Re: [asterisk-users] UPDATE... 2 down, 1 to go (3 questions - variables, upgrading, and IRC)

On Fri, 8 Jun 2007, Steve Edwards wrote: > On Fri, 8 Jun 2007, Jared Smith wrote: > > > On 6/7/07, Nick Seraphin <[EMAIL PROTECTED]> wrote: > >> Still need an answer to this one. > > > > I wrote a response yesterday, but it looks like it didn't come through > > for some reason. The answer is t

Re: [asterisk-users] UPDATE... 2 down, 1 to go (3 questions - variables, upgrading, and IRC)

On Fri, 8 Jun 2007, Jared Smith wrote: > On 6/7/07, Nick Seraphin <[EMAIL PROTECTED]> wrote: > > Still need an answer to this one. > > I wrote a response yesterday, but it looks like it didn't come through > for some reason. The answer is to use the DumpChan() application and > watch the CLI w

[asterisk-users] Softphone for smartphone such as Nokia N90 / 93 / N95

looking for good sip softphone for wifi and 3G network. 1 are there any sip softphone ( with gsm/g723/G729 codec ) for smartphone such as Nokia N90 / 93 / N95 ? 2 are there any sip softphone ( with gsm/g723/G729 codec ) for Window mobile5 Or wm2003 ? 3 How is the sound qua

[asterisk-users] CDR accuracy

Hi all users, I has been joining this user list for about 1 year, and always has seen the successful story about the Asterisk act as IP PBX and even communication appliances solutions. And thank for this list to help each other and make everyone success. I also being inspired by this user-list and

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

Ya, I have done that, below is zapata.conf. Also we had an TMP card with analog lines. & SIP cals were great on them. & now when we switched over. SIP calls have echo.. which shouldnt be at all. [channels] language=en #include zapata_additional.conf context=from-pstn switchtype=national pridial

[asterisk-users] SIP Transit problem

Hi! Hope someone can help me. I'm trying to pass SIP traffic from one asterisk to another through a third server. Here is the desired scenario: ServerA -- SIP --> ServerB -- SIP --> ServerC When a call is placed on a ServerA local, I can see that ServerB receives the call and dials ServerC. But

Re: [asterisk-users] Changing the messages that are played when a user is unavailable/busy

Thnx for your quick replies. I will try all of the above methods :-) On Fri, 2007-06-08 at 13:02 -0400, Justin Moore wrote: > On 6/8/07, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote: > > Log into your mailbox. Press "0", then press the option listed to > > record your unavail and busy greetin

[asterisk-users] Asterisk 1.4 with Unicall

I have a small call center running with Asterisk 1.4.4 and Unicall. Everything seems to be working but twice now we had to reset the server because all lines stopped working. You can see users dialing in and reaching the queue but the agents never get the call and the lines are not release

[asterisk-users] FW: Delivery Status Notification(Failure)

[EMAIL PROTECTED], you are email bombing me, please fix your blackberry! Thanks, Steve Totaro http://www.asteriskhelpdesk.com > -Original Message- > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] > Sent: Friday, June 08, 2007 7:38 PM > To: Steve Totaro > Subject: Delivery Status Noti

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

On Sat, 9 Jun 2007, Deepak Naidu wrote: But now when we are live, there is a terrible echo between 2 SIP calls. If I call the same extension from outside the voice is clear. My impression is that the transcoding that takes place between two purely software SIP calls never goes through the TE

Re: [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING

Moy: I have working an Asterisk 1.4.4 with Unicall rn MFR2. The only problem i have is the RxFAX application, that broke every time... With and error in the linking to the spandsp library. If i have time this weekend i will review to fix the app, Thanks. On 6/4/07, Tobias Wolf <[EMAIL P

[asterisk-users] Bad Echo between SIP calls

Hi, We have a PRI connection & when its was on test networks we had echo problems withoutside line. So I bought a TE212P card resolve the echo problem. Which did to an extent. Its using asterisk 1.2.18 & RHEL4-Update 4. But now when we are live, there is a terrible echo between 2

Re: [asterisk-users] choppy sound with playback, background, etc... but not with musiconhold

> On Fri, 8 Jun 2007, Paco Brufal wrote: >> I have an asterisk 1.2.18 working fine, the only problem is that all >> applications that play audio, sound like "tremolo" or "vibrato", but >> musiconhold plays fine. > Do you have ztdummy loaded? Yes, I tried that, but the problem persists :( --

[asterisk-users] Zaptel 1.2.18 and 1.4.3 released!

The Asterisk and Zaptel development team is proud to announce the releases of Zaptel 1.2.18 and 1.4.3. This releases contains some updated hardware support as well as numerous bug fixes, including: * A fix for the potential for a rare deadlock between zaptel and the wct4xxp, wcte11xp, and w

Re: [asterisk-users] choppy sound with playback, background, etc... but not with musiconhold

On Fri, 8 Jun 2007, Paco Brufal wrote: Hello, I have an asterisk 1.2.18 working fine, the only problem is that all applications that play audio, sound like "tremolo" or "vibrato", but musiconhold plays fine. The same audio file (wav, mp3, ...) works fine with Musiconhold() but

Re: [asterisk-users] call problem...

On Fri, Jun 08, 2007 at 08:24:45PM +0100, Carlos Jerónimo wrote: > Hi, tahnks for your answer. but i haved install with command apt-get > install asterisk, but i don't have package asterisk-addons. > if i download asterisk-addons by digium site, run well with asterisk > debian pakages?? Not exactl

RE: [asterisk-users] Can Asterisk RAS?

The IAXMODEM might get you half way there...but if you want to connected it to a windows box (which I assume is why you use the RAS acronym), you'll have to look for remote serial port software. -MD- -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christ

Re: [asterisk-users] Can Asterisk RAS?

On 6/8/07, Christopher Dobbs <[EMAIL PROTECTED]> wrote: I am trying to set up somthing so I can dial into my asterisk box, and have it behave as a modem bank. Is there anything like that already, or am I going to have to write my own. I checked googls and found no leads, but thought I would ask

[asterisk-users] Replacing SX-2000 Centigram Voicemail with Asterisk?

We have a customer with an obsolete Centigram voicemail system who would like to replace it with Asterisk. Any one with experience doing this or information on the signalling and trunking used to connect the Mitel SX-2000 to the Centigram server? -- George Pajari (dCAP), netVOICE communicatio

Re: [asterisk-users] call problem...

Hi, tahnks for your answer. but i haved install with command apt-get install asterisk, but i don't have package asterisk-addons. if i download asterisk-addons by digium site, run well with asterisk debian pakages?? thanks 2007/6/8, Tzafrir Cohen <[EMAIL PROTECTED]>: On Fri, Jun 08, 2007 at 02:5

Re: [asterisk-users] Write to multiple databases as redundancy scheme

On 6/8/07, Justin Moore <[EMAIL PROTECTED]> wrote: On 6/8/07, Chris Mason (Lists) <[EMAIL PROTECTED]> wrote: > It would be better to let MySQL handle that - use the built-in > replication facilities. It's easy to setup. That's a great idea for backup purposes, but if the OP is wanting true redu

[asterisk-users] Re: Write to multiple databases as redundancy scheme

Can Asterisk write to multiple MySQL databases in different machines, at the same time, as a backup scheme? If it does, where can that be configured? In res_mysql.conf file? No, you cannot write to 2 different mysql servers with res_mysql. Just use MySQL replication as an alternative. Easy to s

[asterisk-users] Re: custom cdr fields and cdr_mysql, howto?

On 6/7/07, JR Richardson <[EMAIL PROTECTED]> wrote: Hi All, http://www.voip-info.org/wiki/index.php?page=Asterisk+func+cdr Under example: exten => s,2,Set(CDR(MyFavoriteBand)=Foo Fighters) exten => s,3,Set(CDR(MyFavoriteSong)=Hero) and under description: -userfield: The channel's user specifie

[asterisk-users] Log interpretation

Hi, Are there any decent (commercial or free) LOG parsers for A*k. Its *really* hard to debug issues involving multiple calls (eg meetme) when all of the messages are interlaced with each other. There must be an easier way. (A*K 1.2.18) Adrian ___ --B

RE: [asterisk-users] Write to multiple databases as redundancy scheme

UltraMonkey (www.ultramonkey.com) and MySQL Cluster (http://dev.mysql.com/doc/refman/5.1/en/mysql-cluster.html) It works a charm. - Brad > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Justin Moore > Sent: Friday, June 08, 2007 2:13 PM > To: [

[asterisk-users] Can Asterisk RAS?

I am trying to set up somthing so I can dial into my asterisk box, and have it behave as a modem bank. Is there anything like that already, or am I going to have to write my own. I checked googls and found no leads, but thought I would ask here before I tried writing my own, just to make sure

Re: [asterisk-users] call Hold event asterisk

On Fri, 2007-06-08 at 11:12 -0600, Anthony Francis wrote: > Lee Jenkins wrote: > > sathish s wrote: > >> i need to catch the call hold event from my asterisk-java program. > >> Im using net.sf.asterisk.*; for communicating with asterisk server. > >> I need to get the call hold status on my java

[asterisk-users] Need help on Text entry for asterisk through touchpad

Hi, I need to build Text entry application by using asterisk. I already tried this with spandsp application along with app_dtmftotext.c file, it was not working because of some version problem. Is there any way of building the text entry application through touch pad. Regards K.Rajesh.

[asterisk-users] Polycom phone registration problem

Hi, One of my users is in trouble with his polycom phone hooked to an asterisk server. The phone works fine for a few days, and then disappears from the registered sip peers in asterisk. The user is able to place outbound phone calls, but can't receive incoming calls until the network plug is un

Re: [asterisk-users] Changing the messages that are played when a user is unavailable/busy

Timothy Parez wrote: > I have my custom sounds which should be played instead of the default > ones when a user is busy or unavailable: > > "The person at extension XXX is not available right now, please" > > Of course I can simply replace the files, but the problem is my > implementation s

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

iowait time? I'm not familiar with that. Where are you seeing that? Also, is it a reproducible problem? --- Matthew Fredrickson Software Engineer Digium, Inc. On Jun 8, 2007, at 11:23 AM, Steve Hanselman wrote: It probably did but we run in updates every week and nobody can state exactly w

Re: [asterisk-users] getting at ${CALLERIDNUM}

On Thu, Jun 07, 2007 at 04:52:31PM -0400, Jared Smith wrote: > On 6/7/07, Matthew Pease <[EMAIL PROTECTED]> wrote: > > I'm having awesome fun with Asterisk & voicepulse connect together. > > So cool. > > I'm glad you're having fun! > > > I'm trying to have the caller id read back to me.Do

RE: [asterisk-users] agi with java?

> Java is not a great solution for AGIs because they are script you should > fire up and terminate very fast, while the overhead of launching a JVM, > loading all classes, etc, is pretty large. Also, you don't want multiple > JVMs in parallel loading everything multiple times. How about writing yo

Re: [asterisk-users] Write to multiple databases as redundancy scheme

Justin Moore wrote: On 6/8/07, Chris Mason (Lists) <[EMAIL PROTECTED]> wrote: It would be better to let MySQL handle that - use the built-in replication facilities. It's easy to setup. That's a great idea for backup purposes, but if the OP is wanting true redundancy, that won't help much. What

Re: [asterisk-users] Console duplicate output problem

Eric "ManxPower" Wieling wrote: This is really strange. Every message to the (VGA) console is written twice to the screen, but not on the SSH connection. Any clues how to stop this behavior? Stop running in graphics mode. OK, that's a great clue, but can you tell me how to disable now? B

Re: [asterisk-users] Write to multiple databases as redundancy scheme

On 6/8/07, Chris Mason (Lists) <[EMAIL PROTECTED]> wrote: It would be better to let MySQL handle that - use the built-in replication facilities. It's easy to setup. That's a great idea for backup purposes, but if the OP is wanting true redundancy, that won't help much. What happens then when th

Re: [asterisk-users] Write to multiple databases as redundancy scheme

[EMAIL PROTECTED] wrote: Hi, Can Asterisk write to multiple MySQL databases in different machines, at the same time, as a backup scheme? If it does, where can that be configured? In res_mysql.conf file? Not that I'm aware of, but you can setup MySQL to mirror the data to a slave database.

Re: [asterisk-users] Write to multiple databases as redundancy scheme

Why would you do this why put the overhead inside asterisk when mysql has perfectly good replication mechanisms built in? On Jun 8, 2007, at 12:44 PM, [EMAIL PROTECTED] wrote: Hi, Can Asterisk write to multiple MySQL databases in different machines, at the same time, as a backup schem

[asterisk-users] Write to multiple databases as redundancy scheme

MySQL has its own ways of doing this kind of thing. Take a look at the documentation on MySQL's website related to replication. Bobby _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent:

Re: [asterisk-users] Write to multiple databases as redundancy scheme

It would be better to let MySQL handle that - use the built-in replication facilities. It's easy to setup. -- Chris Mason (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scan

Re: [asterisk-users] call Hold event asterisk

Lee Jenkins wrote: sathish s wrote: i need to catch the call hold event from my asterisk-java program. Im using net.sf.asterisk.*; for communicating with asterisk server. I need to get the call hold status on my java program . I can able to get the music on hold status but i cannot able to

Re: [asterisk-users] Changing the messages that are played when a user is unavailable/busy

On 6/8/07, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote: Log into your mailbox. Press "0", then press the option listed to record your unavail and busy greetings. I'm no expert, so someone feel free to correct me if I'm wrong, but you should be able to make one or two recordings and then e

[asterisk-users] Write to multiple databases as redundancy scheme

Hi, Can Asterisk write to multiple MySQL databases in different machines, at the same time, as a backup scheme? If it does, where can that be configured? In res_mysql.conf file? Does anyone ever made it? Regards, Ricardo. ___ --Bandwidth and Colo

[asterisk-users] Re: Not getting CID Name from PRI

On 6/8/07, Kyle Sexton <[EMAIL PROTECTED]> wrote: Having a problem w/ not getting CID name from a PRI. CID Name appears in the PRI debug, but even after a Wait(4) it still appears after the phone is ringing. Here is the relevant info from my PRI debug output. Line 4 is a NoOp showing me tryin

Re: [asterisk-users] Best Codec

I'm wary of using g711 of public broadband networks. ... It'd be interesting to see some comparisons or comments from people using g726 as this does seem to be supported by quite a few hardware devices. We are using g711 pretty much exclusively for all residential customers in the US and it work

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

It probably did but we run in updates every week and nobody can state exactly when the problem started only "a few weeks ago" - not very helpful. I can see that when I hear the issue the iowait time is high on the processor. Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL

[asterisk-users] choppy sound with playback, background, etc... but not with musiconhold

Hello, I have an asterisk 1.2.18 working fine, the only problem is that all applications that play audio, sound like "tremolo" or "vibrato", but musiconhold plays fine. The same audio file (wav, mp3, ...) works fine with Musiconhold() but not with Playback() or Background()...

Re: [asterisk-users] Hot GXP-2000

Carlos, We had this happen once here with a batch of phones received from Grandstream about a year ago now. Email Grandstream on it and they should know exactly what the problem is, I believe they ended up replacing the phones for us. Jessee Holmes Atacomm / Ataractic Corporation www.ata

Re: [asterisk-users] agi with java?

Lenz wrote: Hello Matthew, Java is not a great solution for AGIs because they are script you should fire up and terminate very fast, while the overhead of launching a JVM, loading all classes, etc, is pretty large. Also, you don't want multiple JVMs in parallel loading everything multiple tim

Re: [asterisk-users] Best Codec

Yes. In fact it's around 32kbps, for a high duration call. MRTG statistics. Are you using G729A ou B? (VAD can reduce the usage). Att, Ricardo Martins. Henry Cobb escreveu: On 6/7/07, Ricardo Martins <[EMAIL PROTECTED]> wrote: We use G.729. Consumes only 35kbps of bandwidth and has a level 4

Re: [asterisk-users] Console duplicate output problem

Barton Fisher wrote: Anybody have an answer? TIA This is really strange. Every message to the (VGA) console is written twice to the screen, but not on the SSH connection. Any clues how to stop this behavior? -- Executing BackGround("Zap/216-1", "custom/3566/91_|m|") in new stack

[asterisk-users] No/unknown event '0' on timer

Hey guys, I'm currently running Asterisk 1.2.18 Under Mandriva Linux. Three Facilities are hooked together via IAX2 (Trunked) over a OpenVPN connection on a 10mbit (uplink/downlink) internet connection. I was parked for around thirty seconds at a remote facility. All of a sudden, the call

RE: [asterisk-users] Best Codec

> I know that g729 is the king-all, but I want to know what the rest of > the professional are using out there. g729 has a cost involved, so does > the cost really offset the performance? Or is it better to go with g711 > to start off? I'm wary of using g711 of public broadband networks. Althoug

Re: [asterisk-users] Changing the messages that are played when a user is unavailable/busy

Timothy Parez wrote: Hi, I have my custom sounds which should be played instead of the default ones when a user is busy or unavailable: "The person at extension XXX is not available right now, please" Of course I can simply replace the files, but the problem is my implementation shouldn't

[asterisk-users] Hot GXP-2000

This is off topic for Asterisk but I need a suggestion. I have a customer (travel agency) that has recently begun complaining that their GXP-2000 phones are getting very hot, they say that around mid day the handset is so hot that it can burn your ear. These phones are in constant use and

Re: [asterisk-users] UPDATE... 2 down, 1 to go (3 questions - variables, upgrading, and IRC)

On Fri, 8 Jun 2007, Jared Smith wrote: On 6/7/07, Nick Seraphin <[EMAIL PROTECTED]> wrote: Still need an answer to this one. I wrote a response yesterday, but it looks like it didn't come through for some reason. The answer is to use the DumpChan() application and watch the CLI when it's cal

RE: [asterisk-users] Q931 Error with H323

Dovid, Please provide a simple network diagram for members of this list. "Q931 cause 44 error" is a layer 3 ISDN error (Requested circuit/channel not available) most likely mapped backwards from PRI T1 interworking. John Treble Ottawa, Canada From:

[asterisk-users] Not getting CID Name from PRI

Having a problem w/ not getting CID name from a PRI. CID Name appears in the PRI debug, but even after a Wait(4) it still appears after the phone is ringing. Here is the relevant info from my PRI debug output. Line 4 is a NoOp showing me trying to echo Name and Number. Line 6 dials the extens

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

Did it accompany an update you made? If you can find out what version the problem started occurring, that would help in fixing the problem. Matthew Fredrickson Software/Hardware Engineer Digium, Inc. On Jun 8, 2007, at 2:59 AM, Steve Hanselman wrote: The setup. Asterisk is on a 3G Zeon Dell

Re: [asterisk-users] call problem...

On Fri, Jun 08, 2007 at 02:52:39PM +0100, Carlos Jerónimo wrote: > Hi, i got Ubuntu 6.06 installed and theres a problem with asterisk. > > I've sucessfully installed it with the command: > #apt-get install asterisk > > Then after installing FreePBX i get this error when restarting asterisk: > [EM

[asterisk-users] Changing the messages that are played when a user is unavailable/busy

Hi, I have my custom sounds which should be played instead of the default ones when a user is busy or unavailable: "The person at extension XXX is not available right now, please" Of course I can simply replace the files, but the problem is my implementation shouldn't (MUST NOT) mention the

[asterisk-users] call problem...

;SIP/4000-163c", "EXTTOCALL=2000") in new stack -- Executing Set("SIP/4000-163c", "CFUEXT=") in new stack -- Executing Set("SIP/4000-163c", "CFBEXT=") in new stack -- Executing Set("SIP/4000-163c", "RT=") in new stack -- Exec

Re: [asterisk-users] getting at ${CALLERIDNUM}

On 6/8/07, Matthew Pease <[EMAIL PROTECTED]> wrote: when will it be out? Soon... it's going through the copyediting process right now. I can't give any more specific timeframe than that, as I don't know how long it'll take to get through the entire process, but if I had to make a wild guess I'

Re: [asterisk-users] UPDATE... 2 down, 1 to go (3 questions - variables, upgrading, and IRC)

On 6/7/07, Nick Seraphin <[EMAIL PROTECTED]> wrote: Still need an answer to this one. I wrote a response yesterday, but it looks like it didn't come through for some reason. The answer is to use the DumpChan() application and watch the CLI when it's called. -Jared

Re: [asterisk-users] Noise on FXS ports (Sangoma)

John Novack wrote: > In this troubleshooting case, it probably is better that there is NO > dialtone, which would make the hiss easier to hear. > I am curious what the OP found > When Asterisk is stopped, does the hiss continue? That's tough to assess because the other problem I have had with it i

[asterisk-users] Asterisk, NAT and canreinvite=yes

Hi, I can not get this working: Asterisk on public IP. Two SIP phones behind NAT - in the same LAN. I works perfectly (two way sound) when each peer (friend) can not reinvite - audio stream goes through Asterisk. The problem pops up when I define canreinvite=yes on each peer definision so I supp

Re: [asterisk-users] agi with java?

Hello Matthew, Java is not a great solution for AGIs because they are script you should fire up and terminate very fast, while the overhead of launching a JVM, loading all classes, etc, is pretty large. Also, you don't want multiple JVMs in parallel loading everything multiple times. This

Re: [asterisk-users] agi with java?

- Original Message - From: "Matthew Pease" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, June 07, 2007 5:32 PM Subject: [asterisk-users] agi with java? Hi all - Searching for java agi in the mailing list archives turns up ancient

Re: [asterisk-users] Noise on FXS ports (Sangoma)

In this troubleshooting case, it probably is better that there is NO dialtone, which would make the hiss easier to hear. I am curious what the OP found When Asterisk is stopped, does the hiss continue? That would help to narrow down the location of the problem It sure sounds to me as if it is a h

Re: [asterisk-users] DUNDi and reinvites...

I'm talking out my rear so someone please apply an attitude adjustment if I'm way off base. But, if you are using Dundi as a lookup engine it should know the contact information both endpoints and how to reach them perhaps not ONLY knowing how to comunicate via another asterisk box. Much li

[asterisk-users] tdd.c - How does one use this code?

How can I interface a terminal into the TDD portion of asterisk?___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Polycom phone registration problem

Rob Schall wrote: > Auto-congesting is a "inconclusive" error message from what I've found. > In this case, it probably means, "I haven't heard from that phone or the > response ping failed", etc. > > So in this case, I'd say you're right, the phone probably is giving up, > and isn't trying to re-

RE: [asterisk-users] Asterisk & MS RTC Library & Ethernet Capacity

Thank you, Gordon! Yes, MS RTC is just SIP streams of alaw. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Friday, June 08, 2007 10:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Ast

Re: [asterisk-users] Noise on FXS ports (Sangoma)

On Thu, Jun 07, 2007 at 04:50:55PM -0600, Stephen Bosch wrote: > Tzafrir Cohen wrote: > > To generate a FXS dialtone without Asterisk, use fxstest (make fxstest) > > from the zaptel source directory. > > Can I break this dial tone with DTMF? No. -- Tzafrir Cohen icq#168497

Re: [asterisk-users] Asterisk & MS RTC Library & Ethernet Capacity

On Fri, 8 Jun 2007, Asterisk wrote: Hi guys, I was wondering whether there's anyone who could share his/her experiences with using Microsoft RTC Library. In particular I am wondering what Ethernet capacity should I have in scenario of 30 people using Microsoft RTC Library for SIP communication

[asterisk-users] Unexpected behaviour shown by "meetme kick confno usernumber"

Hi, I have Asterisk 1.4.4 on my linux box. Whenever i try to kick a participant in conference say "59681446" using following command meetme kick 59681446 1 where "1" is the participant number, following are the actions that asterisk takes * IVR "You have been kicked from this conference"

[asterisk-users] Asterisk & MS RTC Library & Ethernet Capacity

Hi guys, I was wondering whether there's anyone who could share his/her experiences with using Microsoft RTC Library. In particular I am wondering what Ethernet capacity should I have in scenario of 30 people using Microsoft RTC Library for SIP communication (PBX is obviously Asterisk :-) ) concur

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

The setup. Asterisk is on a 3G Zeon Dell 2850 running Fedora Core 5/6 (all yum updates applied), the TE410 lives on it's own interrupt. Asterisk sits between our telco and a PRI enabled PBX. These are the relevant versions installed: Linux: 2.6.20-1.2316.fc5smp Zaptel: 1:1.4.2.1-34.fc5 Asterisk:

Re: [asterisk-users] Provisioning Linksys PAP2T ATA's

Nick, Pretty much - it builds the XML output on the fly, and delivers it over HTTP (PHP/Apache). It "works the same" for all of the mainstream Linksys kit - including SPA phones. Generally, where we've installed IP phones, we've also installed an Asterisk appliance in the form of a Linux b