[Asterisk-Users] Re: Record inbound and outbound calls to and from one number.

Tim Mattison wrote: > Good call. > > For our American readers... does anyone know where I can obtain a list > of states/counties and their regulations in regards to call recording? > > On Sun, 2005-01-30 at 22:10 +, Mike Dent wrote: > > or maybe country? or should that be County? :) > > > >

Re: [Asterisk-Users] Japan

Eicon Diva Server BRI = ISDN I think.. JAson Leo Ann Boon wrote: Jason Frisch wrote: Hi all, I am trying to setup Asterisk here in Japan in my office. However I am having a hard time finding hardware that is supported. I tried Voicetronix but they said that they are too busy to create a driver. I

Re: [Asterisk-Users] Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 crashes with Ouch ... error while writing audio data: : Broken pipe

On Sun, 30 Jan 2005, Martin List-Petersen wrote: Citat Remco Barende <[EMAIL PROTECTED]>: I have Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 up and running. Everything seems to be running fine but after some time asterisk just goes crazy (even withouth any incoming or outgoing call activity perviously). If

Re: [Asterisk-Users] Japan

Jason Frisch wrote: I asked Softbank and it seems that using SIP etc directly is not an option. Something to do with theVoIP-TA being used for communications between the providers "call-agent". Sounds like they're using MGCP. At this point, Asterisk is not able to act as an MGCP endpoint, it can

Re: [Asterisk-Users] Japan

Jason Frisch wrote: Hi all, I am trying to setup Asterisk here in Japan in my office. However I am having a hard time finding hardware that is supported. I tried Voicetronix but they said that they are too busy to create a driver. I knew someone in Japan who had a working Asterisk + Eicon Diva Ser

[Asterisk-Users] Trunked IAX or not

Has anyone benchmarked Asterisk on a dedicated single versus dual processor machine? http://www.astertest.com/ Cheers, Philipp The test results that Philipp pointed out show some protocol comparisons that include "iax2 trunking / alaw" and "iax2 / alaw" and concludes that "IAX2 trunking is more t

Re: [Asterisk-Users] PRI for Data and Voice

Dave, howzit You can use asterisk with a quad E1 card to divide your E1. So anyone who dials in using 1234 for example, route to your portmaster and anyone who dials in using 1235 use for IVR/voip, whatever. Good luck Regards Clive On 29 Jan 2005 at 15:11, David Norton wrote: > > Hi, > > Curren

Re: [Asterisk-Users] Japan

I asked Softbank and it seems that using SIP etc directly is not an option. Something to do with theVoIP-TA being used for communications between the providers "call-agent". Jason Steven Critchfield wrote: On Mon, 2005-01-31 at 12:53 +0900, Jason Frisch wrote: Sorry for my ignorance, but what is

Re: [Asterisk-Users] DIAX softphone - Asterisk server rejecting

Hi, - Original Message - From: "Pradhip KCL" <[EMAIL PROTECTED]> Hi all trying some basic functinality I was able to download asterisk and complie. I have downloaded DIAX soft phone also. I am trying to make an internal call between two softphones. I read the http://www.voip-info.org/wiki

Re: [Asterisk-Users] how to stop ringing after congestion.

Jon Gabrielson wrote: When there are no zap channels available, I signal congestion using the following: exten => _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _9NXX,2,Playtones(congestion) exten => _9NXX,3,Congestion The congestion sound plays correctly, but the ringing continue

Re: [Asterisk-Users] asterisk tries to dial out on lines already in use.

On Sun, 30 Jan 2005 18:14:38 -0600, Jon Gabrielson <[EMAIL PROTECTED]> wrote: > Asterisk should be able to do this, there are several cases > when this is essential. The first is a shared/party line where > asterisk cannot have guaranteed access for whatever reason. > In our case, that reason happ

[Asterisk-Users] how to stop ringing after congestion.

When there are no zap channels available, I signal congestion using the following: exten => _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten => _9NXX,2,Playtones(congestion) exten => _9NXX,3,Congestion The congestion sound plays correctly, but the ringing continues in the background

Re: [Asterisk-Users] Zap channels in AU hanging up on STD pips

Hi Howard, Which provider are you with? We're with Primus Business here in Melbourne, and haven't had anything like what you're describing. For reference, here's a snip of my zapata.conf: [channels] language=en context=local signalling=fxs_ks usecallerid=no echocancel=yes echocancelwhenbridged=y

[Asterisk-Users] Zap channels in AU hanging up on STD pips

Is anyone having/had a problem with a TDM400P card hanging up on STD outbound calls as soon as the called party answers. I'm guessing that * is responding to the STD pips in some way. -- Howard. LANNet Computing Associates; Your Linux people -

Re: [Asterisk-Users] Japan

On Mon, 2005-01-31 at 12:53 +0900, Jason Frisch wrote: > Sorry for my ignorance, but what is J1? I actually hope to use Softbanks > fiber-based IPtel > service, but I believe they require VoIP TA so I guess the end result is > just a standard > analog line. J1 is a Japanese T1 or close equivalen

Re: [Asterisk-Users] Monitor calls timeout

On Mon, 31 Jan 2005 09:26:05 +0800, el Flynn <[EMAIL PROTECTED]> wrote: > did you try setting using AbsoluteTimeout in the context? e.g. > > exten => s,1,Answer > exten => s,2,AbsoluteTimeout(0) > exten => s,3,Monitor(wav,testrecod,m) Thanks for the suggestion, but it's no good. It still times o

[Asterisk-Users] DIAX softphone - Asterisk server rejecting

Hi all trying some basic functinality I was able to download asterisk and complie. I have downloaded DIAX soft phone also. I am trying to make an internal call between two softphones. I read the http://www.voip-info.org/wiki-Asterisk+installation+tips website to configure channels and extensions

Re: [Asterisk-Users] detailed asterisk howto

http://www.voip-info.org/wiki-Asterisk+installation+tips this might help you On Mon, 31 Jan 2005 14:49:25 +1100, Duane <[EMAIL PROTECTED]> wrote: > szj wrote: > > Hi, all: > > > > I am a newbie to the asterisk and its architecture. :( > > After reading some help in the tarball of Asterisk, I am

Re: [Asterisk-Users] Slackware + Asterisk + asterisk-addons

There were some changes recently to the internal structure of the realtime config. Make sure you have the latest CVS copy of both asterisk and asterisk-addons and it should fix the compile error with res_config_mysql. On Sun, 2005-01-30 at 23:16 -0500, Bobby Lacey wrote: > Hello > > I am tryin

Re: [Asterisk-Users] CAC Access Bank

Matt Schulte wrote: We have an old CAC and we're trying to get groundstart working on it, we think it may be a dip switch setting. Does anyone have the config settings and/or the manual to config this thing? Any help is greatly appreciated :-) I have a CAC Access Bank 1 with 24 fxs configured as fo

[Asterisk-Users] Slackware + Asterisk + asterisk-addons

Title: Message Hello   I am trying to get asterisk-addons installed so that I can use the mysql cdr feature. OK, I have the MySQL server (mysqld) installed, but I noticed that mysql-devel is also required. I tried to compile asterisk-addons and got a:   --CUT--- res_config_mysql.c:422: error:

Re: [Asterisk-Users] Japan

Sorry for my ignorance, but what is J1? I actually hope to use Softbanks fiber-based IPtel service, but I believe they require VoIP TA so I guess the end result is just a standard analog line. Jason Cory Andrews wrote: Jason - I believe the Sangoma T1/E1/J1 boards may work in Japan, I will chec

Re: [Asterisk-Users] detailed asterisk howto

szj wrote: Hi, all: I am a newbie to the asterisk and its architecture. :( After reading some help in the tarball of Asterisk, I am still in the mess. So I want to know where I can find a detailed explanation of the Asterisk which including the Architecture, Install, Configure, usage example docu

Re: [Asterisk-Users] x100P wildcard discontinued ?

Varun - Model TDM01B is the replacement for the X100P, which has I believe been discontinued by Digium. Cory Andrews Senior Partner VOIPSupply.com + 800.398.VOIP X22 [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello, Is the x100P wildcard discontinued ? If yes then which card rep

[Asterisk-Users] detailed asterisk howto

Hi, all: I am a newbie to the asterisk and its architecture. :( After reading some help in the tarball of Asterisk, I am still in the mess. So I want to know where I can find a detailed explanation of the Asterisk which including the Architecture, Install, Configure, usage example document. Maybe

Re: [Asterisk-Users] Japan

Jason - I believe the Sangoma T1/E1/J1 boards may work in Japan, I will check and see if they offer driver support for J1, I do not know of an analog solution. Cory Andrews Senior Partner VOIPSupply.com + 800.398.VOIP X22 [EMAIL PROTECTED] Jason Frisch wrote: Hi all, I am trying to

Re: [Asterisk-Users] Asterisk@home and Zap Channels

At 08:46 AM 1/30/2005, you wrote: AMP does not support ZAP entensions. only sip and iax. Maybe in a future release. You could post to the AMP site and ask for this feature. For now you will need to hack extensions.conf to get it to work. Thanks to everyone for the help, I have gotten it to answer i

[Asterisk-Users] x100P wildcard discontinued ?

Hello, Is the x100P wildcard discontinued ? If yes then which card replaces the x100P card ? Thanks Varun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or up

[Asterisk-Users] x100p issues + TDM400P

Hello, I have been wanting to use digium x100p to get started. But it seems to have compatibility issues for different regions. I am refering to the " 600 ohm US pstn standard only". I am in India so If I were not to use x100p card then what card I need to go in for? I also read that x100

[Asterisk-Users] Hitting IOCTL??

I have recently started seeing the following message a lot: "We have hit out IOCTL" Can someone please explain what this means and/or how to fix it? It just recently started appearing and seem to mainly come from when I hang up an analog extension on a TDM400 card FXS port. Robert

[Asterisk-Users] Japan

Hi all, I am trying to setup Asterisk here in Japan in my office. However I am having a hard time finding hardware that is supported. I tried Voicetronix but they said that they are too busy to create a driver. If anyone has had success with any particular hardware please let me know. I plan to hav

RE: [Asterisk-Users] where to buy x100p

I might also add some of the clone cards use an Intel chip. Those clones that were sold in the US use one chip set, while those sold in some other countries used a similar (but different number) chip set. Therefore, some x100p clones _might_ work fine in non-600 ohm countries, but its likely becaus

Re: [Asterisk-Users] widcard x100P doubt

- Original Message - From: Lyle Giese <[EMAIL PROTECTED]> Date: Sunday, January 30, 2005 7:50 pm Subject: Re: [Asterisk-Users] widcard x100P doubt > Kewlstart is part of the spec for forward disconnect on loopstart > lines in > North America(I am not familar with telco specs elsewhere i

Re: [Asterisk-Users] asterisk tries to dial out on lines already inuse.

It sounds like you have multiple devices sharing the same physical lines? I think you will continue to have problems until you can rearrange the setup to avoid line sharing to allow Asterisk to have dedicated access. Might have more luck with ISDN. Craig - Original Message - From: "Jon

Re: [Asterisk-Users] Processing incoming calls with multiple contextstover PRI

In zaptel.conf, put the line associated with 8350 in the context bpns-external and when an external call comes in on 8350, it will drop to the s step in bpns-external.  I would suggest that you do something with the call if they don't bother to dial an extension, like send to a general voice

Re: [Asterisk-Users] Processing incoming calls with multiple contextstover PRI

Hi,Jason,     The TDM400P card failed to get the Callee number or DID, so the * don't know how to route the call. There are something difference between the analog line and the PRI line.       Regards.       David     http://www.iaxtalk.com   - Original Message -

Re: [Asterisk-Users] Processing incoming calls with multiple contextst over PRI

Jason Brown wrote: So I have a problem. A customer of mine wants a PBX, owns an office building. I want to sell him on asterisk. He has 4 tenants. I am using my asterisk box to simulate it. My asterisk box has a TDM400P card, not a PRI card. Don't know if it makes any difference. Just a guess abo

Re: [Asterisk-Users] Processing incoming calls with multiple contextst over PRI

Jason Brown wrote: Now I understand it is looking for the startup point. I don’t understand why. 2 other asterisk guys I know swear it’s supposed to work, although they are using sip/iax and not zap for input. And why would you think those would act similarly? They don't. Zap channels without ISD

[Asterisk-Users] Meetme2 web - nothing happens on click ?

Hi, I've installed meetme2 according to instructions. Everything seems ok, members of conference are displayed, but nothing happens if I click on 3 action buttons (kick out, talk&listen, ...). Any hints how to deal with this ? In what way exactly does meetme2 kick user off the conference ? Tha

[Asterisk-Users] Processing incoming calls with multiple contextst over PRI

So I have a problem. A customer of mine wants a PBX, owns an office building. I want to sell him on asterisk.  He has 4 tenants. I am using my asterisk box to simulate it. My asterisk box has a TDM400P card, not a PRI card. Don’t know if it makes any difference.   Anyway, I want to route

Re: [Asterisk-Users] Caller ID in AU

I have updated the Wiki with this info as I have seen it come up a few times. Nathan. Gary wrote: Don't forget Howard, that Caller-ID presentation is an extra chargeable service. has it been turned on on these lines and confirmed ?? (its handy to carry a caller-id in your kit for checking:-) On S

Re: [Asterisk-Users] Monitor calls timeout

jurgen wrote: Problem is, Asterisk times out and disconnects after 10 seconds, stopping the recording. If I run something else in the context, say the infamous Monkey Sounds, everything's fine, and the call just keeps going, annoying the people on the line with monkey sounds. For some reason, the

Re: [Asterisk-Users] New Firefly version

Adam Hart wrote: Few people have claimed success, I'm not sure how though. Any chance of a native linux version then? :) -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunicati

Re: [Asterisk-Users] New Firefly version

Duane wrote: Adam Hart wrote: As always, I'm happy to announce a new version of Firefly. Firefly 1.9.8 has more of what you want and less of what you don't http://www.virbiage.com/firefly/download/firefly-thirdparty.exe There's a few bug fixes - notably fixed the Reject button and sending of audio

Re: [Asterisk-Users] asterisk tries to dial out on lines already in use.

Asterisk should be able to do this, there are several cases when this is essential. The first is a shared/party line where asterisk cannot have guaranteed access for whatever reason. In our case, that reason happens to be because we also use our outgoing lines for faxing. The second is that witho

[Asterisk-Users] OH323 compile error : CVS-HEAD

I am getting the following error when compiling oh323-0.7.1 with Asterisk CVS (2004-12-21: Updated versions 0.7.1 (for Asterisk CVS HEAD) make[1]: Leaving directory `/usr/src/asterisk-oh323-0.7.1/wrapper' make: *** [subdirs_build] Error 1 bash-2.05b# asteriskaudio.cxx:167: (Each undeclared identi

Re: [Asterisk-Users] Trying to make but it fails

On Sunday 30 January 2005 23:29, [EMAIL PROTECTED] wrote: > I get these errors, and i am stuck here. I don't know what to do. > I am using latest stable Zaptel 1.0.4 and Asterisk 1.0.5. I just ran it on > a freinds machine and it went well there. [...] > chan_zap.c:3669: dereferencing pointer to in

[Asterisk-Users] Monitor calls timeout

Hi all, We're in a transition between OldPhoneSystem and Asterisk. One of the things that's needed to be done right now with OldPhoneSystem is the ability to record calls. I thought "Asterisk can record calls", so I set about to make it happen. And it does, sort of. I made a .call file that rings

[Asterisk-Users] Trying to make but it fails

I get these errors, and i am stuck here. I don't know what to do. I am using latest stable Zaptel 1.0.4 and Asterisk 1.0.5. I just ran it on a freinds machine and it went well there. chan_zap.c:3647: dereferencing pointer to incomplete type chan_zap.c:3648: dereferencing pointer to incomplete typ

Re: [Asterisk-Users] Record inbound and outbound calls to and from one number.

Good call. For our American readers... does anyone know where I can obtain a list of states/counties and their regulations in regards to call recording? On Sun, 2005-01-30 at 22:10 +, Mike Dent wrote: > or maybe country? or should that be County? :) > > Mike > > > > On Sun, 30 Jan 2005 16

[Asterisk-Users] conference room capacity question

hi; have couple of questions regarding meet_me conference room application: 1) is there a maximum allowable number of concurrently active conference rooms per server? 2) what is the maximum allowable number of users in a given conference room before quailty creeps out? thanks moe smadi _

Re: [Asterisk-Users] Record inbound and outbound calls to and from one number.

or maybe country? or should that be County? :) Mike On Sun, 30 Jan 2005 16:49:29 -0500, Tim Mattison <[EMAIL PROTECTED]> wrote: > Depending on your state. :P > > On Sun, 2005-01-30 at 11:42 -0500, Mark Phillips wrote: > > Don't forget to warn your callers about the recording. > > > > Tim Matt

Re: [Asterisk-Users] asterisk tries to dial out on lines already in use.

On Sun, 2005-01-30 at 13:40 -0600, Jon Gabrielson wrote: > Can't asterisk look for a dialtone? Even a $5 modem > can detect whether or not there is a dialtone. Maybe you should just use your $5 modem and write your own software. Asterisk is a PBX. PBXs shouldn't have to deal with your bastardiz

Re: [Asterisk-Users] where to buy x100p

The above assumes the x100p is connected to a pstn network that is reasonably close to the 600 ohm US standard the card was designed to interface with. Using the card in several other countries where the pstn standards are different will likely result in echo that can not be addressed with the abo

Re: [Asterisk-Users] Record inbound and outbound calls to and from one number.

Depending on your state. :P On Sun, 2005-01-30 at 11:42 -0500, Mark Phillips wrote: > Don't forget to warn your callers about the recording. > > Tim Mattison wrote: > > Try the monitor application instead of record. I think that'll do what > > you're looking for. > > > > On Fri, 2005-01-28 at

[Asterisk-Users] 302 Moved temporarily problem / Sipura 3000

  I can send calls from asterisk to a Sipura FXO interface (SIP/300) from any SIP phones including SIP/205 which is the Sipura 3000 FXS interface.   The problem I have is when a call from the PSTN is sends to Asterisk. On extnesion conf I dial all the SIP clients I get a 302 Moved temporaril

Re: [Asterisk-Users] Polycom changing policy - allowing firmwaredownloads?

On Jan 29, 2005, at 1:21 AM, Tim Courcy wrote: Anyone can register on the PRC and get access to the Manuals. But you have to be a certified reseller to get access to the firmware.. Gee, that would make me stay away from their phones -mark ___ Asteris

[Asterisk-Users] Caller ID on H323

Hi Friends   I have a problem presenting Caller ID on my H323 GW.     Scenario:  Sip Phone à à Asterisk à à H323 GW à à PSTN (E1)   From PSTN to the Sip phone works fine I put this lines in extentions.conf exten => 1234,1,SetCallerID, “${CALLERIDNUM}” ex

RE: [Asterisk-Users] Digium and Intel Chipset compatability

I spent several days last year trying to get a TE410 to work on an HP DL360, but never seemed to get it to work. I called Digium's (usually great) tech support at the time, and their response to me was 'if you can't get it to work we can't help you'. Please let me know if you have any success,

[Asterisk-Users] One way call when the * server and phone in a local network

Hi everyone, I started playing with Asterisk server a few days ago. So far, I only have made it 50% work. Here is my situation, IP phone A and Asterisk server are in a same local network behind an ADSL router (public ip = a.b.c.d), and the * server is set as the DMZ host of the router. IP phone

Re: [Asterisk-Users] International Order for Grandstream and Sipura

Dhennys , Estou interessado no SIPURA também. Você já tem alguma idéia em como adquirir? Estou em São Carlos, SP. Podemos, eventualmente, dividir custos de envio. -- Antonio José dos Santos Brandão On Sat, 29 Jan 2005 05:20:30 -0200, Dhennys Pestana <[EMAIL PROTECTED]> wrote: > (I appologize

RE: [Asterisk-Users] where to buy x100p

Hi! > wow! excellent information, this should be added to the wiki under > x100p / tdm400. So did _you_ add it? Cheers, Philipp ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users T

Re: [Asterisk-Users] Single or Dual Processor? High volume MeetMe

Hi! > Has anyone benchmarked Asterisk on a dedicated single versus dual > processor machine? http://www.astertest.com/ Cheers, Philipp ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-

[Asterisk-Users] Callgroup with bristuff ISDN?

Hi list! I'm still trying to figure out about the groups in asterisk. If I understand correctly, if you assign a certain group number and you assign the same call group number to a sip device the device will reing even though you did not specifically specify it in extension.conf? How will this w

[Asterisk-Users] IAX2 firmware for PA168x (Giptel G100, Siptronic ST-100 etc)

Hi there, this is just a short note about one of the PA168x based phones out there which I obtained as "Giptel G100" (aka Siptronic ST-100): For some reason this phone would refuse to register with Asterisk using SIP, but after uploading the IAX2 firmware instead it finally came to life: http:

Re: [Asterisk-Users] Caller ID spoofing

Calin Serbanescu schrieb: unfortunatelly i have to accept their terms and rewrite caller id... but again, i am newbie in scripting with agi and i can't find any example on the net about this... do you have any link to such script ? ... what I should maybe also mention: My script in the recent emai

Re: [Asterisk-Users] Caller ID spoofing

thanks very much, i'll try that... that is, no sleep until it's ready :) On Sun, 2005-01-30 at 21:17 +0100, Roger Schreiter wrote: > Calin Serbanescu schrieb: > > ... > > the net about this... do you have any link to such script ? > > > No, I don't have such a link. > But on the voip-wiki pages

Re: [Asterisk-Users] Caller ID spoofing

Calin Serbanescu schrieb: ... the net about this... do you have any link to such script ? No, I don't have such a link. But on the voip-wiki pages there are some examples for agi-scipting. There are APIs for some common languages, e.g. Perl, which is maybe one of the fastet ways to code simple scr

Re: [Asterisk-Users] asterisk tries to dial out on lines already in use.

i have the same problem... i've also added a feature request to bug tracker (http://bugs.digium.com/bug_view_page.php?bug_id=0002612) regarding this issue. On Sun, 30 Jan 2005 13:40:06 -0600, Jon Gabrielson <[EMAIL PROTECTED]> wrote: > Can't asterisk look for a dialtone? Even a $5 modem > can d

Re: [Asterisk-Users] Caller ID spoofing

unfortunatelly i have to accept their terms and rewrite caller id... but again, i am newbie in scripting with agi and i can't find any example on the net about this... do you have any link to such script ? thanks On Sun, 2005-01-30 at 20:42 +0100, Roger Schreiter wrote: > Calin Serbanescu schrieb

Re: [Asterisk-Users] Caller ID spoofing

Calin Serbanescu schrieb: ... e164 numbers in my h.323 network and my ISDN provider doesn't accept those identities (CIDs). So, i have to spoof the outgoing CID depending on incoming CID. Is there any possible way of doing this by AGI? How? any examples are welcome. Hi, I'm not sure, whether I'v

Re: [Asterisk-Users] asterisk tries to dial out on lines already in use.

Can't asterisk look for a dialtone? Even a $5 modem can detect whether or not there is a dialtone. Thanks, Jon. On Sunday 30 January 2005 10:37 am, Wilson Pickett wrote: > > When I place a call with asterisk, asterisk will try to dial > > out on the first line even if the first line is alread

[Asterisk-Users] Single or Dual Processor? High volume MeetMe

Has anyone benchmarked Asterisk on a dedicated single versus dual processor machine? Or could any Asterisk developers comment on whether it is architected in such a way that threads could run on multiple CPUs (especially MeetMe2)? At a higher level, can I host more simultaneous lines and/or c

Re: [Asterisk-Users] Vocera Badges

John Middleton wrote: Anyone got any experiences of these with *, and also costings? Someone mentioned them on the list several months ago, but I don't think anyone mentioned actually using it. -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users m

[Asterisk-Users] Caller ID spoofing

Hello everybody! I am having the following problem and since I am a beginner in playing with asterisk, i can't solve it: I am trying to integrate my existing H.323 network in real world telephony by ISDN cards. The problem is that i DON'T want to change all e164 numbers in my h.323 network and m

Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip

On January 30, 2005 12:18 pm, Brian Dingman wrote: > There is the little problem of having to switch numbers and then > communicating to everyone that the number has changed. This also only > seems to be a problem on inbound calls. And why, praytell, did you go into production with DIDs from a pro

RE: [Asterisk-Users] Asterisk on MS Virtual Server

<< Were the pops/drops/buzzes a problem only with communications via a telephony card?>> We ran Asterisk in a VPC instance (Redhat 8.0) for 3 months while we evaluated Asterisk. The only reason we had to move to a version of Linux running directly on hardware was a need to run X101P cards. We ha

[Asterisk-Users] Asterisk and Grandstreams on marginal broadband...

We've got an office with a marginal broadband connection. Do you think Asterisk + Grandstreams can be tuned to give a good quality call on with this jitter/latency ? This is the only broadband supplier in the area, so changing carriers would be difficult. I made a call from the Grandstream with

Re: [Asterisk-Users] iaxComm version 1.0 released

>Any work to support some USB Phones!? The ability to dial using the phones >keypad!? Not yet, but I'll probably add suport for the TigerJet phone eventually. That would be -awesome.- The mgmt at my company wants that ability, and integrating with IAX protocol would make portability huge.

Re: [Asterisk-Users] Asterisk on MS Virtual Server

Hi Greg, On Sun, 2005-30-01 at 09:42 -0500, Greg Boehnlein wrote: > It "works", but you will have timing issues and very poor audio quality. > > I've run linux both under Vmware as well as running it under CoLinux > directly on windows w/ no emulation neccessary. All emulation / > virtualizatio

Re: [Asterisk-Users] Sipura SPA-841 auto-answer support [patch]

Geoff Speicher wrote: Sipura has implemented auto-answer in version 0.9.5 of the SPA-841 firmware. However, it is implemented via the Call-Info header, which Asterisk stable doesn't currently support. The attached patch implments a quick hack to support the Call-Info header from the Dial() applica

Re: [Asterisk-Users] Asterisk on MS Virtual Server

On Sun, 30 Jan 2005, Paul Tyreman wrote: > http://www.digium.com/index.php?menu=astwind > > I think this may be worth a look, I'm downloading it as I type this > e-mail... > > I didn't know Asterisk had the possibility of being run on a windows machine > and while it's not as stable as a Linux

RE: [Asterisk-Users] where to buy x100p

wow! excellent information, this should be added to the wiki under x100p / tdm400. Especially the info that the X100P is 600 ohm only would have made it a lot clearer to me that I needn't buy an X100P (clone) for use in Europe On Sun, 30 Jan 2005, Rich Adamson wrote: Why does the X100P have ech

[Asterisk-Users] Re: Sipura SPA-841 auto-answer support [patch]

> The best place for patches to Asterisk is the Asterisk bug tracker, > which can be found at http://bugs.digium.com . > > *Please* don't forget to read the guidelines: > http://www.digium.com/bugguidelines.html and file a disclaimer. > > Thanks for the patch! I'll look for it on the bug tracker

[Asterisk-Users] Thank you

Thanks to everyone who responded with both advice and words of caution. I have no doubt I am going to be getting in over my head. That is just my M.O. but I'm still surviving an arguably better off for it. The bottom line is this: my organization needs a phone system. We can't afford state-of

[Asterisk-Users] D/41D

Hello, I'm attempting to setup a test asterisk server. I have a couple of old 4 port D/41D ISA dialogic cards. Has anyone had any success at getting them to work? I've done some searching on the internet and it seems like some have them working but I have not been able to find any help docs on h

Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip

There is the little problem of having to switch numbers and then communicating to everyone that the number has changed. This also only seems to be a problem on inbound calls. On Sat, 29 Jan 2005 23:34:49 -0500, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote: > On January 29, 2005 11:29 pm, Brian Ding

Re: [Asterisk-Users] Silly question: Why multiple lines on SIP phones?

On Sat, 29 Jan 2005 21:17:56 -0500 (EST), Paul Dugas wrote: >This is probably going to sound really silly and I must be confused about >it. Maybe someone can set me straight. > >I've been tinkering for a while with * and a number of different FXO/FXS >cards, SIP phones, and ATAs trying to get a f

[Asterisk-Users] Setting call forward for Agent's in a Queue

Hi!, I'm trying to set up a Queue (which works fine now :-) Sip clients can login in to the Queue with dialing 91 on there phone. And as soon as there are customers the Queue calls the agents back. I would like that the queue calls the agents also if it's phone is call-forwarded. With agents (sip

Re: [Asterisk-Users] Record inbound and outbound calls to and from one number.

Don't forget to warn your callers about the recording. Tim Mattison wrote: Try the monitor application instead of record. I think that'll do what you're looking for. On Fri, 2005-01-28 at 13:30 -0800, David Shaw wrote: Hello All, I would like to record inbound and outbound calls to and from one nu

Re: [Asterisk-Users] asterisk tries to dial out on lines already in use.

> When I place a call with asterisk, asterisk will try to dial > out on the first line even if the first line is already being > used by someone else. Any ideas on what I'm doing > wrong? My question would be, how would asterisk know the line is in use if it isn't controlling it?

[Asterisk-Users] Vservices.inv of Julian Pawlowski anoyne has the macro-dailer for this?

Hi, I've found the Vertical Service Codes / vservices.inc of Julian in the cache of google. It's an very extended extensions include with all the *21 *67 etc services implemented so it is stored to ODBC or if you replace it to Dbget/put etc. I'm wondering if somebody has the macro/agi for using t

RE: [Asterisk-Users] agent logoff

I have a separate extension set up for logoff that doesn't pass the callerid to AgentCallbacklogin. So my agents dial 301 to log on and 302 to logoff. A lot of the phones (Cisco for example) allow you set up speed dials, so its even easier for an agent to log on and off. -Original Message--

Re: [Asterisk-Users] Anyone having problems with LiveVoIP?

LiveVoIP has many problems. #1 seems to be customer service. #2 they use Level 3. I am switching to Txlink On Sun, 2005-01-30 at 09:15, Ryan Laginski wrote: > Hi, > I made the mistake of ordering an 800 number as well. I have the same > problem you have, asterisk registers, but I get a fast busy w

RE: [Asterisk-Users] where to buy x100p

> Why does the X100P have echo and the Wildcard TDM400P have no echo? > > I thought the only advantage of using the TDM400P was that it used > less interrupts than the X100P? > > Are there any other advantages? One of the major differences between the two cards is that tdm fxo modules have supp

Re: [Asterisk-Users] Record inbound and outbound calls to and from one number.

Try the monitor application instead of record. I think that'll do what you're looking for. On Fri, 2005-01-28 at 13:30 -0800, David Shaw wrote: > Hello All, > > I would like to record inbound and outbound calls to and from one > number. > > I tried to add lines to my extensions.conf: > > DAY=`

Re: [Asterisk-Users] where to buy x100p

The digium X100P is probably fine. What most of these people are talking about are the X100P clones which are of varying qualities. I have had no problems, but results vary. Cheers, Jon. On Sunday 30 January 2005 09:42 am, dean collins wrote: > Why does the X100P have echo and the Wildcard T

Re: [Asterisk-Users] Asterisk on MS Virtual Server

http://www.digium.com/index.php?menu=astwind I think this may be worth a look, I'm downloading it as I type this e-mail... I didn't know Asterisk had the possibility of being run on a windows machine and while it's not as stable as a Linux implementation, it might just do for the moment, as I d

Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip

On January 30, 2005 09:31 am, Greg Boehnlein wrote: > Same reason people stick with Gentoo after a stage one installation. ;) I > have a theory about Gentoo that explains the rabid nature of Gentoo fans. > I believe that people that radically defend Gentoo and it's stage one > installation process

[Asterisk-Users] newlines in application data strings (e.g. userevent)

exten => s,9,UserEvent(AgentMoreTime,Agent: ${agent}\r\nUntil: ${wrapupat}); Fragment "\r\n" parses into "rn". "\\r\\n" turns into "\r\n" (uninterpreted). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/li

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