[Asterisk-Users] Jingle support - can we test the feature ?

Hi, we would like to build IM-Voice community for our students around Asterisk, Jingle, Jabber. Can we already test those features ? Anyone already running such setup? Any more info ? Thanks in advance, regards, Rob. ___ --Bandwidth and Coloca

Re: [Asterisk-Users] Meetme codec translation and callerID library.

tom wrote: Also, does anyone know if there's a way to dynamically alter incoming Caller-IDs to add Caller ID text to them. ie. call comes in with ID 01234 567890 gets changed to "A Company" <01234 567890> ? Look at cid_rewrite, here: http://muware.com/asterisk You'll have to adjust the nu

Re: [Asterisk-Users] dundi trouble

in the documents, it said that   ; 'dest' is the destination to supply for reaching that number.  The; following variables can be used in the destination string and will; be automatically substituted:; ${NUMBER}: The number being requested ; ${IPADDR}: The IP address to connect to; ${SECRET}: T

Re: [Asterisk-Users] dundi trouble

those two number is setuped in extensions.conf..     have get the root of cause   I Need to change the {IPADDR} to the ip address.. the asterisk version is 1.2.0   don't know how about asterisk 1.2.6///   thanks  On 4/20/06, Ronald Wiplinger <[EMAIL PROTECTED]> wrote: 陈帆 wrote:> whe

[Asterisk-Users] asterisk 1.2.7.1 crashing my newly built system

Hello folx! I just started to play with *. I first installed it this past weekend on my Solaris 9 ultra 5 test box. Now I'm attempting to put it on a freshly built Linux box a mere few hours old. I've installed asterisk 1.2.7.1, libpri-1.2.2, zaptel 1.2.5 and the latest asterisk-sounds 1.2.1. This

Re: [Asterisk-Users] ANNOUNCE: Asterisk Jobs and Consulting Site

On Thursday 20 April 2006 00:13, Matt Gibson wrote: > I would like to announce the availability of a new site dedicated > to finding and creating jobs in the Asterisk VOIP field. I've created > this site, after noticing there are no sites dedicated to providing > quality job postings and hiring abi

[Asterisk-Users] TDMoE

Hello everyone: Somebody knows what i have to do to configure TDMoE between two asterisk and use PRI signalling in between??? Regards. Cristian. _ Don’t just search. Find. Check out the new MSN Search! http://search

[Asterisk-Users] sip.conf codecs: ulaw, alaw and g729

Hi, When ever I put g729 in allow for trunk the other two codecs (ulaw and alaw) stop working and I get the frame type error for them, but g729 works fine. I've cleared general part of sip.conf of codec info to be on safe side. If ulaw and alaw are the only ones allowed they work fine. Asterisk

[Asterisk-Users] ANNOUNCE: Asterisk Jobs and Consulting Site

Greetings, I would like to announce the availability of a new site dedicatedto finding and creating jobs in the Asterisk VOIP field. I've createdthis site, after noticing there are no sites dedicated to providing quality job postings and hiring abilities to people in the field. http://www.asterisk-

Re: [Asterisk-Users] dundi trouble

陈帆 wrote: when i test dundi why there report 80001 is exist in 127.0.0.0 instead of 192.168.8.131 ? localhost*CLI> dundi lookup [EMAIL PROTECTED] DUNDi lookup returned no results. DUNDi lookup completed in 34 ms loca

Re: [Asterisk-Users] What's the best way to combine multiple VOIP lines into a single number?

On Wednesday 19 April 2006 19:14, tom wrote: > I don't see what the problem is, most providers will forward a second > call to you made to the same number/account, some will forward as many > calls as your bandwidth can handle. If they're all from the same provider it shouldn't be a problem. And

RE: [Asterisk-Users] Asterisk service crashes

Interesting, I haven't set a hostname since I built the server almost a year ago. I wonder why only now would the problem arise. William _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Jones Sent: Wednesday, April 19, 2006 10:56 PM To: Asterisk Users Mailing Li

[Asterisk-Users] dundi trouble

when i test dundi   why there report 80001 is exist in 127.0.0.0 instead of 192.168.8.131 ?     localhost*CLI> dundi lookup [EMAIL PROTECTED]DUNDi lookup returned no results.DUNDi lookup completed in 34 mslocalhost*CLI> dundi lookup [EMAIL PROTECTED]  1. 0 IAX2/dundi:[EMAIL PROTECTED]/8000

RE: [Asterisk-Users] Upgrade from 1.2.4 to 1.2.7.1

Thanks, I found it though. I needed the latest addons package. mysql CDR wasn’t there.   From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Wednesday, April 19, 2006 11:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE:

Re: [Asterisk-Users] attended transfer issue

Eric "ManxPower" Wieling wrote: Damon Estep wrote: Is the current release different than what I am running, # transfer on my systems are all blind, no attended option. 1.0.x only supported blind DTMF transfer hack. 1.2.x supports both blind and supervised DTMF transfer hacks. See feature

RE: [Asterisk-Users] Upgrade from 1.2.4 to 1.2.7.1

Run asterisk from the command line and dont put it in the background:   like so:   asterisk -cvv   This will tell you what the error is. fix it and rerun the service. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]Sent: Wednesday, Ap

RE: [Asterisk-Users] Asterisk service crashes

I had a similar problem, and it was because my hostname had issues... I'm not sure why/how, but if my hostname was valid, and had a valid fwd/reverse dns entry, everything was OK again.. -Steve From: Josué Conti [mailto:[EMAIL PROTECTED] Sent: Wed 4/19/2006 9

RE: [Asterisk-Users] Upgrade from 1.2.4 to 1.2.7.1

I deleted the modules directory, then ran make and make install. I then did “service asterisk start” and asterisk –r but it says:   [EMAIL PROTECTED] asterisk-1.2.7.1]# asterisk -r Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) [EMAIL PROTECTED] aste

Re: [Asterisk-Users] lost audio after zaptel

This can happen if you haven't set up the digium card properly - strange but true. -- Paul Hales Technical Manager Asterisk IT mob: 0434 225 491 Doug Langley wrote: Hi. Last week I installed asterisk 1.2.7.1 on a machine running Fedora core 5 with kernel 2.6.15 Today we installed a t100p

[Asterisk-Users] lost audio after zaptel

Hi. Last week I installed asterisk 1.2.7.1 on a machine running Fedora core 5 with kernel 2.6.15 Today we installed a t100p digium card and I installed libpri, zaptel and rebuilt asterisk with make all and make install. Now after rebuilding asterisk, there is no audio coming back from any IVR

RE: [Asterisk-Users] Upgrade from 1.2.4 to 1.2.7.1

Sorry forgot about that one. it does a make all and then a bininstall, (installs binaries)   It does NOT however remove your modules, I would still do the rm -rf /usr/lib/asterisk/modules/* before I do a make.   From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMA

RE: [Asterisk-Users] Upgrade from 1.2.4 to 1.2.7.1

Ok, I thought that was the case but I seem to remember doing “make” then “make upgrade” in the past. Is this no longer the way to do it or just another way?   From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Wednesday, April 19, 2006 9:51 PM To:

RE: [Asterisk-Users] Upgrade from 1.2.4 to 1.2.7.1

This is the order that is recomended   Zaptel libpri asterisk   after unzip/untar cd into use directory and run make and then a make install.   I would suggest you clean out your modules directory to be safe.   rm -rf /usr/lib/asterisk/modules/*   From: [EMAIL PROTECTED] [mailto:[E

RE: [Asterisk-Users] Meetme codec translation and callerID library.

snip > > Can Meetme be made to work with G.729? (I gather not) > IIRC, MeetMe does it 'mixing' using SLIN (Signed Linear, * should transcode to/from g.729 to SLIN. > If a call comes in (internally or externally), the call comes in as a > G.729 call, which then re-negotiates to a G.711u call when

[Asterisk-Users] Upgrade from 1.2.4 to 1.2.7.1

List,   I wish to upgrade from 1.2.4 to 1.2.7.1 I have downloaded & unzipped the file but how do I compile it?   Do I need to “make clean” then “make” and “make upgrade”? Or “make” then “make install”?   Thanks,   William      

RE: [Asterisk-Users] Ring a grop of extension, then playback a file, then transfer to external number

I think he wanted now instead of not. Changes the whole meaning of the question!!! snip > > I not want to add a playback of a file ("Please waite > while you are > > being transfered") before transfering the call to the cell phone. Snip, snip I think he wanted to say: "I NOW want to add a

Re: [Asterisk-Users] Ring a grop of extension, then playback a file, then transfer to external number

Andre Courchesne - Consultant wrote: Ok, Here is what I got working: A call comes in from a Zap line. 5 SIP extension ring if nobody picks up, the call is transfered to a cell phone number. That works. I not want to add a playback of a file ("Please waite while you are being transfe

Re: [Asterisk-Users] attended transfer issue

Damon Estep wrote: Is the current release different than what I am running, # transfer on my systems are all blind, no attended option. 1.0.x only supported blind DTMF transfer hack. 1.2.x supports both blind and supervised DTMF transfer hacks. See features.conf in 1.2.x __

Re: [Asterisk-Users] MultiBRI in Australia - found one - maybe

On Wednesday 12 April 2006 18:51, MBIT Technologies wrote: [regarding the Draytek Minivigor 128] > Any idea where I can get some of these units in Melbourne? According to Draytek AU they have been discontinued :( hads -- Age is important only if you're cheese and wine. ___

Re: [Asterisk-Users] Asterisk & GNUDialer issue

Hello Facundo, But for test if it will be able compiles the libraries of asterisk again, but execute before "make clean". It places in the list the results. Good luck Best Regards Josué  2006/4/18, Facundo Ameal <[EMAIL PROTECTED]>: Hello everybody, I'm installing an Asterisk 1.2.7.1 with GNUDial

Re: [Asterisk-Users] Fwd: sip.conf and jump from register to the extension

Thomas Winter wrote: > I have done additional tests, because the documentation sample was not 100 % > identical to my register command. > > OK: > register => 44198:[EMAIL PROTECTED]/200 > This jumps to 200, s is also working > > NOT OK: > user:[EMAIL PROTECTED]/200 > It looks for extension user

Re: [Asterisk-Users] What's the best way to combine multiple VOIP lines into a single number?

tom wrote: Andrew Kohlsmith wrote: On Wednesday 19 April 2006 12:23, Leo Burd wrote: I'm new to Asterisk and I'm wonder what's the best way to combine multiple VOIP lines into a single phone number... Do you mean VOIP lines as in "numbers my customers can call to get to m

Re: [Asterisk-Users] What's the best way to combine multiple VOIP lines into a single number?

Andrew Kohlsmith wrote: > On Wednesday 19 April 2006 12:23, Leo Burd wrote: > >> I'm new to Asterisk and I'm wonder what's the best way to combine multiple >> VOIP lines into a single phone number... >> > > Do you mean VOIP lines as in "numbers my customers can call to get to me" or > VOIP

RE: [Asterisk-Users] attended transfer issue

> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith > Sent: Wednesday, April 19, 2006 11:53 AM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] attended transfer issue > > On Wednesday 19 April 2006

RE: [Asterisk-Users] Codec problem from SIP to H323

Thanks for the answer ;) I'm using H323 (the one that comes under /channels/h323 with asterisk source). Before upgrading asterisk I prefer to try what you say (using OH323). Do you know which one is better? OH323 or ooH323c? (the second comes with asterisk-addons) Thanks again. Alejandro

RE: [Asterisk-Users] Codec problem from SIP to H323

Try to upgrade asterisk to version 1.2.4 Are you using OH323 or H323 ? I had same problem with 1.2.1 using H323(addon) , Installed 1.2.4 and OH323 and everything worked fine. Cheers, Oliver -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Me

Re: [Asterisk-Users] Error installing asterisk

yum install libidn-develOn 4/19/06, Luis Fernando Ramírez Cueva <[EMAIL PROTECTED]> wrote: I am instaling asterisk on Fedora core 3. I have instaled zaptel-1.2.3, libpri-1.2.2, but when I am instaling (make install) asterisk I have the following error:_GNU_SOURCE  -O6 -march=i6

[Asterisk-Users] Codec problem from SIP to H323

Hello. I have a codec problem to send calls from a SIP device to a H323 gateway. First I'll explain the scenario: - Asterisk 1.2.1 - The SIP phone can use any codec I want. - The H323 gateway can only use g729 (cause it's not under my administration) - SIP phone has g729 configured, so my asteris

[Asterisk-Users] Error installing asterisk

I am instaling asterisk on Fedora core 3. I have instaled zaptel-1.2.3, libpri-1.2.2, but when I am instaling (make install) asterisk I have the following error:_GNU_SOURCE  -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer  -fPIC   -c -o app_zapscan.o app_zaps

Re: [Asterisk-Users] Fwd: sip.conf and jump from register to the extension

Well, not sure about that. My register username is 44198w, I just took off the "w" for ease of demonstration, and it jumps into 200 fine. It wouldn't make much sense for the username have to be alphanumeric, with at least one number. Aaron On Wed, 19 Apr 2006, Thomas Winter wrote: I have

Re: [Asterisk-Users] Fwd: sip.conf and jump from register to the extension

I have done additional tests, because the documentation sample was not 100 % identical to my register command. OK: register => 44198:[EMAIL PROTECTED]/200 This jumps to 200, s is also working NOT OK: user:[EMAIL PROTECTED]/200 It looks for extension user and is ignoring 200 or anythink else

[Asterisk-Users] Voice mail issuse when pressing 0

An outside caller started to leave voice mail. The CLI shows: Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/sip/4232/INBOX/msg format: gsm, 0x8295d40 -- x=1, open writing: /var/spool/asterisk/voicemail/sip/4232/INBOX/msg format: wav, 0x829e2c0 -- User

Re: [Asterisk-Users] Fwd: sip.conf and jump from register to the extension

Have you tried sending it to a different extension number? I've got the registrations working on my home server where I register with one number and have it drop in on a totally different number in the context. register => 44198:[EMAIL PROTECTED]/200 Aaron On Wed, 19 Apr 2006, Thomas Winter

Re: [Asterisk-Users] Fwd: sip.conf and jump from register to the extension

Hi, [general] context=Sip_in register => 1234:[EMAIL PROTECTED]/s s is the same, it still looks for an extension 1234 in the context Sip_in and did not use /s Asterisk is 1.2.7 Am Wednesday 19 April 2006 22:48 schrieb Aaron Daniel: > I'm not gonna say much for the documentation, but I would

Re: [Asterisk-Users] Fwd: sip.conf and jump from register to the extension

I'm not gonna say much for the documentation, but I would suggest if you want to bypass that problem, add /s (or whatever extension) to the register statement so you know for absolute sure that incoming calls on the registration will go to the extension that you expect. Aaron On Wed, 19 Apr

[Asterisk-Users] RE: Delayed voice for 10 secs

Please post pertinent config files and a CLI output so the list can help with the 10 sec delay You set codec selection in SIP.conf. This selects preferred codec from top to bottom as well as jitter buffer settings and the RTP timeout. Sip.conf disallow=all allow=g729 allow=gsm allow=ulaw jitterbu

[Asterisk-Users] Fwd: sip.conf and jump from register to the extension

Hi, the documentation of sip.conf is telling me this: ;register => 1234:[EMAIL PROTECTED] ; ; This will pass incoming calls to the 's' extension In reality it jumps to the extension 1234 in the context and not to s So it is much more complicate to write an proper dialplan. Is this an bug o

RE: [Asterisk-Users] Ring a grop of extension, then playback a file, then transfer to external number

I do the same thing with outbound transfers.. Here is my code. exten => s,3,Playback(pls-wait-connect-call) I do this right before the dial command. - Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andre Courchesne - Consultant Sent: Wednesday,

[Asterisk-Users] Ring a grop of extension, then playback a file, then transfer to external number

Ok, Here is what I got working: A call comes in from a Zap line. 5 SIP extension ring if nobody picks up, the call is transfered to a cell phone number. That works. I not want to add a playback of a file ("Please waite while you are being transfered") before transfering the call to the

[Asterisk-Users] Re: SLIN format

In article <[EMAIL PROTECTED]>, Watkins, Bradley <[EMAIL PROTECTED]> wrote: > The OP was referring to how sox interprets filename extensions. In that > case, Kevin's .raw and .sw extensions are correct. That's what I get for always being in a hurry :-( Completely missed the reference to sox. Che

Re: [Asterisk-Users] clearing "stuck" channels without a restart

I have noticed a bug with the iax trunks that will hang if a firewall or something else network related changes routes or ip addressII am running CVS-v1-0-08/08/05-13:29:29 on the box. On 4/19/06, Bill Gibbs <[EMAIL PROTECTED]> wrote: I will try that next time, thanks!  This is the

[Asterisk-Users] Re: regexp in gotoif

On Fri, 7 Apr 2006 16:22:33 +0200 Christian B <[EMAIL PROTECTED]> wrote: > Hello! > > this is a short one: in a gotoif-statement i would like to match a > variable to a number, where the number could have digits from 2-6. > asterisk only seems to be capable to match such a digit-range when used >

RE: [Asterisk-Users] PRI blocking on incoming calls

We tried the pri intese debug last night but this only showed us what was going on with the Nortel trunk not the MCI T1 side. Sadly it was of no use. Any other ideas?   Anyone??   Thanks   From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oscar Carriles Sent: Tues

RE: [Asterisk-Users] clearing "stuck" channels without a restart

I will try that next time, thanks!  This is the first time it’s happened with a month of heavy usage and to be honest, I was testing out the hold/transfer/blind transfer stuff with Polycoms when my VPN kept locking up so that may have had something weird to do with it…   Bill  

Re: [Asterisk-Users] Asterisk 1.2.7.1 and IAX modem / channel

Pimjai Wesnarat wrote: Or does IAXModem not work with Asterisk 1.2.7.1? IAXmodem works just fine with Asterisk 1.2.7.1. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options vi

[Asterisk-Users] Asterisk 1.2.7.1 DTMF anomaly

Greetings, I'm using asterisk to connect our three locations together with a sort of inter-company auto attendant connected like this: PBX (fxs) <-> Sipura 3k (fxo) <-> Asterisk <-IAX-> remote asterisk It works like this: Person picks up their phone and dials a number to get to the auto att

Re: [Asterisk-Users] clearing "stuck" channels without a restart

you can do a soft hangup that shoud workIf it does not then you can enable callwaiting on the phones so they can atleast receive calls to the extentionOn 4/19/06, Bill Gibbs <[EMAIL PROTECTED]> wrote: 192.168.1.107    199 6bd3fb49505  00102/0  ulaw  No   Tx: ACK  

[Asterisk-Users] Asterisk IVR / Scalability

Hi i am looking for a good ivr system for my company. these are my question are there any good ivr's that can be easily integrated with asterisk ? and are there any  large scale deployment of asterisk to date ? thanks Giridhar Bandi ___ --Bandw

Re: [Asterisk-Users] Re: Asterisk and 7960s

I've been following this thread and it's clear is mud... Would someone care to summarize? Is it possible to automatically display the caller's number (true ANI in my case), caller name and caller address on a 7960 that's running 8.2? We currently rewrite CID Number and CID Name with a PHP sc

Re: [Asterisk-Users] Outgoing voice distortion with Unicall

On Wed, 2006-04-19 at 10:58 -0500, Rich Adamson wrote: > Carlos Chavez wrote: > > On Wed, 2006-04-19 at 08:37 +0200, Stepan Hradsky wrote: > >> Hi, > >> > >> I had similar problem and problem was in SIP ATA device (we use Sipura > >> 2100). They was set from factory to send 30ms voice frame, > >>

[Asterisk-Users] Meetme codec translation and callerID library.

Can Meetme be made to work with G.729? (I gather not) If a call comes in (internally or externally), the call comes in as a G.729 call, which then re-negotiates to a G.711u call when if gets transferred to a MeetMe room. Is there a way to set up asterisk that will allow me to have internal phones

Re: Re: [Asterisk-Users] Call Center with No TDM components

 [EMAIL PROTECTED] ~]# cat /proc/loadavg> > > > >  1.52 1.19 1.07 1/168 25019 This was with 5 agents with 1.2 pacing ratio and linux runlevel init 3 with only essential daemons running. Machine : P4 HT 3.4 GHz  // I GB // 80 GB SATA from HP Linux: 2.6.11 SMP Asterisk:1.2.5 net-sec Vicidial: 1.1.1

Re: [Asterisk-Users] Re: Asterisk and 7960s

The @ip-address is actually a documented cisco "fix" to another problem. I'd have to look it up, cause I don't remember exactly what it was, but it's been on the list somewhere, and I think EVERYONE that's used 8.2 has the same problem with the firmware. I would suggest using 7.4 or 7.5.

Re: [Asterisk-Users] attended transfer issue

On Wednesday 19 April 2006 13:32, Damon Estep wrote: > Understand, but the point was that every SIP device has its own method, and > it would be nice if asterisk had a blind/attend transfer feature as > described so we are not dependent on the SIP UA vendors to try and > normalize the world. If ast

RE: [Asterisk-Users] attended transfer issue

From: [EMAIL PROTECTED] on behalf of Eric "ManxPower" Wieling Sent: Wed 4/19/2006 8:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] attended transfer issue John Novack wrote: > > > Eric "ManxPower" Wieling wrote

[Asterisk-Users] clearing "stuck" channels without a restart

192.168.1.107    199 6bd3fb49505  00102/0  ulaw  No   Tx: ACK    192.168.0.100    110 5c5a4953-65  00101/5  ulaw  Yes  Rx: ACK      Those channels are stuck talking to each other.  The phones are disconnected yet that connection remains.  I can clear

[Asterisk-Users] Where to buy Eicon DIVA cards

Hi! Can someone recommend a Eicon DIVA cards distributor in Austria/Europe? thanks klaus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/l

RE: [Asterisk-Users] Re: SLIN format

The OP was referring to how sox interprets filename extensions. In that case, Kevin's .raw and .sw extensions are correct. - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, April 19, 2006 12:56 PM To: asterisk-user

RE: [Asterisk-Users] Asterisk service crashes

asterisk -g didn't work. It crashed as well. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Wednesday, April 19, 2006 9:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk service crashes -c d

[Asterisk-Users] Re: SLIN format

In article <[EMAIL PROTECTED]>, Kevin P. Fleming <[EMAIL PROTECTED]> wrote: > Steve Kennedy wrote: > > In sox terms is SLIN .ul (as in unsigned linear). > > No. ul is ulaw. SLINEAR is .raw, or .sw (signed word). According to format_sln.c, it is .raw or .sln But IMO standard .wav format with mono

Re: Faxing and PCI (was Re: [Asterisk-Users] Digium cards,

Olivier Krief wrote: 1. Does using iaxmodem imply that, for example, incoming fax calls are processed that way ? - PSTN calls are terminated on TDM board (Digium, Sangoma, ...) Yes. The PSTN call terminates there and (usually) takes a short IAX2 hop over the loopback adapter to iaxmodem.

Re: [Asterisk-Users] bad voice quality

for the Non-commercial version of G729? Try this wget http://kvin.lv/pub/Linux/Asterisk/codec_g723-gcc-pentium2.so wget http://kvin.lv/pub/Linux/Asterisk/codec_g729-gcc-pentium2.so On 4/18/06, Alex Mosburger <[EMAIL PROTECTED]> wrote: There is a free version of G.729 available? I would be very inte

Re: [Asterisk-Users] What's the best way to combine multiple VOIP lines into a single number?

On Wednesday 19 April 2006 12:23, Leo Burd wrote: > I'm new to Asterisk and I'm wonder what's the best way to combine multiple > VOIP lines into a single phone number... Do you mean VOIP lines as in "numbers my customers can call to get to me" or VOIP lines as in "multiple providers I can call ou

[Asterisk-Users] How to route all incoming calls on an analog trunk to a specific ring group

I have a customer that is using a digium tdm2400p card with multiple analog lines coming in. They want to be able to forward all calls on specific trunks to specific ring groups. I tried this with the inbound routing and specifying the phone number of the trunk and directing it to a ring group b

Re: [Asterisk-Users] Callerid matching in extensions.conf

On 09:34, Wed 19 Apr 06, Douglas Garstang wrote: > I'm using Asterisk 1.2.7.1. Has the callerid number matching functionality in > extensions.conf changed recently? > > exten => ,1,NoOp(${CALLERID}) > > hestia*CLI> > -- Executing NoOp("SIP/2944093-d24d", ""Cletus the Slaw Jawed Yokel"

[Asterisk-Users] What's the best way to combine multiple VOIP lines into a single number?

Hello there, I'm new to Asterisk and I'm wonder what's the best way to combine multiple VOIP lines into a single phone number... I currently have 4 incoming VOIP lines, each one with a different number. Ideally, I would love to provide my customers with a single number for all those lines.

[Asterisk-Users] Asterisk 1.2.6 and 9133i

Hi, I'm running asterisk 1.2.6 and aastra phones 9133i with the latest firmware. The MWI light is not coming on on the phones.A) how can I debug if asterisk is actually sending the 'new message' notification? Is it possible I could have this turned off by accident? Has anyone else had succes

Re: [Asterisk-Users] SLIN format

Steve Kennedy wrote: > In sox terms is SLIN .ul (as in unsigned linear). No. ul is ulaw. SLINEAR is .raw, or .sw (signed word). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit

Re: [Asterisk-Users] Unable to allocate socket: Too may open files

Stefan Günther wrote: Hello, we are curently benchmarking an asterisk system 1034 sip users are logged into this system and the test software is trying to establish 400 concurrent calls. In the CLI I see the following messages: Apr 19 14:20:51 WARNING[4045]: rtp.c:911 ast_rtcp_new: Unable t

[Asterisk-Users] Delayed voice for 10 secs

Hi List !!I have a lot a questions about this incredible tool but short is my time to learn it, so I apologize if my last question was too general. I got another more especific trouble. I administrating an ISP and I have my Asterisk installed on a server for testing my network performance. I follow

Re: [Asterisk-Users] Outgoing voice distortion with Unicall

Carlos Chavez wrote: On Wed, 2006-04-19 at 08:37 +0200, Stepan Hradsky wrote: Hi, I had similar problem and problem was in SIP ATA device (we use Sipura 2100). They was set from factory to send 30ms voice frame, when we change frame to 20ms everything work perfectly. Where in the S

Re: [Asterisk-Users] Music on Hold bug? User disconnect Sip user agent

Johann Steinwendtner wrote: Did you try rtpholdtimeout in sip.conf ? Just tried it with rtpholdtimeout=60 did a reload from the console, and tried again. Unplugging the phone and sitting on hold for 3 minutes. Never disconnected. Just a reminder, I'm doing this over an IAX trunk to a SIP ph

RE: [Asterisk-Users] Asterisk service crashes

Do I stop the asterisk service first and then do asterisk –g?   If I try asterisk –g while the service is running, I get:   [EMAIL PROTECTED] bpiper]# asterisk -g Asterisk already running on /var/run/asterisk/asterisk.ctl.  Use 'asterisk -r' to connect. [EMAIL PROTECTED] bpiper]#  

Re: [Asterisk-Users] Re: Asterisk and 7960s

Thanks very much to you both for helping me with this issue. On 4/19/06, Aaron Daniel <[EMAIL PROTECTED]> wrote: > > Thanks for letting me know regarding the "@ip-address" problem. I take > > it you have experienced something similar with this firmware? > The @ip-address is actually a documented c

[Asterisk-Users] SLIN format

In sox terms is SLIN .ul (as in unsigned linear). Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___

[Asterisk-Users] Callerid matching in extensions.conf

I'm using Asterisk 1.2.7.1. Has the callerid number matching functionality in extensions.conf changed recently? exten => ,1,NoOp(${CALLERID}) hestia*CLI> -- Executing NoOp("SIP/2944093-d24d", ""Cletus the Slaw Jawed Yokel" <2944093>") in new stack == Auto fallthrough, channel 'SIP/29

Re: [Asterisk-Users] Asterisk service crashes

> Here is a ps aux of the services while the server is crashed. Does anyone see any service that would have a conflict with the asterisk service? H, maybe just me, but I personally wouldn't run anything but asterisk on my server, yet alone adding... cups server font server http serve

Re: [Asterisk-Users] Outgoing voice distortion with Unicall

On Wed, 2006-04-19 at 08:37 +0200, Stepan Hradsky wrote: > Hi, > > I had similar problem and problem was in SIP ATA device (we use Sipura > 2100). They was set from factory to send 30ms voice frame, > when we change frame to 20ms everything work perfectly. > Where in the Sipura configur

Re: [Asterisk-Users] Re: Asterisk and 7960s

On 15:08, Wed 19 Apr 06, dave wrote: > The external directory is part of the cisco phones, nothing to do with > * really. The internal extensions I have set up all work fine with the > name using the callerid="Joe Blow" <1234> method you suggest. The plan > was to add customers phone numbers to the

Re: [Asterisk-Users] Call Center with No TDM components

On Wed, 19 Apr 2006, Abhimanyu Rapria wrote: > Transcoding and Recording is being done at > Dialer and load average is > 1.5 for 12 agents and pacing of 1.1 to > 1.2 What is the average CPU utilization you observe with these load averages? Regards, Gerald.

Re: [Asterisk-Users] Polycom 501 resource full problems ...

You can change the storage method on the Polycom phones from using NVRAM to VRAM to increase the number of entries (limited to 25 with NVRAM according to the Polycom Admin Guide) that a phone can store. The relevant setting is dir.local.volatile.2meg=1 or dir.local.volatile.4meg=1, dependin

Re: [Asterisk-Users] Sip channel variables

There are several approaches: 1. Set in Asterisk DB RECORD/${EXTEN} on and on outgoing calls check is set for that user than do recording otherwise skip Monitor cmd in dialplan. 2. Set account code in sip.conf for certain user and in dialplan you could check that variable ${ACCOUNTCO

Re: [Asterisk-Users] Music on Hold bug? User disconnect Sip user agent

Did you try rtpholdtimeout in sip.conf ? Hans Marco Mouta schrieb: How do I report a Bug to Digium? or asterisk project? On 4/19/06, *Doug Lytle* <[EMAIL PROTECTED] > wrote: Marco Mouta wrote: > I've tested maxexpirey=120 and even with this, asterisk didn'

Re: [Asterisk-Users] Phones that work well through NAT

On Tuesday 18 April 2006 21:06, Sean Garland wrote: > So I have * box shorewall/linux NAT firewall internet - > WRT54G with openwrt - IP500 > > I have 5060, 4569, and 1 through 2 forwarded to * box from > internet. I have tried everything I can think of on the wrt to ge

Re: [Asterisk-Users] Re: Asterisk and 7960s

Thanks for letting me know regarding the "@ip-address" problem. I take it you have experienced something similar with this firmware? The @ip-address is actually a documented cisco "fix" to another problem. I'd have to look it up, cause I don't remember exactly what it was, but it's been on the l

RE: [Asterisk-Users] PRI caller ID

Perpahs you need a Wait(2) before Answer() in the dialplan, because telco send CallerIDName after some time (second ring, I suppose). Mimmus > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Jonathan k. Creasy > Sent: Wednesday, April 19, 2006 3:

[Asterisk-Users] Sip channel variables

Hi,   I am making a Dialplan in which I have to record some outgoing calls from some users and not from the others.   I made to outgoing calls Macros, when that records calls (using Monitor()), and another that doesn't: [macro-outgoinglocal] and [macro-outgoinglocalrecord]   Is it possible

RE: [Asterisk-Users] Music on Hold bug? User disconnect Sip user agent

Http://bugs.digium.com     From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marco MoutaSent: Wednesday, April 19, 2006 10:38 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Music on Hold bug? User disconnect Sip user

[Asterisk-Users] Re: Asterisk and 7960s

>Thanks for letting me know regarding the "@ip-address" problem. I take >it you have experienced something similar with this firmware? Yup. Due to this I only run 8.2 on my phone, and use 7.4 on any other. You can roll back to 7.4 and be ok, don't use 7.5 as it has a bug relating to the phone regi

Re: [Asterisk-Users] attended transfer issue

John Novack wrote: Eric "ManxPower" Wieling wrote: John Novack wrote: Damon Estep wrote: There is some kind of issue with SIP transfer interaction between some SIP phones and asterisk, I have personal experience with Polycom phones not being able to do a blind xfer using the feature ke

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