Re: [asterisk-users] Connecting GSM Phone to Asterisk Box

On Tue, 31 Jul 2007, Jeng Yu wrote: > Hi All, > > I have a telephony project for which I need > to build a prototype to demo for management. > The prototype must work on a GSM phone network. > > In the demo system, a call from GSM phone comes > into the demo box. The demo box runs CallWeaver. >

Re: [asterisk-users] Problems building zaptel 1.4.4

On Tue, Jul 31, 2007 at 10:04:42PM -0400, hugolivude wrote: > Hi, > > I'm having trouble compiling zaptel 1.4.4 on SUSE 10.1. I'm really > only interested in getting ztdummy to work because this is a dev > machine with no zaptel h/w. SUSE 10.1 is a 2.6 kernel: > > asterisk-dev:/home/hugh # unam

[asterisk-users] Asterisk ref book

Hi all, Can some one tell me about the book name call " Asterisk Configuration Guide" comment? Or Review about this book. Does it useful for entry to medium level skills of Asterisk system? Thank you. ___ --Bandwidth and Colocation Provide

Re: [asterisk-users] Fwd: Live Answering Service with Direct SIP Connections and Light Accounts starting at $14.95/mo

My god! How could you DO such a thing! FLOG HIM! BEAT HIM! Trash his company and declare open warfare! You know... on second thought... all that takes work. Bah... forget it. I ain't movin' off the couch. N. voiplist wrote: > A thousand apologies, I use gmail and when I type Asterisk both t

Re: [asterisk-users] Fwd: Live Answering Service with Direct SIP Connections and Light Accounts starting at $14.95/mo

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 voiplist wrote: > A thousand apologies, I use gmail and when I type Asterisk both the > users and biz list drop down. Somehow I picked the wrong one and I am > terribly sorry about that. > > Please accept my apologies for this error, believe it or not

Re: [asterisk-users] Live Answering Service with Direct SIP Connections and Light Accounts starting at $14.95/mo

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 voiplist wrote: > A while back I spent days searching for an answering service that > would allow us to forward our calls via SIP or IAX but to my surprise > found nothing. *Asterisk Users Mailing List - Non-Commercial Discussion* Guess what that mea

[asterisk-users] Fwd: Live Answering Service with Direct SIP Connections and Light Accounts starting at $14.95/mo

A thousand apologies, I use gmail and when I type Asterisk both the users and biz list drop down. Somehow I picked the wrong one and I am terribly sorry about that. Please accept my apologies for this error, believe it or not it was an honest mistake. I have been on the list long enough to know t

[asterisk-users] Live Answering Service with Direct SIP Connections and Light Accounts starting at $14.95/mo

A while back I spent days searching for an answering service that would allow us to forward our calls via SIP or IAX but to my surprise found nothing. I have been a member of the Asterisk community for a few years and had Asterisk users in mind as we created our offering. I am now proud to announ

Re: [asterisk-users] TE120P in Canada

You want pri_cpe. Once the problem with the line is resolved, that will matter. Klaverstyn, David C wrote: > Hi All, > > > > I'm having problems trying to get a TE120P operational in Canada. > > > > I keep getting a congestion error when I try to make a call. I'm not > sure if my switch

Re: [asterisk-users] .call file problem

Config file parsers frequently "read until end of line" where end of line is a CR or LF. No CR or LF, no last line read. I think that bug was fixed fairly recently in Asterisk. Other software has similar issues in the past. Nitesh Divecha wrote: > Thanks Eric, > > It solved the problem by hav

Re: [asterisk-users] Problems building zaptel 1.4.4

On 7/31/07, hugolivude <[EMAIL PROTECTED]> wrote: > ./configure works: > > configure:: *** Zaptel build successfully configured *** > > but make fails: > > make -C /lib/modules/2.6.16.13-4-default/build > SUBDIRS=/usr/src/packages/SOURCES/Asterisk/zaptel-1.4.4 modules > make[2]: Entering directory

Re: [asterisk-users] asterisk or asterisknow

Hello, I prefere, asterisk Best Regards On 7/31/07, Al lists <[EMAIL PROTECTED]> wrote: > > You can use both Asterisk or AsteriskNow to have meetme (conference room) > > On 7/30/07, fateme fatah <[EMAIL PROTECTED] > wrote: > > > Hi: > > I want to have conference call service.You offer me use a

Re: [asterisk-users] Problems building zaptel 1.4.4

On 7/31/07, hugolivude <[EMAIL PROTECTED]> wrote: > That's right: > > find / -name 'autoconf.h' > > still comes up empty... > > > On 7/31/07, Matt Riddell <[EMAIL PROTECTED]> wrote: > > -BEGIN PGP SIGNED MESSAGE- > > Hash: SHA1 > > > > hugolivude wrote: > > > I should have also mentioned th

Re: [asterisk-users] Problems building zaptel 1.4.4

Hello, In Asterisk 1.4 and zaptel 1.4, don't work make linux26, zaptel and asterisk works with kernel 26, and only work with ./configure make menuselect make make install Best Regards Carlos Rojas Lima - Peru On 7/31/07, hugolivude <[EMAIL PROTECTED]> wrote: > > Hi, > > I'm having trouble c

Re: [asterisk-users] Problems building zaptel 1.4.4

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 hugolivude wrote: > I should have also mentioned that I saw in another post that I may > need to install autoconf: > > ftp://ftp.gnu.org/gnu/autoconf > > I downloaded this, did ./configure, make, make install but it did not > fix my problem. I still

Re: [asterisk-users] .call file problem

Thanks Eric, It solved the problem by having a blank line... I wonder why we need a blank line... Cheers, Nitesh Eric "ManxPower" Wieling wrote: > Make sure you have a blank line at the end of your .call file. > > Nitesh Divecha wrote: > >> Hello All, >> >> Something strange I found that m

Re: [asterisk-users] Problems building zaptel 1.4.4

I should have also mentioned that I saw in another post that I may need to install autoconf: ftp://ftp.gnu.org/gnu/autoconf I downloaded this, did ./configure, make, make install but it did not fix my problem. I still cannot find autoconf.h... Thanks, H On 7/31/07, hugolivude <[EMAIL PROTECTED]

[asterisk-users] Problems building zaptel 1.4.4

Hi, I'm having trouble compiling zaptel 1.4.4 on SUSE 10.1. I'm really only interested in getting ztdummy to work because this is a dev machine with no zaptel h/w. SUSE 10.1 is a 2.6 kernel: asterisk-dev:/home/hugh # uname -r 2.6.16.13-4-default It seems that my problem is related to autoconf.

[asterisk-users] TE120P in Canada

Hi All, I'm having problems trying to get a TE120P operational in Canada. I keep getting a congestion error when I try to make a call. I'm not sure if my switching, parity, etc is correct. I'm hoping that someone will be able to verify my config. The Telco is SaskTel, with a 10 channe

Re: [asterisk-users] Dropouts and echo

Drat! That also could be a a major cause of the issues he is having. Dan Austin wrote: > Tom Wrote: >> Hi all, > >> We have recently implemented an Asterisk system using Trixbox >> (asterisk v1.4.4 at the moment, yet to move to 1.4.9) but are >> getting pressure to switch back to our old key

[asterisk-users] check out the cursor movement on this website!!!

picturephone -dot- com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] .call file problem

No, Atis means 'make it writable'. .call should be removable after execution. - balu raman On 7/31/07, Nitesh Divecha <[EMAIL PROTECTED]> wrote: > > Thanks Atis, > > Yes and the .call executes fine... but after 60 seconds it executes > again automatically without any application executing it. > >

Re: [asterisk-users] Dropouts and echo

But the OP stated this was SIP <-> SIP calls -- As Dan mentioned, check environmental issues -- hard walls, poor handset quality, noisy desks, volume levels too high? Eric "ManxPower" Wieling wrote: > by Cisco As for Echo Canceling, that is the job of the device that > does VoIP/PSTN gatew

Re: [asterisk-users] Dropouts and echo

Tom Wrote: > Hi all, > We have recently implemented an Asterisk system using Trixbox > (asterisk v1.4.4 at the moment, yet to move to 1.4.9) but are > getting pressure to switch back to our old key system unless > we fix two major issues. So please help me avoid switching back! Have you trie

Re: [asterisk-users] Dropouts and echo

Turn OFF CDP on the phones. I don't know if those phones support CDP, but since CDP is the Cisco Discovery Protocol and those Linksys is owned by Cisco As for Echo Canceling, that is the job of the device that does VoIP/PSTN gateway functions. Tom Lanyon wrote: > Hi all, > > We have rece

Re: [asterisk-users] outbound caller ID

On 7/30/07, Vieri <[EMAIL PROTECTED]> wrote: > > > I would like to know if one can set the outgoing > caller ID within Asterisk when calls are going out > through: > > 1) an analog POTS line (I suppose not) No. 2) a telco BRI line (I don't think so) > 3) a telco PRI line (maybe) Both are the s

Re: [asterisk-users] Connecting GSM Phone to Asterisk Box

Chan_bluetooth is now chan_mobile and included in trunk/asterisk-addons. That would be my suggestion. It works very well. Thanks, Steve Totaro Andrew Joakimsen wrote: > > > On 7/31/07, *Jeng Yu* <[EMAIL PROTECTED] > > wrote: > > Hi All, > > I have a telephony

Re: [asterisk-users] Lightweight IAX balancer

I have not had a chance to look at this, but if it is a fully functional and threaded iax load balancer, thats extremely cool. -bk Stanisław Pitucha wrote: > Very low chances for that module if any. I haven't been using OpenSER much > and I don't think I'll be using it soon - but who knows. Let

Re: [asterisk-users] Turn off SIP 183 Session Progress in Asterisk 1.4.8

On 7/31/07, Richard Brady <[EMAIL PROTECTED]> wrote: > > [Resent due to non-descriptive subject line.] > > Hi folks > > When connecting two SIP users, is there any way to stop Asterisk from > sending SIP 183 Session Progress messages, either globally or > per-peer? Yes, the option is progressinba

Re: [asterisk-users] asterisk on 64-bit?

On 7/31/07, Benjamin Jacob <[EMAIL PROTECTED]> wrote: > > Hello ppl, > Searched all over, but couldn't find anything conclusive. > Does an off-the-shelf version of Asterisk run without any issues on a > 64-bit machine? > Does anyone have any 'conclusive' figures? > > Apologies if this is a repeat q

Re: [asterisk-users] Connecting GSM Phone to Asterisk Box

On 7/31/07, Jeng Yu <[EMAIL PROTECTED]> wrote: > > Hi All, > > I have a telephony project for which I need > to build a prototype to demo for management. > The prototype must work on a GSM phone network. > > In the demo system, a call from GSM phone comes > into the demo box. The demo box runs Call

Re: [asterisk-users] DTMF integration pana d500

Well, what i described it's called "follow id" first you hook up , and then the pbx sends you the digits, after that the call is connected until you hang up; this is the most common method for connecting voicemails to pbx (the analogic way). I think for the caller id you need an add-on card in the

[asterisk-users] Number disappears when picking up a call

Hi, when I pick up a call with *8, the number of the caller isn't show on the phone that picked up the call. Is there a way/chance to keep or transfer the number of the caller? We are currently using Asterisk version 1.4.1. Thanks for any hints, Stefan --

Re: [asterisk-users] Royalty for On Hold Music ?

On Tue, Jul 31, 2007 at 05:22:20PM -0400, Jon Pounder wrote: > Quoting John Millican <[EMAIL PROTECTED]>: > there are plenty of radio stations with internet feeds of their audio, > piping that in would not change any coverage area since anyone with > internet could listen anywhere already, you

[asterisk-users] Dropouts and echo

Hi all, We have recently implemented an Asterisk system using Trixbox (asterisk v1.4.4 at the moment, yet to move to 1.4.9) but are getting pressure to switch back to our old key system unless we fix two major issues. So please help me avoid switching back! An overview: We have about 12 Li

Re: [asterisk-users] DTMF integration pana d500

Yes and No The D500 is a terrible thing First problem: it sends some horrible DTMF, so if your voicemail is configured to send #H and *H, it will not work, configure it to send numbers, like 8H and 9H (H is a placeholder for the extension). I also managed to use the MWI (message light), it's a perl

[asterisk-users] not hearing dtmf tones

I have a polycom 501 phone, when I call into the dialplan and do the following I do not hear the DTMF digit. exten => 51,1,playback(invalid) exten => 51,2,SendDTMF(1) This is on the 1.2.23 asterisk. I am trying to re-create calling sendDTMF in an agi and not hearing the digit either. The above

Re: [asterisk-users] Royalty for On Hold Music ?

Quoting John Millican <[EMAIL PROTECTED]>: there are plenty of radio stations with internet feeds of their audio, piping that in would not change any coverage area since anyone with internet could listen anywhere already, you're only providing that to the listener through a phone handset in

Re: [asterisk-users] Royalty for On Hold Music ?

On Tuesday July 31 2007 4:44 pm, Joe acquisto wrote: > . . . > > > Even if you can find non-original-artist recordings of such music, the > > *compositions* are registered with BMI and ASCAP, and you'll need > > blanket licenses to play them. (Well, if you only wanted one or two > > tracks, you mi

Re: [asterisk-users] 1and1 dedicated servers have been down for a few hours .

my shared webhosting is going strong... daveC Asterisk guy wrote: > 1and1 dedicated server's service has been down for a few hours , > unable to reach them by phone or email. do anyone know what is going > on there ? > > Mario > --

Re: [asterisk-users] Royalty for On Hold Music ?

. . . > Even if you can find non-original-artist recordings of such music, the > *compositions* are registered with BMI and ASCAP, and you'll need > blanket licenses to play them. (Well, if you only wanted one or two > tracks, you might negotiate specific licenses, but I'm not sure it > would be c

Re: [asterisk-users] .call file problem

Make sure you have a blank line at the end of your .call file. Nitesh Divecha wrote: > Hello All, > > Something strange I found that my .call file is running twice... > Just after 60 sec it will run again, without any application invoking it. > > This is my .call file: - > = > Ch

Re: [asterisk-users] .call file problem

Thanks Atis, Yes and the .call executes fine... but after 60 seconds it executes again automatically without any application executing it. Cheers, Nitesh Atis wrote: > Is your .call file writable by asterisk? > > $ chmod 777 sample.call > > On 7/31/07, Nitesh Divecha <[EMAIL PROTECTED]> wrote

Re: [asterisk-users] Description for each sound files

On Tue, Jul 31, 2007 at 02:46:15PM -0500, Kevin P. Fleming wrote: > Tzafrir Cohen wrote: > > > It seems that a bunch of 1.2 users answered this guy and wondered why > > this text file is not there. It's there in 1.2 and not there in 1.4, and > > hence also removed from the Debian package. > > >

Re: [asterisk-users] Grandstream RTPkeepalive packetscausing Asteriskwarning

Thanks Steve, I'll keep digging, regards, Drew Steve Langstaff wrote: Sorry Drew - I don't know enough about Asterisk to comment on the effects of this on the RTP stack. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Drew Gibson *Sent:* 31 July 2007 15:0

Re: [asterisk-users] .call file problem

Is your .call file writable by asterisk? $ chmod 777 sample.call On 7/31/07, Nitesh Divecha <[EMAIL PROTECTED]> wrote: > Hello All, > > Something strange I found that my .call file is running twice... > Just after 60 sec it will run again, without any application invoking it. > > This is my .call

[asterisk-users] Problems using TE412P and TDM400B in a IBM x3650

Another day, another apparant unexplained hardware incompatibility. I have a TE412P and a TDM400B living quite happily in a whitebox using an Intel motherboard: http://www.intel.com/design/servers/boards/se7230nh1-e/index.htm I tried to move to an IBM x3650 system. It uses a slightly newer chip

[asterisk-users] .call file problem

Hello All, Something strange I found that my .call file is running twice... Just after 60 sec it will run again, without any application invoking it. This is my .call file: - = Channel: SIP/xo-out/19097773456 Callerid: 9097773456 MaxRetries: 3 RetryTime: 30 WaitTime: 15 Context: c

Re: [asterisk-users] Royalty for On Hold Music ?

On Tue, Jul 31, 2007 at 03:05:32PM -0400, Jay R. Ashworth wrote: > On Tue, Jul 31, 2007 at 01:37:00PM -0500, voiplist wrote: > > I have done this in the past and I don't recall ever finding any > > "popular" music by "popular" artist. > > For example, if I wanted to play oh I don't know an origina

Re: [asterisk-users] software bloat - is this really useful to anyone?

Tzafrir Cohen wrote: > OTOH: /usr/sbin/asterisk depends on libopenh323 . Now this *is* bloat. Did you bother to check into why this is the case before spreading this unfounded information on the mailing list? First of all, this is *NOT* true unless the chan_h323 module was built, which the vast

Re: [asterisk-users] Description for each sound files

Tzafrir Cohen wrote: > It seems that a bunch of 1.2 users answered this guy and wondered why > this text file is not there. It's there in 1.2 and not there in 1.4, and > hence also removed from the Debian package. > > I didn't see any open bug about this but I vaguely recall some folks > asking

Re: [asterisk-users] PhpAgi call generation

Thanks Nasir, That helped alot... Cheers, Nitesh Nasir Iqbal wrote: > Oh, > > you need Dial application instead of origination. > > so no need to AGI Script simply add > > > the dialplan called for ".call" should look like this > > exten => yourexten,1,BackGround(your_menu_ivr) > exten => youre

Re: [asterisk-users] Reboot Grandstream Phone

Ok, This tools directly log on the web interface, so if anyone knows about the SIP NOTIFY message ... Thanks yann -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Gordon Henderson Envoyé : mardi 31 juillet 2007 18:17 À : Asterisk Users Mailing List - N

Re: [asterisk-users] Royalty for On Hold Music ?

There's no royalty free popular songs by popular artists. Not only would you have to pay royalties, you would have to secure the rights from the artists' representatives just to get the permission to play the songs in the first place. Darrell S. Long BestWeb Corporation john beaman wrote:

Re: [asterisk-users] Royalty for On Hold Music ?

On Tue, Jul 31, 2007 at 01:37:00PM -0500, voiplist wrote: > I have done this in the past and I don't recall ever finding any > "popular" music by "popular" artist. > > For example, if I wanted to play oh I don't know an original song > performed by the original artist such as Nora Jones or The Bea

Re: [asterisk-users] Royalty for On Hold Music ?

No, you will not. According to the music industry those artists are all are entitled to compensation for every time their song is broadcast, which includes MoH. AFAIK, there are no "popular songs" by "popular artists" that are royalty-free. John Beaman Telecom Specialist Voice Telecommunica

Re: [asterisk-users] Wrinkled faxes or missing lines with Hylafax + IAXModem + Asterisk

We have a PRI and use a sangoma a101d to a PRI. The Asterisk and IAXModem are on the same box. Here is a link to the output from cat /proc/interrupts http://fluxbox.pastebin.ca/640841 I put it here since you recommended putting this question on the IAXModem list. Thanks for any help Lee Howar

Re: [asterisk-users] Royalty for On Hold Music ?

I have done this in the past and I don't recall ever finding any "popular" music by "popular" artist. For example, if I wanted to play oh I don't know an original song performed by the original artist such as Nora Jones or The Beatles will I find this sort of thing at a Royalty Free Site? On 7/

Re: [asterisk-users] Royalty for On Hold Music ?

On Tue, Jul 31, 2007 at 12:49:45PM -0500, voiplist wrote: > So is there a simple way to license decent, up to date music? Can I > just go to a website, click a buy button, pay my money and download > the song? In some cases, yes. > It seems idiotic that you need 15 lawyers and a million bucks use

[asterisk-users] FW: NYC Special Event - Join Us

Anyone know what they are going to be announcing? Any relevance to Asterisk? When are we going to see a decent video conferencing implementation into Asterisk? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Syd

Re: [asterisk-users] Royalty for On Hold Music ?

Just Google for: royalty free music, and will find plenty of sites that will serve your needs. John Beaman Telecom Specialist Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 >>> [EMAIL PROTECTED] 7/31/2007 12:49:45 PM >>> So is there a simple way to li

Re: [asterisk-users] PhpAgi call generation

Oh, you need Dial application instead of origination. so no need to AGI Script simply add the dialplan called for ".call" should look like this exten => yourexten,1,BackGround(your_menu_ivr) exten => yourexten,n,WaitExten() exten => 1,1,Dial(SIP/xo-out/$supervisor_num) ;for Supervisor exten =

Re: [asterisk-users] Wrinkled faxes or missing lines with Hylafax + IAXModem + Asterisk

Michael Rice wrote: >Hello. > >We are running Asterisk 1.2.23 iaxmodem-0.2.1 and hylafax-4.3.3 > >When we send faxes the people who receive the faxes complain that they >look wrinkled or smashed up. Sometimes they are missing random lines. >Has anyone seen this happen, or know how to fix it? >

Re: [asterisk-users] Unicall/Dont know how to handle Accepted

Victor Toofic wrote: > El Sun, Jul 29 de 2007 a las 20:04 +0800, Steve Underwood comentaba: > >> What versions of software did you use to get a screwed up result like >> that? The message "Don't know how to handle signalling event Accepted" >> is printed at the end of a case statement which do

Re: [asterisk-users] Royalty for On Hold Music ?

So is there a simple way to license decent, up to date music? Can I just go to a website, click a buy button, pay my money and download the song? It seems idiotic that you need 15 lawyers and a million bucks use decent on hold music. Maybe I just don't know the procedure. I am all for paying the

[asterisk-users] AsteriskNOW and Custom VoIP

Guys, I've downloaded AsteriskNOW few days ago so I'm new to this product. The first issue is on service provider area. I've already used a VoIP account already configured with my ISP, it works fine! This configuration has been used until now with the client SJphone, Now I would use this prof

[asterisk-users] Repost: Aastra BLF directed pickup: anyone have this pickup-mgernoth file?

Hello all Sorry for the repost, but I believe this is essential for Aastra phones to use directed pickup with BLF in 1.2 (and it's still experimental in 1.4 I think). pickup-mgernoth-2006-07-28.patch.txt If anyone knows otherwise than that this is still required in latest 1.2 or 1.4 please reply

[asterisk-users] Connecting GSM Phone to Asterisk Box

Hi All, I have a telephony project for which I need to build a prototype to demo for management. The prototype must work on a GSM phone network. In the demo system, a call from GSM phone comes into the demo box. The demo box runs CallWeaver. Callweaver picks up the GSM call, answers it and plays

[asterisk-users] AstriCon -- Last chance to speak!

We're getting close to wrapping up the final list of speakers for AstriCon, but wanted to give the Asterisk community one more chance to speak up and be heard. If you're interested in presenting at AstriCon, please go to http://www.astricon.net/ and click on the "Speak at AstriCon" link on the rig

[asterisk-users] Wrinkled faxes or missing lines with Hylafax + IAXModem + Asterisk

Hello. We are running Asterisk 1.2.23 iaxmodem-0.2.1 and hylafax-4.3.3 When we send faxes the people who receive the faxes complain that they look wrinkled or smashed up. Sometimes they are missing random lines. Has anyone seen this happen, or know how to fix it?

Re: [asterisk-users] Queue Time to Speak to Agent Algorithm?

On Tue, 2007-07-31 at 07:51 -0800, Mojo with Horan & Company, LLC wrote: > Can you verify the correctness of that URL? 404 from here. > Thanks! Strange... it works for me here. Maybe one of the web mirrors is out of sync? Anyway, if you're got the source code to Asterisk 1.4 handy, it should be

Re: [asterisk-users] Zaptel update trouble

We're using a TE212P card with one PRI connection. Thanks, Nick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Tuesday, July 31, 2007 11:27 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Zaptel update trouble On

Re: [asterisk-users] g729 setup help

Patrick, Make sure you have install the g729 modules correctly as per the instructions and restarted Asterisk. Other method is you can configure your Asterisk to do Pass-thru g729 which you don't require to install any g729 license on Asterisk. As far as both of your phones has g729 installed

Re: [asterisk-users] Royalty for On Hold Music ?

Thanks Jared, Yes I am using with Asterisk only. So I am using the inbuilt music from Asterisk for onhold. -- Deepak Jared Smith <[EMAIL PROTECTED]> wrote: On Tue, 2007-07-31 at 06:36 +0100, Deepak Naidu wrote: > I think we need to pay for the later, but I am not sure if we need to > pay for t

Re: [asterisk-users] SunRocket / ALLO / etc special offer

Matt wrote: > I'll take either > > Actually now that I have had a chance to think about what I did (sorry > bad week here). Yes, I will admit I did patrionize the users list... > sorry if I offended anyone. I just figured I'd try to help any > SunRocket users out that may not be on the biz l

Re: [asterisk-users] Silly MeetMe() question.

On Tue, 31 Jul 2007, Jaswinder Singh wrote: > Do you have proper version of zaptel installed corresponding to your > asterisk version ? At the time that I downloaded the source I got the latest version of zaptel as well. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/

Re: [asterisk-users] Silly MeetMe() question.

Do you have proper version of zaptel installed corresponding to your asterisk version ? On 31/07/07, Knud Müller <[EMAIL PROTECTED]> wrote: > > Alex Balashov schrieb: > > On Mon, 30 Jul 2007, Knud Müller wrote: > > > >> what does your modules directory contain? Can you find a file > >> /usr/lib/as

[asterisk-users] OT: Polycom Directory XML via PHP

Hi guys, Any one know about any php or other GUI to manage Polycom Directory XML contacts ? I would like to setup a web page were I manage Directory contacts to provide XML file to feed my polycom phones on Boot with all the contacts. Does any one has already done this? I notice a bash script on

Re: [asterisk-users] 1and1 dedicated servers have been down for a few hours .

Is this a web hosting forum or mailing list ? On 31/07/07, Asterisk guy <[EMAIL PROTECTED]> wrote: > > 1and1 dedicated server's service has been down for a few hours , unable > to reach them by phone or email. do anyone know what is going on there ? > > Mario > > ___

Re: [asterisk-users] Zaptel update trouble

On Tue, Jul 31, 2007 at 10:42:15AM -0500, Nick Whitaker wrote: > Hi all, > > When I try to update zaptel to 1.4.3 or 1.4.4 it seems to hang the > server at "loading zap hardware modules". I can install 1.4.2.1 and it > will load OK at startup although I have to run > ../xpp/utils/genzaptelconf -

Re: [asterisk-users] PhpAgi call generation

Thanks Nasir, By putting "'Exten'=> your_extensions_here" it will create a new channel to that extension, correct? What I want to do is to join two channels... Join the User A channel which is active with supervisor. Cheers, Nitesh Nasir Iqbal wrote: > Hi Nitesh, > > you are missing Extensi

Re: [asterisk-users] Reboot Grandstream Phone

On Tue, 31 Jul 2007, Yann JOUANIN wrote: > Hi all, > > There is a patch in the Mantis that provides a function to reboot GS phones. > In order to use it, I need to know how the digest for SIP Notify messages is > calculated. > > If anyone knows that, please let me now I don't know about the SIP m

Re: [asterisk-users] PhpAgi call generation

Hi Nitesh, you are missing Extension try with $call = $asm->send_request('Originate', array('Channel'=>"SIP/xo-out/$supervisor_num", 'Context'=>'default', 'Exten'=> your_extensions_here, 'Priority'=>1, 'Callerid'=>$cid)); or

[asterisk-users] Reboot Grandstream Phone

Hi all, There is a patch in the Mantis that provides a function to reboot GS phones. In order to use it, I need to know how the digest for SIP Notify messages is calculated. If anyone knows that, please let me now Regards, Yann ___ --Bandwidth and

Re: [asterisk-users] Queue Time to Speak to Agent Algorithm?

Can you verify the correctness of that URL? 404 from here. Thanks! Jared Smith wrote: > On Tue, 2007-07-31 at 10:16 -0400, Matt wrote: >> Does anyone know the algorithm that Asterisk uses to figure out when >> you'll be speaking with an agent?I've heard it say such bizarre >> things as '2 min

[asterisk-users] Zaptel update trouble

Hi all, When I try to update zaptel to 1.4.3 or 1.4.4 it seems to hang the server at "loading zap hardware modules". I can install 1.4.2.1 and it will load OK at startup although I have to run ../xpp/utils/genzaptelconf -u before rebooting the server or I run into trouble on shutdown. I tried r

[asterisk-users] g729 setup help

Hi I am trying to make this setup work phone1---g729---asterisk1---sip---asterisk2---g729---phone2 I have tried several configurations but none worked I keep getting transcoding errors I have installed one g729 licence on each asterisk, but I can't verifiy because the show g729 command is not

Re: [asterisk-users] Queue Time to Speak to Agent Algorithm?

On Tue, 2007-07-31 at 10:16 -0400, Matt wrote: > Does anyone know the algorithm that Asterisk uses to figure out when > you'll be speaking with an agent?I've heard it say such bizarre > things as '2 minutes' when there area 2 calls waiting ahead of a > person and 1 agent logged in. I don't kno

Re: [asterisk-users] Welcome to the "asterisk-users" mailing list (Digest mode)

I meant to send that just to you, not the list - My apologies. I wasn't trying to be the public list cop. AR On 7/31/07, Richard Brady <[EMAIL PROTECTED]> wrote: > > Hi Alex > > Apologies for that, I noticed this immediately after I sent the email > and resent it with fresh headers and a descript

Re: [asterisk-users] Welcome to the "asterisk-users" mailing list (Digest mode)

Hi Alex Apologies for that, I noticed this immediately after I sent the email and resent it with fresh headers and a descriptive subject line. I will be more careful in future. Regards, Richard -- Richard Brady T: +44 (0)7771 623 348 E: [EMAIL PROTECTED] ___

Re: [asterisk-users] Grandstream RTPkeepalive packetscausing Asteriskwarning

Sorry Drew - I don't know enough about Asterisk to comment on the effects of this on the RTP stack. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Drew Gibson Sent: 31 July 2007 15:02 To: Asterisk Users Mailing List - Non-C

[asterisk-users] Queue Time to Speak to Agent Algorithm?

Does anyone know the algorithm that Asterisk uses to figure out when you'll be speaking with an agent?I've heard it say such bizarre things as '2 minutes' when there area 2 calls waiting ahead of a person and 1 agent logged in. ___ --Bandwidth and Co

[asterisk-users] PhpAgi call generation

Hello All, Can anyone help me with this... This is what my program does: - 1) At certain time the system generates a ".call" and make a call to User A. 2) When User A picks up the phone call, system will play a menu select option. a) Press 1 to call your supervisor. b) Press 2 to

Re: [asterisk-users] Royalty for On Hold Music ?

On Tue, Jul 31, 2007 at 07:25:00AM -0400, Jared Smith wrote: > On Tue, 2007-07-31 at 06:36 +0100, Deepak Naidu wrote: > > I think we need to pay for the later, but I am not sure if we need to > > pay for the inbuilt asterisk(freepbx) on hold music. > > I'm no lawyer, but here's what I understand.

Re: [asterisk-users] Grandstream RTP keepalive packetscausing Asteriskwarning

Thanks Steve, am I correct in assuming that Asterisk will ignore this packet? It will not have any harmful effect if the caller is on hold for an hour (120 packets)? In order to feed back to Grandstream, is there a "correct" format for a keepalive message or is this one adequate? regards,

Re: [asterisk-users] Welcome to the "asterisk-users" mailing list (Digest mode)

Please start new threads for new messages (don't "reply" and just wipe out the body). The headers still exist so you wind up with screwy threading in the list archives (ditto for those of us who have e-mail software that supports threading). AR On 7/31/07, Richard Brady <[EMAIL PROTECTED]> wrote:

[asterisk-users] ExternalIVR() broken in 1.4.9?

Hello, After upgrading to 1.4.9 the above function does not work anymore; it claims that "child went away" while the child is probably not born at all... Before I open a bug on it, anyone has a clue? Thanks! __Yehavi:

Re: [asterisk-users] Zaptel channel reservation

Jack wrote: > Actually I preferred to reserve the channels in asterisk, but this seems > to be the easiest way. > > Does anybody know if the mapping from telco channels to zap channels is > fixed? Is the first telco channel always mapped to the first zap channel > or is this mapping dynamic?

[asterisk-users] Turn off SIP 183 Session Progress in Asterisk 1.4.8

[Resent due to non-descriptive subject line.] Hi folks When connecting two SIP users, is there any way to stop Asterisk from sending SIP 183 Session Progress messages, either globally or per-peer? Scenario as follows: Call from UA1 to Asterisk (UA2) to UA3. UA3 sends RTP before SIP OK to Ast

Re: [asterisk-users] Welcome to the "asterisk-users" mailing list (Digest mode)

Hi folks When connecting two SIP users, is there any way to stop Asterisk from sending SIP 183 Session Progress messages, either globally or per-peer? Call from UA1 to Asterisk (UA2) to UA3 UA3 sends RTP before SIP OK to Asterisk (UA2) Asterisk (UA2) detects early audio from UA3 and sends 183 Ses

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