. Gary Gary Carr wrote: Wondering if it is possible or if something already exist to setup * to offer Internet Call Waiting. For those that do not know what it is, it's a small application that runs on a users computer that will pop up a window let

RE: [Asterisk-Users] ASTCC

Chris, There is no "official" documentation, but here is what I have found in the control panel. *BRANDS* This is where you can setup different cards with different Service fees. I'm not sure what the INC column is for, I usually leave it set at 6. I think it sets 6 seconds to the minimum bill

Re: [Asterisk-Users] using * for Internet call waiting

The TDM part is pretty simple. The end user needs the call forward busy feature on thier line with the calls being forwarded to the * server. Taking it from there and sending it to a app on the users machine is whats left. I was thinking it could be sent with sip and a long timeout value. Gary

Re: [Asterisk-Users] using * for Internet call waiting

That's pretty close to what we are looking for but we want the user to have the option of taking the call which would disconnect the modem connection and allow the call to ring thru to the phone. Not sure how to accomplish that. I am sure our programmer could code a client but he has no experien

RE: [Asterisk-Users] ASTCC

For everyone's information and so it is on the list somewhere, there is a copy of this at http://voip-info.org/tiki-index.php?page=ASTCCGuide This also includes the explanation of the CDR problem. > -Original Message- > From: Dave Kettmann > Sent: Friday, April 22, 2005 8:03 AM > To: As

[Asterisk-Users] Asterisk acting as PBX + SIP Proxy ... possible?

Hello, I'm in the process of implementing the following setup External SIP phones at another location(s) (nat = yes) | | Analog phone line | | |-- |ext if 142.x.x.41 | |Asterisk | |int if 192.168.0.1 |-- | Internal SIP Phones (nat

Re: [Asterisk-Users] One touch voicemail on Cisco 7940/60

On Fri, 2005-04-22 at 11:13 +0100, Simon Morris wrote: > On Thu, 2005-04-21 at 21:36 +0100, Ron Wellsted wrote: > > -BEGIN PGP SIGNED MESSAGE- > > Hash: SHA1 > > > > Morris, Simon wrote: > > > Hello, > > > > > > I'd like to program my Cisco phones to authenticate themselves to > > > voicema

[Asterisk-Users] No such context/extension

To All, I am a new to Asterisk and dialplans have me stumped I just inherited 2 Asterisk servers conected as IAX peers. Now from what i can tell when Asterisk Server (ask-CHIC) needs to make a call to an extension which resides on the other server (ask-MAIN) it goes over a IAX channel. Now i a

[Asterisk-Users] Alcaterl IP-touch phones

Hi all, Has any one tested Asterisk with the new Alcatel IP-touch phones (IP phones with xml) Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update option

Re: [Asterisk-Users] No such context/extension

On 22/04/05, MDM <[EMAIL PROTECTED]> wrote: > [EMAIL PROTECTED] asterisk]# cat iax.conf > ; BMS-ask-Main-asterisk - Incoming - > ; > [ask-mail] > type=user > secret=ask-mail > context=from-ask-main > disallow=all > allow=ulaw There are probably some typos in there which might be adding to your pr

Re: [Asterisk-Users] Debugging zaphfc + PBX integration

Gavin Hamill wrote: I know the cables themselves are wired correctly because our local PBX support made them, and they work perfectly when plugged into a real BT ISDN2e wallbox it seems as if this is exactly your problem. the wallbox has a NT pinout => straight trough cable the hfc card has a TE

[Asterisk-Users] Dynamic queue member behaviour

I found that if I dynamically add, for example SIP/8000, to a queue, then calls in the queue will sorta pile up on the 9 extensions on that phone - not what we want to happen. If I log in to the queue using AgentLogin, then the behaviour is as expected - one call at a time. Is there a way around

Re: [Asterisk-Users] One touch voicemail on Cisco 7940/60

On Fri, April 22, 2005 9:27 am, Simon Morris said: > So.. how to bypass the "Enter your mailbox number" stage in voicemail > and go straight to the password prompt. Remove the "s" at the beginning of the argument to VoiceMailMain() http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Voi

Re: [Asterisk-Users] One touch voicemail on Cisco 7940/60

That solution does exactly what I asked BUT! I'd like it to dial and know which extension I'm coming from and then prompt for the password > exten => _8501,2,VoicemailMain(s${CALLERIDNUM}) Just remove the 's' from the line above. Not to sound like a smart ass, but this is all very well document

Re: [Asterisk-Users] Alcaterl IP-touch phones

I don't believe they speak SIP - not yet, that is ;) Herve. - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, April 22, 2005 3:36 PM Subject: [Asterisk-Users] Alcaterl IP-touch phones Hi all, Has any one tested Aster

Re: [Asterisk-Users] Asterisk Restart after crash

In your asterisk script in init.d that calls safe_asterisk change this: start() { # Start daemons. echo -n $"Starting asterisk: " if [ -f $SAFE_ASTERISK ] ; then DAEMON=$SAFE_ASTERISK fi if [ $AST_USER ] ; then ASTARGS="-U $AS

[Asterisk-Users] can't make my PRI dial out

I have a full PRI installed on my * machine. I can get inbound calls just fine but can't make outbound ones. Zaptel.conf says; span=1,1,0,esf,b8zs bchan=1-23 dchan=24 zapata.conf says language=en context=default switchtype=4ess pridialplan=unknown signalling=pri_cpe channel=>1-23 echocancel=yes g

Re: [Asterisk-Users] Provisioning lines 5 and 6 via TFTP

On Apr 22, 2005, at 2:13 AM, C F wrote: Can you please post your .cnf files? On 4/21/05, Robert Goodyear <[EMAIL PROTECTED]> wrote: Has anyone experienced a problem provisioning lines 5 and 6 of a Cisco 7960 via a SIPx.CNF over TFTP? I'm going to try Ron Wellsted's suggestion re the .CNF files

Re: [Asterisk-Users] Recommended Linux Dist. for Asterisk

Using RH 9 with * Regards, Umair Bari David Choo wrote: We used gentoo internally. I also have * running on CentOS, RHEL. Best Regards, == David Choo Systems Engineer Business & Technology Division "Engineered for Changing Businesses" Espore Corp Pte Ltd 68 K

Re: [Asterisk-Users] DTFM tones almost completly muted.

On 22/04/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > > > On Fri, 22 Apr 2005, Peter Bowyer wrote: > > > On 22/04/05, Ian Hailey <[EMAIL PROTECTED]> wrote: > > > Hello everyone, > > > > > > I am trying to receive DTMF commands on asterisk from PSTN calls > > > terminated at my asterisk box

Re: [Asterisk-Users] can't make my PRI dial out

On April 22, 2005 11:48 am, Mark Phillips wrote: > Nothing happens. I get the same (non)error. > I get plenty of output when receiving a call however. Odd... Here is my zapata.conf setup for my PRI: --- [channels] context=BellPRI switchtype=national pridialplan=unknown priin

RE: [Asterisk-Users] QOS Routers

You may want to check out edgewaternetworks. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Max Clark > Sent: Friday, April 22, 2005 11:42 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] QOS Routers

[Asterisk-Users] Rejected connect attempt

Hello, I have seen the following in my log files. For the life of me I can not work out why. Apr 22 22:10:40 NOTICE[19236] chan_iax2.c: Rejected connect attempt from 65.39.205.121, who was trying to reach 'i@' Would someone explain why, or point me in the direction I can read about it? Many th

RE: [Asterisk-Users] Mysql using Sip and voicemail

Actually, I just got the SIP phones working right before I received. I am not exactly sure what I changed, but I basically went through and recreated, recompiles, etc then it worked. I will keep the "sip debug" in mind if I have any more sip problems. Thanks for your help. Still can not figur

Re: [Asterisk-Users] Debugging zaphfc + PBX integration

On Friday 22 April 2005 14:46, Frank Sautter wrote: > Gavin Hamill wrote: > it seems as if this is exactly your problem. Sorry Frank, but this one isn't as simple as cabling... I've made reference in this thread already that I do have both straight + ISDN crossover (3/4 and 5/6 swapped) cables,

RE: [Asterisk-Users] Line Noise UPDATE - If you've got line noise, read this

We have basically the same setup. My cards are on 5 and 7 as well and I've disabled EVERYTHING is the bios that is not necessary; USB, serial, parallel, ect. I would think that if it was an IRQ issue, the call wouldn't tank when I connected it on the card with it's own IRQ. I just got in my new Cis

[Asterisk-Users] TDM-fxo card and zttest - logic probem?

Been playing around with zaptel/zttest utility and believe there is a logic problem with this 83 line app. (The objective is to better undertand missed frames, interrupts, etc, associated with the TDM card. Maybe we can get a handle on why things like spandsp failures, echo, etc, are occurring in

[Asterisk-Users] No sound with voicemail and musiconhold?!?

Hi! I'am a new user and have problem with sound on a debian sarge. I can't play any sound with musiconhold or voicemail. Sounds on var/lib have good rights and mpg123 is installed. On console asterisk stops in the first playing. Someone have same problem or can help me? -- Antoine __

Re: [Asterisk-Users] Demo phones with advertisement announcements

Craig wrote: There is a great missunderstanding between what you guys are talking now and what I want. I am not looking for advertiser to support free calls. However, I am asked now more often "Can I test your service for a few days?" These tests I have to pay from my own pocket. The customers sh

[Asterisk-Users] chan capi: Long incomingmsn line in capi.conf?

Hello, the incomingmsn line in chan_capi's capi.conf is limited to 80 characters (AST_MAX_EXTENSION default value). My problem: I have to include several MSNs but NOT all. The interface is a 30 channel PRI card with a number area of 600 numbers, splitted in different functions. Some numbers are

Re: [Asterisk-Users] Route SIP calls to provider

Your SIP provider doesn't need registration? Sounds good. Can you share the IP address please? Regards Cameron - Original Message - From: "iMRAN" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, April 21, 2005 4:35 AM Subject: [Asterisk-

RE: [Asterisk-Users] Mysql using Sip and voicemail

I can give you some help with the SIP stuff. Try it again with "sip debug" turned on and send the output back here. It would be good to see the SIP messages that are being transferred. The Asterisk SIP Stack is good, but not great. You might need to just add or delete an option in the sip.conf fi

[Asterisk-Users] Grandstream : low bandwidth codec (ilbc doesn't work, any other ? )

Hi, I'm trying to setup one of free low bandwidth codecs for Grandstream (ilbc, g726, ...), but with ilbc I just hear "engine running" in handset. Is anyone using ilbc sucessfully with Grandstream? Quality ? Any other alternative ? I use Bristuffed Asterisk Thanks in advance, regards, Rob.

[Asterisk-Users] SS7 for *

Hi!, Do you have a copy of the openss7 stack?? +*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*+ + Luciano Ramos+ + MCP - CCNA - CCNP (on the way :-)+ + Depto. de Internet, TelViso + + [EMAIL PROTECTED]+ + 02320-409125

Re: [Asterisk-Users] can't make my PRI dial out

On Fri, 22 Apr 2005 10:37:32 -0400 Mark Phillips <[EMAIL PROTECTED]> wrote: I have a full PRI installed on my * machine. I can get inbound calls just fine but can't make outbound ones. Zaptel.conf says; span=1,1,0,esf,b8zs bchan=1-23 dchan=24 zapata.conf says language=en context=default switchty

[Asterisk-Users] Recommendations for Spanish Voice Talent

We are putting together an IVR app that requires Spanish prompts. We're using Allison for the English prompts and are looking for recommendations for Spanish. Any thoughts? -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET

RE: [Asterisk-Users] TE11OP -> Mitel 200Sx??

I have done the same thing with an sx200 and a pri circuit zaptel.conf # t1 connected to the PRI circuit span=1,1,0,exf,b8zs # t1 connected to SX200 # the t1 card on my sx200 did d4 ami and I supplied ANI and DNIS through the dial plan span=2,0,0,d4,ami bchan=1-23 dchan=24 e&m=25-47 -

RE: [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P?

Dell 1850 rack mount. We've been sourcing white box servers but can't beat Dell's price in the U.S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Boehnlein Sent: Friday, April 22, 2005 2:57 PM To: Asterisk Users Mailing List - Non-Commercial Di

RE: [Asterisk-Users] Demo phones with advertisement announcements

Yep totally, same in Australia, pricing for long distance calls have crashed and changed the business case for this now and more so in the future. Voice advertising still does have it's place though - maybe in subsidising 'chat' rooms, with these you can definitely set up locally advertising acces

Re: [Asterisk-Users] Recommendations for Spanish Voice Talent

We have recorded some prompts in Spanish by a male speaker, pls contact me offline for sending some of them if it suits your needs. Saludos / Regards Gustavo Russo - Original Message - From: "George Pajari" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussio

Re: [Asterisk-Users] Re: TE110p - universal voltage?

Thanks all. I too have found out that the card is both. Mike Tony Mountifield wrote: In article <[EMAIL PROTECTED]>, Craig Guy <[EMAIL PROTECTED]> wrote: Can anyone with a TE110p confirm that it will fit and work in both a 3.3 and 5 volt pci slot? From photos it looks to be a un

Re: [Asterisk-Users] can't make my PRI dial out

I haven't worked with PRI, but could it be related to an invalid callerid? What about: exten => _X., 1, SetCallerId(123123123) exten => _X., 2, Dial(Zap/g1/${EXTEN}) Julian. On 4/22/05, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote: > On April 22, 2005 11:48 am, Mark Phillips wrote: > > Nothing h

[Asterisk-Users] Re: can't make my PRI dial out

On Fri, Apr 22, 2005 at 11:10:43AM -0500, [EMAIL PROTECTED] wrote: > I have a full PRI installed on my * machine. I can get inbound calls > just fine but can't make outbound ones. If you run "pri debug span x", you might see this behaviour: PRI debugging with the inbound numbers show tha

RE: [Asterisk-Users] Demo phones with advertisement announcements

Yep, remember them well - I think the guy who was running it was called Paul Davies (could be wrong) - no idea if they are still running. I understand the big problem they had was securing advertising contracts, people under estimate the 'startup cost' in securing advertising (lol - something I'm

Re: [Asterisk-Users] can't make my PRI dial out

I made my zapata.conf look like the below (with relevant changes) and then programmed exten 3701 to dial my cell phone (I'm working remotely on this). I added the line exten => 3701,1,Dial(Zap/g1/19173657597) to extensions.conf and get this output from pri debug span 1 when I dial it -- Making n

[Asterisk-Users] Questions about a 7960 and images

Hello All, I was wondering how everyone got along with cisco 7960's. I just picked one up and I am having problems locating an image. I called cisco, but they will not sell to end users... Does anyone know a place where it can be purchased in the US? It has stock firmware, and the skinny seems t

Re: [Asterisk-Users] voice pulse connect - no dtmf

I had the same problem with another provider whom I got no response from as usual..   We had 5 or 6 numbers that worked fine and one that just quit sending DTMF.     - Original Message - From: Doug Harris To: [EMAIL PROTECTED] Digium. Com Sent: Friday, April 22, 20

Re: [Asterisk-Users] can't make my PRI dial out

I know we are moving forward. I didn;t get this last time I tried to dial. Mark Why don't you try changing your switchtype to "national" from "4ess" in your zapata.conf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.di

RE: [Asterisk-Users] Line Noise UPDATE - If you've got line noise, read this

Since this thread has been going on for awhile, I've forgotten whether anyone mentioned that at least some Sipura products shipped with a 10 millisecond rtp time. My spa3k was this way. I changed it to 20 milliseconds and reboot. Might just try that if the setting is avail to you. ---

Re: [Asterisk-Users] Quadbri & bristuff: can * respond only to 1 MSN and leave 1 number to other ISDN phones ?

Hi, > I have problem with Quadbri and bristuffed Asterisk - I guess this is only > configuration trick. I'd like Asterisk to respond only to 1 number on BRI > interface and do nothing on other. Right now, even if I leave out that > number in incoming context, Asterisk takes out and rejects call

[Asterisk-Users] Upgrade Cisco 7940/7960 firmware

For the archive: http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7960/addprot/mgcp/frmwrup.htm#wp1048832 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To U

RE: [Asterisk-Users] QOS Routers

How about a linksys wrt54g with sveasoft firmware? Has some shaping and many other nice features... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Max Clark Sent: Friday, April 22, 2005 1:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion

[Asterisk-Users] Re: QOS Routers

Maybe this fits the bill. It retails for less than $100 > Message: 9 > Date: Fri, 22 Apr 2005 10:42:20 -0700 > From: Max Clark <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] QOS Routers > To: Asterisk Users Mailing List - Non-Commerc

RE: [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P?

On Fri, 22 Apr 2005, William Boehlke wrote: > > SC1425 is great value but note it does not have high availablility > configurations. > > In our opinion, telephony requires dual NICs, dual power supplies and RAID 1 > to have any hope of achieving five nines. > > William Boehlke What box would

RE: [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P?

I'm interested in the answer to this question as well, except that my project scope (and budget) are quite a bit lower than yours. I've always preferred buying Dell, but in this case, I think it may be overkill for this project. I'd really like to know what inexpensive rack-mount servers would wo

[Asterisk-Users] Asterisk + Cisco 2620

Hello, Would I need to do anything special with either Asterisk or the 2620, aside from QOS stuff, to get these two working together? We'll be getting the router in on Monday and I would like to make sure it transitions as smoothly as possible. Currently we just go through an Adtran TA850. W

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