RE: [Asterisk-Users] Configuration details for Asterisk interactionwith Vocal

Title: Message Please put this in the Wiki (www.voip-info.org). It is valuable information and will reach a wider audience there.   -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]Sent: January 3, 2005 1:52 AMTo: aste

[Asterisk-Users] Configuration details for Asterisk interaction with Vocal

I have seen a number of people in this newsgroup asking for information regarding asterisk interworking with Vocal. I was able to configure Vocal and Asterisk so that calls originating from vocal can land on an extension in Asterisk. I would like to share this info with the group   The scena

Re: [Asterisk-Users] extensions.conf sorting

Adi Linden wrote: This page on voip-info.org describes how it is possible to affect the sort order of patterns in extensions.conf. What is doesn't explain is how asterisk really does sort patterns. How does thi

[Asterisk-Users] extensions.conf sorting

This page on voip-info.org describes how it is possible to affect the sort order of patterns in extensions.conf. What is doesn't explain is how asterisk really does sort patterns. How does this happen? Adi ___

RE: [Asterisk-Users] Service contract for * in NYC area

Haha you're right.. I'm tired :P bkw > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Luke Catranis > Sent: Sunday, January 02, 2005 11:08 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [Asterisk-Use

RE: [Asterisk-Users] Service contract for * in NYC area

I'll make my entry into this Flamewar BKW I believe the term is "pot", as there are many colors of felines... As to the content... None of my business. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Monday, January 03, 2005 12:03 AM

RE: [Asterisk-Users] Service contract for * in NYC area

Karl, You talk about unprofessional? Isn't that the cat calling the kettle black? Your post was way beyond unprofessional it's not even funny Thanks, Brian > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Karl Brose > Sent:

[Asterisk-Users] Date problems with IP phones

Has anyone noticed a display date problems with IP phones running the PA168x chip set? At the year end the display date went from FRI 31 DEC 23:59 to SAT 02 JAN 00:00. I'm not sure whether its a problem with the chip set or the firmware/software cos it happens over at least 2 software versions.

Re: [Asterisk-Users] Service contract for * in NYC area

This was posted in a public forum asking for business relationship. But the intent was for disguised market research which could have been stated clearly upfront and there would have been overwhelming positive response attesting to the facts and assisting in his quest. But communicating with "col

RE: [Asterisk-Users] Qs about FXO/FXS cards

Truman Beal wrote: > One thing to look at is the proximedy to the powersupply of > your audio > devices. Some mobos have their chipset integrated in very closely to > their power supply pins. With an unclean power source the fluxuations > would be enough to add some of the white noise which would

RE: [Asterisk-Users] Firefly lockup in Win98

> Randy MacKay wrote: > > When I start Firefly, I see the little firefly box appear (flash on the > > screen), then the screen turns blue with an Error #44. I am > unable to do > > anything, but reboot the computer. > > Is the computer clean from viruses and spyware/adware? Yes, I regularly run

Re: [Asterisk-Users] Call Queue Question

On Sun, 2 Jan 2005 21:48:47 -0500, Glenn Dalgliesh <[EMAIL PROTECTED]> wrote: > When I first started looking at a similar problem started out on the same > path with app_queue but even having access to a friend of mine who actually > help write some of the Queue code we decided it wasn't the right

Re: [Asterisk-Users] Call Queue Question

When I first started looking at a similar problem started out on the same path with app_queue but even having access to a friend of mine who actually help write some of the Queue code we decided it wasn't the right tool for the job. We approached this a little differently using valetparking and agi

[Asterisk-Users] Call Queue Question

Hi, I've been fussing with this for a while now - and cannot seem to get it to work correctly - or rather as I desire it.. I'm trying to implement a 'find-me' feature in my voicemail. Basically - pressing 1 at the voicemail puts the caller into a queue. The members of the queue are my cell phone

Re: [Asterisk-Users] IAX media

On Sun, 02 Jan 2005 18:15:53 -0600, Eric Wieling aka ManxPower wrote: >mohammad wrote: > >> In IAX, because both signaling and rtp ports are uniqe, so Asterisk is >> always in rtp path. Am I right???/ > >IAX and IAX2 use the same port for signaling and audio. IAX and IAX2 do >NOT use RTP. I a

[Asterisk-Users] Terminal Adaptor down after register

Hi,   I have a terminal adaptor connected to Asterisk. I found that when it connect to  Asterisk v1.0.0, The TA send register msg, and get OK msg, but it get Notify msg every 1 minute, The TA reply with "call does not exist" , usually 10 minutes later, The TA is down. But it work well with

Re: [Asterisk-Users] Service contract for * in NYC area

Mr Brose. I don't see why you did this? why was this needed? All I wanted was to show my client that * is popular. Without this none of us can sell this if we want. Nobody buys a pbx they can't get support on (we are talking about a $50,000 installation). Therefore I had to show my client that it i

RE: [Asterisk-Users] Service contract for * in NYC area

Karl, Naughty, you know better than to do this in a public forum. If you're directing comments like this at a person you're leaving your ass flapping in the wind since it could be considered slander, defamation or my favorite libel. If you don't have hard facts to back something like this

RE: [Asterisk-Users] Codec Selection in Asterisk

Well disallow=all allow=ulaw,alaw,g729 That works in cvs-stable and head now. For sip order can/does matter on global and per user/peer. More people need to test and comment on this: http://bugs.digium.com/bug_view_page.php?bug_id=0002971 Don't say we didn't ask you to comment when it goes in

[Asterisk-Users] Using Asterisk as a TA?

Not sure that's an accurate subject line, but... I've started looking at Asterisk as a possibility for a small PBX here. I'm thinking of an ISDN (BRI) card for connection to the telco, with some analogue converters (Sipura) for existing phones too. I'm pretty sure that the functionality I want i

Re: [Asterisk-Users] Qs about FXO/FXS cards

On January 1, 2005 06:24 pm, Steven Critchfield wrote: > > 1. Power alarms. WTF does that mean? Wish I had some support docs. > > > > 2. On bootup, "Excessive leakage module x, ProSLIC failed Auto > > Configuration". Again, WTF? Reboot and it's ok. But, just a reboot > > after driving 100+ m

Re: [Asterisk-Users] Qs about FXO/FXS cards

On January 1, 2005 04:09 pm, Rich Adamson wrote: > b. don't ever post anything to the -dev list regarding a TDM card as >that is NOT the forum for digium cards or drivers, Eh? If you're hacking on the code for wctdm, -dev is most certainly an appropriate place to post. If you're just going

Re: [Asterisk-Users] Firefly lockup in Win98

Randy MacKay wrote: When I start Firefly, I see the little firefly box appear (flash on the screen), then the screen turns blue with an Error #44. I am unable to do anything, but reboot the computer. Is the computer clean from viruses and spyware/adware? Is it running any "firewall" software which

Re: [Asterisk-Users] Re: phones with two ethernet ports

I like the quality of the uniden UIP200. They just relesed new firmware and I don't yet have it workin behind a nat, but I am working on that. Uniden didn't post new docs for the new firmware yet. Lyle - Original Message - From: "Erick Perez" <[EMAIL PROTECTED]> To: Sent: Sunday, Janu

Re: [Asterisk-Users] Service contract for * in NYC area

Hi List, Don't deal with this guy (C F ), he is misleading people and just using the replies to show that what ever he is selling to his clients is popular. After a couple of exchanges: >C F wrote: >Thanks for your replay, at this time I'm just waiting for a commitment >from the client to take a

Re: [Asterisk-Users] IAX media

mohammad wrote: In IAX, because both signaling and rtp ports are uniqe, so Asterisk is always in rtp path. Am I right???/ IAX and IAX2 use the same port for signaling and audio. IAX and IAX2 do NOT use RTP. ___ Asterisk-Users mailing list Asterisk-User

Re: [Asterisk-Users] Meetme

Serge Schumacher wrote: exten => 550,1,Answer exten => 550,2,Wait(1) exten => 550,4,MeetMe(18|Md) exten => 550,5,Hangup and when I call 550 I get this error and the MusicOnHold (exten => 550,4,MeetMe(18|Md)) also doesn't work: -- Executing Answer("SIP/ses-0730", "") in new stack -- Execut

RE: [Asterisk-Users] Firefly lockup in Win98

When I start Firefly, I see the little firefly box appear (flash on the screen), then the screen turns blue with an Error #44. I am unable to do anything, but reboot the computer. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Matt > Sent: Sunday, Jan

Re: [Asterisk-Users] Indications UK - cant get away from american sounding dial tone

Howard Lowndes wrote: It's Sunday and I need a break so I decided to go thru the list, so I had to laugh to myself when reading this and decided to write this I hope someone elae finds this amusing as well. Why would we want to move away from the UK dial tones? They work well there and in mos

RE: [Asterisk-Users] ArtDio IPF-2000 or Sipura SPA-841

Adi - Yes I believe the IPF-1000 is formally discontinued, although I believe we still have some stock of them, and we will certainly warranty and support them. You can verify stock status by emailing one of our sales associates, [EMAIL PROTECTED] SPA-841 still has not shipped from Sipura, we hav

[Asterisk-Users] how can my agent fallback into an extension after they complete a call.

Hi, I'm trying to get my Agent onto an extension when they finish a call. Actually i'm doing that way: Call enter the the context 'cc-incoming' get into a Queue and then, when someone hangup (caller or callee) I launch an AGI which initiate a call to my Agent and then ask for a report code. That

Re: [Asterisk-Users] Meetme

Can someone see what's wrong here please ? exten => 550,1,Answer exten => 550,2,Wait(1) exten => 550,4,MeetMe(18|Md) exten => 550,5,Hangup The priority 3 is missing ... 1, 2 then timeout... -- Joel Vandal ScopServ Inc. ___ Asterisk-Users mailing list As

Re: [Asterisk-Users] Indications UK - cant get away from american sounding dial tone

On Mon, 2005-01-03 at 10:38, C F wrote: > Maybe some light finaly. > The only thing the English and Americans don't have in common is the > Langauge. I wish they would change their PSTN signaling to the > american way so this statement becomes true. Would you be the one to > start? "Two nations di

Re: [Asterisk-Users] phones with two ethernet ports

On Sun, 2 Jan 2005 16:35:12 -0500, Erick Perez <[EMAIL PROTECTED]> wrote: > Hi there, what phones are available that have two ethernet ports? Check out the Polycoms. IP500 and IP600. -Chuji ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

Re: [Asterisk-Users] Qs about FXO/FXS cards

Jim Van Meggelen wrote: Frankly, what is most interesting is the fact that your systems are trouble-free. Certainly if you were to ask if such systems could be put into production, you would probably be advised not to expect much. One last thing. I have a somewhat special PSTN connection. I subscri

Re: [Asterisk-Users] pridialplan=unknown ?

On Sun, 2 Jan 2005, Glenn Dalgliesh wrote: > After setting the pridialplan=unknown I seeing the Called Number TON change > to Unknown Number Type but not the Calling Number TON. Should both be > following this parameter or not. If not is their another option to change > the Calling Number TON? pr

Re: [Asterisk-Users] Meetme

Try making sure that the numbers match the one in meetme.conf. I don't know if this makes a difference but since in my config this is how it is setup and it works, just give it a try. On Mon, 3 Jan 2005 00:34:25 +0100, Serge Schumacher <[EMAIL PROTECTED]> wrote: > Hi, > > Can someone see what's

Re: [Asterisk-Users] Firefly lockup in Win98

Randy MacKay wrote: I tried Firefly in WinXP and it works fine. I tried it on Win98, it looks it up. Anyone experience this? Any ideas? Was running fine here until the Windows 98SE box blew itself up...all Linux now...bummer I've lost CounterStrike though! But seriously, where does it crash?

Re: [Asterisk-Users] BroadVoice WiSIP with Asterisk

On Mon, 2005-01-03 at 10:16, Adi Linden wrote: > Any idea where it can be purchased. > > Adi > > On Sun, 2 Jan 2005, Mike Dent wrote: > > > I've heard good reports of the Senao Wi-Fi SIP phone. You could try their web site if you can get past the stupid Chinese M$ IIS 404 page. > > Mike > ___

Re: [Asterisk-Users] Qs about FXO/FXS cards

Jim Van Meggelen wrote: If I may, I'd like to ask you some general questions about the environment these systems are running in. - How are these systems powered and grounded? Not optimally by a longshot. On the Athlon machine, my main machine, all the equipment is plugged into 2to3 prong adapters.

Re: [Asterisk-Users] Indications UK - cant get away from american sounding dial tone

Maybe some light finaly. The only thing the English and Americans don't have in common is the Langauge. I wish they would change their PSTN signaling to the american way so this statement becomes true. Would you be the one to start? It's Sunday and I need a break so I decided to go thru the list,

[Asterisk-Users] Meetme

Hi, Can someone see what's wrong here please ? I've installed the ztdummy driver to enable meetme, put his in my extension.conf exten => 550,1,Answer exten => 550,2,Wait(1) exten => 550,4,MeetMe(18|Md) exten => 550,5,Hangup this in my meetme.conf [rooms] ; ; Usage is conf => confno[,pin] ; con

Re: [Asterisk-Users] Indications UK - cant get away from american sounding dial tone

On Mon, 2005-01-03 at 09:59, Andrew Brown wrote: > Have a problem which can't find solution to on WIKI.. > > Trying to get * to use UK based indication tones. i.e. british ring, dial > tone, busy signal. > > Have changed the indications.conf file to default to UK. However this seems > to have no

Re: [Asterisk-Users] IAXy reliability issues >> try an Ag-168 ?

The manufacturers are chinese, IP Trading is in Sydney Australia, they sell the units as a distributor (I believe) and thats where i have been getting mine. I've had unsolicited emails from different chinese manufacturers but they have yet follow up on my replies/questons. Gary On Sun, 2 Jan 2

Re: [Asterisk-Users] BroadVoice WiSIP with Asterisk

Any idea where it can be purchased. Adi On Sun, 2 Jan 2005, Mike Dent wrote: > I've heard good reports of the Senao Wi-Fi SIP phone. > Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asteris

[Asterisk-Users] Indications UK - cant get away from american sounding dial tone

Have a problem which can't find solution to on WIKI.. Trying to get * to use UK based indication tones. i.e. british ring, dial tone, busy signal. Have changed the indications.conf file to default to UK. However this seems to have no affect. What am i missing. Am using 1.0.3 stable. Many thank

Re: [Asterisk-Users] BroadVoice WiSIP with Asterisk

I've heard good reports of the Senao Wi-Fi SIP phone. Mike On Sun, 2 Jan 2005 16:12:29 -0600, Adi Linden <[EMAIL PROTECTED]> wrote: > After some reading I see that DECT requires dedicated base station > hardware. I am looking for something to work with established 802.11 > infrastructure. I've u

[Asterisk-Users] pridialplan=unknown ?

After setting the pridialplan=unknown I seeing the Called Number TON change to Unknown Number Type but not the Calling Number TON. Should both be following this parameter or not. If not is their another option to change the Calling Number TON? > Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT

Re: [Asterisk-Users] FC2 & ztcfg - cannot find channel 2

More on this problem. Sometimes I get the following error message instead of the original one that I quoted: # ztcfg -vvv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) 2 channels conf

[Asterisk-Users] IAX media

Hi;       In IAX, because both signaling and rtp ports are uniqe, so Asterisk is always  in rtp path. Am I right???/   If yes, is there anyway to bypass Asterisk from rtp traffic?     Regards Mohammad ___ Asterisk-Users mailing list Asterisk-Users@li

Re: [Asterisk-Users] Qs about FXO/FXS cards

Thanks, I've applied the fix to two systems here. Guess I'll have to wait for recurrence as well. But, hopefully I'll have to wait forever. :) Rich > > > > Have you noticed that a TDM with fxo modules is more/less stable then > > a TDM with only fxs modules? > > > > Gu

[Asterisk-Users] Can I receive faxes with Fritz card & Asterisk ?

Hi, I'm reading that spandsp works only with zaptel channels. What are my options if I want to receive faxes through ISDN Fritz card with Asterisk and possibly forward it as emails ? Regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists

RE: [Asterisk-Users] BroadVoice WiSIP with Asterisk

After some reading I see that DECT requires dedicated base station hardware. I am looking for something to work with established 802.11 infrastructure. I've used a Cisco 7920 on Cisco CallManager. It works quite well. Unfortunately its price is way out of line for my purposes. Adi On Sun, 2 Jan 2

Re: [Asterisk-Users] phones with two ethernet ports

On Mon, 2005-01-03 at 08:35, Erick Perez wrote: > Hi there, what phones are available that have two ethernet ports? > I want to do some cabling at a new installation and i heard there are > such phones (SIP i guess) out there. That way i dont have to run two > cat5 to the user desktop. > I think 3C

Re: [Asterisk-Users] Re: phones with two ethernet ports

Erick Perez wrote: any others not so expensive? does these 3com sip phones work with * ? http://www.grandstream.com The BT102 has 2 ethernet ports. Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman

Re: [Asterisk-Users] Subject: Re: Dial with no phone line connected

> >> I have more FXO ports on TDM400's than I have PSTN lines available for > >> testing. When all the lines were used up (the FXO ports are all in > >> zap group 2), and I did a Dial(Zap/G2/${phone},5) it thinks the Dial > >> succeeded even though there is neither line voltage nor dial tone.

Re: [Asterisk-Users] Qs about FXO/FXS cards

Rich Adamson wrote: Have you noticed that a TDM with fxo modules is more/less stable then a TDM with only fxs modules? Gut feeling (no reasonable analysis at all) from various postings tend to suggest the TDM with fxo's is less stable. Would you agree or not? Yes. Also, could you share the driver

RE: [Asterisk-Users] phones with two ethernet ports

Some of the Cisco phones do (7940,7960 etc, also the 7910+SW but this is skinny only). Steve -Original Message- From: Erick Perez [mailto:[EMAIL PROTECTED] Sent: 02 January 2005 21:35 To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] phones with two ethernet ports Hi there,

Re: [Asterisk-Users] Codec Selection in Asterisk

Yes, this makes a lot of sense and gives me the information I am looking for! Thanks, Adi > The order does matter. > > You want to offer ulaw, alaw and g729. I want to offer g726, g729 and > gsm. > > In the order you have them listed in your stanza you offer them to me. > In the order I have them

[Asterisk-Users] Re: phones with two ethernet ports

Update: found the 3Com® 3101 Basic Speaker Phone Provides dual port 10/100 switched Ethernet for one-wire connectivity between the phone and a PC any others not so expensive? does these 3com sip phones work with * ? On Sun, 2 Jan 2005 16:35:12 -0500, Erick Perez <[EMAIL PROTECTED]> wrote: > Hi

[Asterisk-Users] phones with two ethernet ports

Hi there, what phones are available that have two ethernet ports? I want to do some cabling at a new installation and i heard there are such phones (SIP i guess) out there. That way i dont have to run two cat5 to the user desktop. I think 3COM had one but can't find the web site reference for the t

RE: [Asterisk-Users] BroadVoice WiSIP with Asterisk

Adi, spend your money on something else. I've tried it (Zyxel) and it is simply bad. If you want to go cordless - go VOIP-DECT. That seems to be the best working solution at the time. Check wiki for reference. Vel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Be

Re: [Asterisk-Users] Codec Selection in Asterisk

The order does matter. You want to offer ulaw, alaw and g729. I want to offer g726, g729 and gsm. In the order you have them listed in your stanza you offer them to me. In the order I have them in mine I offer them to you. In this case only g729 matches our lists and so we choose g729 for the cal

Re: [Asterisk-Users] Qs about FXO/FXS cards

> Victor Rini wrote: > > This has been an interesting discussion. I'll chime in with my > > experience here. > > > > I have two servers. One with the cheapest motherboard and athlon > > processor I could find on Newegg.com. The other is a 1999 era > > motherboard with a Via C3 processor, again

[Asterisk-Users] Subject: Re: Dial with no phone line connected

>> I have more FXO ports on TDM400's than I have PSTN lines available for >> testing. When all the lines were used up (the FXO ports are all in >> zap group 2), and I did a Dial(Zap/G2/${phone},5) it thinks the Dial >> succeeded even though there is neither line voltage nor dial tone. >> Can

Re: [Asterisk-Users] Qs about FXO/FXS cards

Victor Rini wrote: This has been an interesting discussion. I'll chime in with my experience here. I have two servers. One with the cheapest motherboard and athlon processor I could find on Newegg.com. The other is a 1999 era motherboard with a Via C3 processor, again a bargain basement specia

[Asterisk-Users] Echo problems

Hi to all, I recently installed a Fritz PCI ISDN BRI card in hope that the echo problems would go away, but unfortunately it didn't solve the problem when calling analog lines, when calling cell phones and PBX with a E1/T1 it works great. Can anyone suggest what would be a way to

[Asterisk-Users] Codec Selection in Asterisk

I am wondering how Asterisk selects codecs between devices. For example, in my sip.conf I have: disallow=all allow=ulaw allow=alaw allow=g729 Does the order matter? Does it mean it will try each codec in succession and use the first that both endpoints support? Thanks, Adi __

[Asterisk-Users] Box unstable after loading zaptel drivers for X100P

Hi list! I have been experiencing a lot of mystical hangs on my box. At first I suspected bad memory or motherboard but I found out that the mystical hangs disappear when I do not load the drivers for my x100p. The box will only hang under high load (copy operations with 30Mb/sec to a raid arra

[Asterisk-Users] Clipping on outbound calls via SIP/IAX

I'm hoping someone can help me with a problem I've been having for a while now. I've googled and wiki'd to no avail. Whenever I place an outbound call from * to a PSTN through a SIP or IAX provider (e.g. Voicepulse or Broadvoice), the first 1/2 to 2 seconds of the remote call are clipped (muted).

Re: [Asterisk-Users] Qs about FXO/FXS cards

One thing to look at is the proximedy to the powersupply of your audio devices. Some mobos have their chipset integrated in very closely to their power supply pins. With an unclean power source the fluxuations would be enough to add some of the white noise which would give you the whine. Exc

Re: [Asterisk-Users] Booting * from CF

Michael Graves wrote: Yeah, I'm familiar with this. I even own a Soekris box for my firewall. However, I need something with more CPU power and the ability to take a TDM400 card. So I'm really trying to make the plain vanilla PC that I presently use into something that's more robust. I'd like to el

RE: [Asterisk-Users] Qs about FXO/FXS cards

Dorn Hetzel wrote: > On Sat, Jan 01, 2005 at 07:23:58PM -0500, Jim Van Meggelen wrote: >> [...] >> What if, for example, the TDM400 issues were a cumulative thing? If >> you had over 6dB of attenuation on the PSTN loop, coupled with >> greater than 5V potential on the neutral-ground of your elecric

RE: [Asterisk-Users] Qs about FXO/FXS cards

[EMAIL PROTECTED] wrote: > This has been an interesting discussion. I'll chime in with my > experience here. > > I have two servers. One with the cheapest motherboard and athlon > processor I could find on Newegg.com. The other is a 1999 era > motherboard with a Via C3 processor, again a bargain

[Asterisk-Users] ArtDio IPF-2000 or Sipura SPA-841

I am looking at some lower cost phone to use with Asterisk. What is the ArtDio IPF-2000 or the Sipura SPA-841 like? Also, I see voipsupply.com has an ArtDio IPF-1000 listed, is this a new or an old model? I cannot find any information on it. Adi ___ Ast

Re: [Asterisk-Users] Softphone in German

> Hello, > X-Lite in German: > http://www.globalipphones.com/xlite Thank you! That is exactly what I was looking for. Thanks, Adi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To

Re: [Asterisk-Users] Booting * from CF

You should be able to boot a full system from a 32M card without a problem. > >I've read J.R. Richardson's paper "Create an Embedded Asterisk Server" > >which outlines making a Debian server that boots from a compressed disc > >image on a CF card. I'm really interested in this as I want my * serv

Re: [Asterisk-Users] MeetMe

Serge Schumacher wrote: Can it be that the MeetMe application is not installed by default even if there is a meetme.conf ? pbx.c:1280 pbx_extension_helper: No application 'MeetMe' for extension (from-sip, 550, 4) If you don't have zaptel installed Astrisk won't build Meetme.

Re: [Asterisk-Users] chan-sccp problem, phone is not registering

Afraid not (yet). I contacted the maintainer of chan_sccp (Jan Czmok) and I sent him 2 tcpdump files one with the skinny and succesfull registration and the other with chan_sccp. This was just before x-mas though and I haven't received a reply yet, probably due to the holidays. The IP600 will

Re: [Asterisk-Users] Softphone in German

Hello, X-Lite in German: http://www.globalipphones.com/xlite Gunnar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.co

Re: [Asterisk-Users] Booting * from CF

On Sun, 02 Jan 2005 09:47:22 -0800, Steven P. Donegan wrote: >Michael Graves wrote: > >>Hi All, >> >>I've read J.R. Richardson's paper "Create an Embedded Asterisk Server" >>which outlines making a Debian server that boots from a compressed disc >>image on a CF card. I'm really interested in this

Re: [Asterisk-Users] Qs about FXO/FXS cards

This has been an interesting discussion. I'll chime in with my experience here. I have two servers. One with the cheapest motherboard and athlon processor I could find on Newegg.com. The other is a 1999 era motherboard with a Via C3 processor, again a bargain basement special. The Athlon system

RE: [Asterisk-Users] Cisco Wireless IP Phone 7920

You can use it with SCCP channel. We have tried and it works well. But I don't recommend this phone it's battery is not enough. It has a 3-4 hours time of standby. Best Regards. Yusuf Alakavuk Teknik Danisman - Technical Consultant Grid Bilisim Teknolojileri A.S. Kustepe Mahallesi Leylak Soka

Re: [Asterisk-Users] spandsp-0.0.2pre6

Hi, Steve Underwood schrieb: Recent versions no longer require libtiff internals. Cool! Recent versions of Debian should have the libtiff bugs fixed. Some people trying to work with spandsp got them in there :-) :-) Debian packages of spandsp and app_dtmftotext, app_rxfax and app_txfax are built

Re: [Asterisk-Users] Booting * from CF

Michael Graves wrote: Hi All, I've read J.R. Richardson's paper "Create an Embedded Asterisk Server" which outlines making a Debian server that boots from a compressed disc image on a CF card. I'm really interested in this as I want my * server to be more like an appliance than a PC. However, the p

[Asterisk-Users] Booting * from CF

Hi All, I've read J.R. Richardson's paper "Create an Embedded Asterisk Server" which outlines making a Debian server that boots from a compressed disc image on a CF card. I'm really interested in this as I want my * server to be more like an appliance than a PC. However, the paper is only an outli

Re: [Asterisk-Users] Dialling 9 for an outside line

C F wrote: You can do something like this: [directdial] ignorepat => 9 exten => 9,1,Dial(Zap/1/) this will give the channel 1 on zap, it justs gives you the Dial tone on it and then you can dial. or you can do this: [localdial] ignorepat => 9 exten => _9NXX,1,Dial(Zap/1/w${EXTEN:1}) this will

RE: [Asterisk-Users] Qs about FXO/FXS cards

> [...] > > I'm running older, but solid hardware and not seeing any > > issues. I'm using a Compaq Proliant 1850R Gen1 dual PII 400 > > with 512MB ram, GB ethernet, and SATA Hardware RAID. Cheap, > > efficient, redundant. And for a Debian box, good enough. > [...] > > I just have to add m

Re: [Asterisk-Users] Sound Card Buzzing

One Sigurd Karlsbakk, there's only one Sigurd Karlsbakk, one Sigurd Karlsbakk, there's only one Sigurd Karlsbakk. Ta la. - Original Message - From: "Roy Sigurd Karlsbakk" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Sunday, January 02, 2005 12:

RE: [Asterisk-Users] ASTCC gsm files

The AGI looks in /var/lib/asterisk/sounds for its sound files with names not fully qualified.   This looks like a bug in the ASTCC makefile or the astcc.agi should refer to the soundfiles in the /usr/share/asterisk/sounds directory explicitly.     Karl Putz -Original Message-From

Re: [Asterisk-Users] IAXy reliability issues >> try an Ag-168 ?

Gary, IP trading is the vendor of this... Does it have any distributor in Europe? Or website to check for informations? Thanks Helder - Original Message - From: "Gary" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, December 31, 2004 3:28

Re: [Asterisk-Users] how is a upgrade performed?

On Friday 31 December 2004 09:41 pm, Charles S. Antrim wrote: > I have a stable server and want to upgrade. How do I upgrade to the > latest version of * ? > Try (wget or http) szmidt.org/asterisk/asterisk-update.sh Run it without any parameters to see your available options. It will perform a b

Re: [Asterisk-Users] No incoming Audio in H323

I have almost the same problem (but in reverse). When I make a call the other person can hear me but I can't hear the other person. I believe it has something to do with the router i'm using to connect the asterisk to the net, because if I connect my pc to that same router to an ethernet port of i

Re: [Asterisk-Users] Dialling 9 for an outside line

You can do something like this: [directdial] ignorepat => 9 exten => 9,1,Dial(Zap/1/) this will give the channel 1 on zap, it justs gives you the Dial tone on it and then you can dial. or you can do this: [localdial] ignorepat => 9 exten => _9NXX,1,Dial(Zap/1/w${EXTEN:1}) this will dial the sev

Re: [Asterisk-Users] Dialling 9 for an outside line

richard wrote: I have the following line in my extensions.conf which when I dial 100 (from my BudgeTone 100, and then wait a few seconds) I will get an outside line. exten =>100,1,Dial(Zap/1,20) What do I need to put in the extensions.conf file so that I can dial 9 (and then a number) and then

RE: [Asterisk-Users] Qs about FXO/FXS cards

> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > [EMAIL PROTECTED] > Sent: Saturday, January 01, 2005 9:12 PM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] Qs about FXO/FXS cards > [...] > I'm running older, but solid hardw

Re: [Asterisk-Users] Qs about FXO/FXS cards

On Sat, Jan 01, 2005 at 07:23:58PM -0500, Jim Van Meggelen wrote: >[...] > What if, for example, the TDM400 issues were a cumulative thing? If you > had over 6dB of attenuation on the PSTN loop, coupled with greater than > 5V potential on the neutral-ground of your elecrical receptacle, > compound

[Asterisk-Users] Dialling 9 for an outside line

I have the following line in my extensions.conf which when I dial 100 (from my BudgeTone 100, and then wait a few seconds) I will get an outside line. exten =>100,1,Dial(Zap/1,20) What do I need to put in the extensions.conf file so that I can dial 9 (and then a number) and then Asterisk automa

[Asterisk-Users] Cisco Wireless IP Phone 7920

Dear Group, Is the "Cisco Wireless IP Phone 7920" compatible with * Is it gooed? Thanks, Willy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update o

[Asterisk-Users] ASTCC gsm files

 Hi ALL;     I installed ASTCC, but the sound files(*.gsm) cannot be opened.All of them are located in : /usr/share/asterisk/sound.   Asterisk says:   Jan  2 06:01:54 WARNING[9759]: file.c:475 ast_openstream: File astcc-tone does not exist in any formatJan  2 06:01:54 WARNING[9759]: res_agi.

  1   2   >