[asterisk-users] Integrating the Openser for VoiceMail and PBX with Asterisk, For Account

Hi Users,I'm  new to Asterisk programming , I'm in working the Voip Technologies by using the  OpenSER for my call routing process and Radius For AAA.But in Asterisk i need   it for only PBX and VoiceMail,For Account  I'm using the  Openser + Radius . Main My doubt is  that,    For

[asterisk-users] Asterisk as a gateway to SER

My asterisk is giving me problems when I use it as a pstn gateway to SER , basically what happens is that its either I get one way audio or no audio at all when I make pstn calls via asterisk from sip clients registered with SER. ___ --Bandwidth and Colo

[asterisk-users] Polycom Expansion Module

Has anyone used the Polycom expansion module with multiple lines? My application is for 20 lines and read there was a limit of 7 at one point. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE

RE: [asterisk-users] saved.gsm -> Voicemail greeting ??

Yes, it does. :) Ok Then, I guess my issue is solved until I see a glitch because i am lacking the .wav|.WAV versions. Thanks All! -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Fri, 15 Sep 2006, Bill Gibbs wrote: I assume

RE: [asterisk-users] saved.gsm -> Voicemail greeting ??

Part of the directory as well as when you get to leave that person a VM (instead of saying the user at extension blah blah blah is unavailable it will read back the greeting file) are the 2 places I have heard it so far. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTE

Re: [asterisk-users] Scaling/Loadbalancing a Call Center and Redundancy

How many lines and agents are you looking at? What kind of call volume? Average expected hold time? VICIDIAL could be an option for you since it does not use Asterisk Queues and can already easily scale across many servers. MATT--- On 9/15/06, Steve Totaro <[EMAIL PROTECTED]> wrote: I have

RE: [asterisk-users] saved.gsm -> Voicemail greeting ??

Trying that now... umm, anyone know what condition makes use of just the "name" in voicemail, is that part of the directory or something? -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Fri, 15 Sep 2006, Bill Gibbs wrote: I assume

RE: [asterisk-users] saved.gsm -> Voicemail greeting ??

I assume it will use the files .gsm too? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs Sent: Friday, September 15, 2006 10:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] saved.gsm -> Voice

Re: [asterisk-users] saved.gsm -> Voicemail greeting ??

Ok, I am getting closer on how to choose the files played. I missed this in the book, but VoIP-info had this: s: Play nothing. (no flags): Play instructions. su: Play unavailable message. u: Play unavailable message, then instructions. sb: Play busy message. b: Play busy message, then instruc

[asterisk-users] call across 2 asterisks

How can I make this work (UA1 makes call to UA2)? If I issue a command Dial(SIP/UA2), the call will fail. Someone in the forum said that Asterisk doesn't support the configuration below. But I am thinking about the possibility to make it works. Anyone can share if it really works? UA1

RE: [asterisk-users] saved.gsm -> Voicemail greeting ??

Example for mailbox 100 under context default /var/spool/asterisk/voicemail/default/100 -rwx-w 1 asterisk asterisk 442604 Aug 22 16:44 busy.wav -rwx-w 1 asterisk asterisk 44976 Aug 22 16:44 busy.WAV -rwx-w 1 asterisk asterisk 25964 Aug 8 02:17 greet.wav -rwx-w 1 asterisk asteri

[asterisk-users] Scaling/Loadbalancing a Call Center and Redundancy

I have been tossing around some ideas about scaling a call center with load balancing and redundancy and would like the comunities input, thoughts, criticism and anything anyone wants to toss in. The most evident thing is to start with beefy servers and only run procs that are required. All o

[asterisk-users] amr codec

I have been searching, but I have not found the answer.. How might I add the amr codec to my asterisk server? I believe I found the amr source from http://www.3gpp.org/ftp/Specs/latest/Rel-6/26_series/26073-600.zip I compiled it but did not end up with any .so files like I thought I would need to

Re: [asterisk-users] saved.gsm -> Voicemail greeting ??

Hi John, Yes, I followed an example that put all my family sound files in /var/lib/asterisk/sounds/local, which is also where this file is. Now I am trying to figure out how to get the unavailable|name|Busy .gsm's I made loaded into a mailbox without playing my sounds back into a phone ;) N

[asterisk-users] saved.gsm -> Voicemail greeting ??

Check in /var/spool/asterisk/voicemail/default/ for a particular extension, don't know how you want to differentiate after hours, etc. Also, you can put files in /var/lib/asterisk/sounds/custom and do with them what you want. on Friday 09/15/2006 Nick Ellson([EMAIL PROTECTED]) wrote > > I see

Re: [asterisk-users] saved.gsm -> Voicemail greeting ??

Ok, I did finally find the file structure (/var/spool/asterisk/voicemail/default//*) , but everything is stored in 3 formats with set names, so I bet there is a process that creates all that, if I have just the 8000hz mono .gsm, is there an entry point or program I can feed this file to? T

[asterisk-users] saved.gsm -> Voicemail greeting ??

I seem to have stumped myself on this one. I had my son rattle off some really great sound bytes for his own extension (busy, after hours, etc) and that was easy to set up with the dial plan. Now I have his actual VM greeting in a .gsm and no idea how to get it into his VM Greeting, I am gue

[asterisk-users] voxee, callerid and trixbox

Anyone have a procedure on how to get caller id to work with trixbox? theres a procedure for it to work with asterisk, but not asterisk. thanks alot.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or

[asterisk-users] Help spread the word about Asterisk!

Recently, Network World published an article about a Texas university migrating their 6,000 students from a Cisco VoIP solution to Asterisk. This is the best example to date of a large-scale Asterisk deployment, considering how secretive the numbers are and where. So, help push this news to the top

[asterisk-users] AEL2 patch for Asterisk 1.2.12.1

I want to try AEL2. This page: http://voip-info.org/wiki/view/Asterisk+AEL2 gives instructions to generate a patch from subversion: svn diff http://svn.digium.com/svn/asterisk/branches/1.2 http://svn.digium.com/svn/asterisk/team/murf/AEL2-1.2 > AEL.patch My doubt is: is "branches/1.2" the lat

[asterisk-users] pickupgroup 1

I have problems with pickupgroup. While 621 can pickup a call to 601 with *8, no phone can pickup a call to 621. Below are the settings for two phones. 601 is static in the sip.conf, while 621 is in the Real-time database. What could be the problem? I have an extension 601: [601] type=friend

[asterisk-users] DTMF Tone Not Passing Help

Hello All, Can anyone help me with this DTMF tone problem. I am running Asterisk 1.2.9.1 svn with couple of Polycom and Snom phones. None of the phones are passing the DTMF tones to remote IVR. Called 1-800-FLOWERS, when asked to press 1, none of the phones were able to transmit the digit

Re: [asterisk-users] Asterisk with cisco 7935

Miles Scruggs wrote: What did you end up using for conference stations? Polycom IP 501s Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ --B

Re: [asterisk-users] Asterisk with cisco 7935

Just wondering if anyone has had any luck getting the cisco 7935 working with asterisk and if so, what is the best way to go about it? on the My testing shows it was a wasted purchase. Using CHAN_SCCP I was able to get it to work, but not stably (i.e. keys stopped functioning, phone lo

Re: [asterisk-users] where download app_txfax?

On 9/15/06, Jerry Geis <[EMAIL PROTECTED]> wrote: I have gone to http://soft-switch.org/downloads/spandsp/ looking to app_rxfax and I dont see it? Where is it? http://soft-switch.org/downloads/spandsp/spandsp-0.0.2pre26/asterisk-1.2.x/app_rxfax.c http://soft-switch.org/downloads/spandsp/spandsp-

[asterisk-users] Re: 4-wire analogue interfaces?

In article <[EMAIL PROTECTED]>, Shane Young <[EMAIL PROTECTED]> wrote: > Quoting Tony Mountifield <[EMAIL PROTECTED]>: > > > Does anyone know of any 4-wire analogue interface cards that could be > > made to work with Asterisk? (I'm not averse to hacking channel drivers) > > A T1 card to a D4 bank

[asterisk-users] FollowMe question

Group  Does anyone have the FollowMe sound files? Do I need to record them? Also does anyone have a working followme.conf file that they would share? Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To U

[asterisk-users] Digium G.729 codec now available for Solaris/SPARC

I have just uploaded Solaris/SPARC G.729 codecs and register tools to our FTP site; they are located with the other non-Linux codecs in the 'unsupported' directory of the g729 subdirectory. There are versions for both Asterisk 1.2 and Asterisk 1.4 in both 32-bit and 64-bit flavors; we will be up

Re: [asterisk-users] Hangup on Panasonic KX-TEM824

No way if you are using fxs on panasonic and fxo on *. jorge [EMAIL PROTECTED] wrote: > I have an Asterisk box connected with a Panasonic KX-TEM824 and can not detect > HANGUP from this. Can anyone help me to get it work. Thanks! > > ___ > --Bandwidth

Re: [asterisk-users] Reliability of the newer IAXy's

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 John Novack wrote: > Regarding the IAXy, newer model- S101i > I have an application for one. Both the IAXy and the Asterisk would be > behind routers ( cheap Linksys ones ) , both ends with a dynamic ( > subject to change ) IP address. > I have RTFM,

RE: [asterisk-users] blf aastra 9133i working but can't pickup calls

I got a chance to patch my Asterisk server this afternoon and was able to confirm that the directed call pickup feature is working (at least for me).  I’m running Asterisk 1.2.12.1 and used the latest pickup patch (http://bugs.digium.com/view.php?id=5014, pickup-mgernoth-2006-07-28.patch.tx

Re: [asterisk-users] Asterisk 1.2.12.1 and Zaptel 1.2.9.1 Released

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 The Asterisk Development Team wrote: > Earlier this week 'refresh' releases of these two projects were put on > our FTP servers, but due to some miscommunication on our end no > announcements were sent out... so here they are :-) :) All good, Daily A

RE: [asterisk-users] Reliability of the newer IAXy's

Not sure what to tell you. But for the price, I might have to try one of these instead: http://cgi.ebay.com/IAX-Native-FXS-for-Digium-Asterisk-VoIP-PBX-Beats-IAXy_W0QQitemZ130026839028QQihZ003QQcategoryZ61841QQssPageNameZWDVWQQrdZ1QQcmdZViewItem - Original Message - From: [EMAIL PROT

Re: [asterisk-users] Reliability of the newer IAXy's

This goes for everyone; With an Asterisk server directly assigned a public static IP and with the client SIP devices behind a NAT (usually Linksys routers) I have yet to have a NAT issue, even when the client is behind dual NATs. On 9/15/06, John Novack <[EMAIL PROTECTED]> wrote: Regarding th

Re: [asterisk-users] How to send DTMF down a channel

could you post the output of the asterisk console in verbose mode? In logger.conf [logfiles] console => notice,warning,error,verbose,debug Regards On 9/15/06, Frank Church <[EMAIL PROTECTED]> wrote: The program in question is an adaptation an AGI calling card program. It is adapted for callba

Re: [asterisk-users] Reliability of the newer IAXy's

The sipura stuff (and lots of other ata's) work just fine behind most nat boxes "if" the asterisk box is on a registered IP. John Novack wrote: Regarding the IAXy, newer model- S101i I have an application for one. Both the IAXy and the Asterisk would be behind routers ( cheap Linksys ones )

[asterisk-users] Asterisk 1.2.12.1 and Zaptel 1.2.9.1 Released

Earlier this week 'refresh' releases of these two projects were put on our FTP servers, but due to some miscommunication on our end no announcements were sent out... so here they are :-) Asterisk 1.2.12.1 fixes one significant bug that was introduced after 1.2.10 but which hadn't been corrected ye

Re: [asterisk-users] Sphinx2

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Douglas Garstang wrote: > Sphinx must have been written by the same people as Asterisk then... Yeah, because the people over at AsteriskDocs wrote Asterisk? Seriously Doug, if you're looking for a supported commercial implementation that works with A

Re: [asterisk-users] Asterisk variables

No such variable exists, why dont you set your own variable before calling goto ? Regards On 9/15/06, Mike <[EMAIL PROTECTED]> wrote: Is it possible to know, when an extension is reached through a Goto command, what the context of the Goto command was? Useless but representive example: [con

Re: [asterisk-users] Reliability of the newer IAXy's

Regarding the IAXy, newer model- S101i I have an application for one. Both the IAXy and the Asterisk would be behind routers ( cheap Linksys ones ) , both ends with a dynamic ( subject to change ) IP address. I have RTFM, such as it is, and really don't see how it can be properly configured

RE: [asterisk-users] Reliability of the newer IAXy's

Thanks for the reply. It was the type of info I was looking for. I was mostly interested because of the IAX protocol and being able to more easily traverse NAT situations. But for the price, I tend to agree and why I have been hesitant at buying one. Robert - Original Message - From

[asterisk-users] Asterisk variables

Is it possible to know, when an extension is reached through a Goto command, what the context of the Goto command was?   Useless but representive example:   [context_a] exten => 99,1,Goto(context_b,test,1)   [context_b] exten => test,1,Noop(${CALLING_CONTEXT})   Would output "context_a" to

RE: [asterisk-users] Sphinx2

Sphinx must have been written by the same people as Asterisk then... > -Original Message- > From: Matt Riddell (IT) [mailto:[EMAIL PROTECTED] > Sent: Friday, September 15, 2006 1:57 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Sphinx2 >

Re: [asterisk-users] Reliability of the newer IAXy's

Honestly for the price its a bad unit. If they were priced $40-50 then yes its a great unit. But at $90, single port, no web config, very basic provisioning and with the S100 we had many issues of reliablity where a SIP ATA did not have the same issues. Maybe it's heat, but thats an issue I would

Re: [asterisk-users] Sphinx2

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Douglas Garstang wrote: > What sphinx documentation? All I could find was docs on the code, not on how > to USE the software. :) One and the same! Sphinx is not a commercial application. I think you might have been mistaken, and are actually lookin

Re: [asterisk-users] Attended transfer and parking calls

Good information. We use mostly analog phones and some of our extensions are soft phones.   One question I still have:   When I use flash, I hear the "TRANSFER" prompt and if I dial an extension the call goes direct without a chance for me to talk to the guy on the other side.   Can it be th

Re: [asterisk-users] Why not g726-32?

RR wrote: All, is there anyone who uses g726-32 ? If not, then does anyone know why don't people use it? I use g726 on iax links between systems and to teliax.com for LD calls. Have no idea if its -32 or what though. What ships with asterisk (in terms of g726) has been working very well for

RE: [asterisk-users] Reliability of the newer IAXy's

They are much improved in their ability to dissipate heat. That was one issue with the original units, they had a tendency to get overheated. Cory Andrews -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lists Sent: Friday, September 15, 2006 3:38 PM To:

[asterisk-users] Reliability of the newer IAXy's

Is anyone out there currently using the newest model IAXy? I was thinking about purchasing one for testing but was wondering if they have gotten any better than the original models. Thanks ___ --Bandwidth and Colocation provided by Easynews.com --

Re: [asterisk-users] Internal message being heard on pstn line

On 9/15/06, Ryder Brook <[EMAIL PROTECTED]> wrote: I am on a sip line, ext 11, checking my voicemail. While I am in the midst of it, an incoming call from a pstn line, dialing 11, caller hears the internal message: "Asterisk mail box,... password" Obviously, the caller is confused. Where

Re: [asterisk-users] Voicemail adjustments

On 9/15/06, Ricardo Carvalho <[EMAIL PROTECTED]> wrote: Hi all, Some questions about Asterisk Voicemail adjustments I want to make: - how can I limit the number of voicemail messages stored per user in their voicemail folder? (to expire voicemail after a specified number of days I know that the

[asterisk-users] New astGUIclient VICIDIAL Release: 2.0.1

Hello, We've released another update to our astGUIclient suite: 2.0.1 http://astguiclient.sf.net/ The client suite runs on most modern web browsers on almost any GUI-capable operating system, and it includes the astGUIclient client-side web app which extends your phone's functionality and the V

Re: [asterisk-users] Attended transfer and parking calls

What sort of handset/ata are you using? On the IP phones it should all be in the menus. On the ATAs we use blind transfer would be and dial the numbe to transfer too, then hang up. Attened transfer is the same thing, except you wait for the other party to answer then hang up, the call is connecte

Re: [asterisk-users] Cisco Distinctive ring using alert-info

Eric "ManxPower" Wieling wrote: Rich Adamson wrote: Julian Lyndon-Smith wrote: I've got a cisco 7960, with (amongst many others) the following in the RINGLIST.DAT file Foghorn foghorn.raw I can manually select this for the ringtone. However, I was wanting to use a normal ringtone, w

Re: [asterisk-users] Has anyone tried to install both digital card

> Some one recommended Sangoma E1 card, they said it has less problem for > interrup conflct? Is that true according to your guys' experience? > > -- > Regards! > Liangliang > > -- It is an excellent card, also very good drivers, amazing support from from Sangoma technicians. Go for it. thanks,

[asterisk-users] Attended transfer and parking calls

Can anyone help me with information on how to implement or use the Attended transfer and parking calls?   I have tried the extension 700 getting a number for the parked call but I was never been able to retrieve the call (don't know how) by dialing the indicated extension number.   Also, we nee

[asterisk-users] Has anyone tried to install both digital card and analog card in one machine

Some one recommended Sangoma E1 card, they said it has less problem for interrup conflct? Is that true according to your guys' experience?-- Regards!Liangliang ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSU

[asterisk-users] Bristuffed asterisk 1.2.10 on Suse 10 - problem with module versionmagic

I'm banging my head on compiling bristuff modules for Suse 10.0 with kernel : Linux laps1 2.6.13-15.11-smp #1 SMP Mon Jul 17 09:43:01 UTC 2006 x86_64 x86_64 x86_64 GNU/Linux and Asterisk 1.2.10-BRIstuffed-0.3.0-PRE-1s. I get this : laps1:~/Voipy/Bristuff/bristuff-0.3.0-PRE-1s/zaptel # modpro

Re: [asterisk-users] Asterisk with cisco 7935

Miles Scruggs wrote: Just wondering if anyone has had any luck getting the cisco 7935 working with asterisk and if so, what is the best way to go about it? on the My testing shows it was a wasted purchase. Using CHAN_SCCP I was able to get it to work, but not stably (i.e. keys stopped func

[asterisk-users] app_txfax segv fault

I am getting this error when trying to use app_txfax. Program received signal SIGSEGV, Segmentation fault. [Switching to Thread 1078266208 (LWP 28837)] 0x003873a0b0df in __read_nocancel () from /lib64/tls/libpthread.so.0 (gdb) where #0 0x003873a0b0df in __read_nocancel () from /lib64/tl

Re: [asterisk-users] University switches to Asterisk

I stumbled upon this yesterday while reading my usual news sites, and added it to Digg.com -- so be sure to digg it for even more exposure -- http://digg.com/tech_news/University_Dumps_Cisco_VoIP_Moving_6_000_Students_to_AsteriskThis is a great example for Asterisk, since most folks remain quiet o

[asterisk-users] Voicemail adjustments

Hi all, Some questions about Asterisk Voicemail adjustments I want to make: - how can I limit the number of voicemail messages stored per user in their voicemail folder? (to expire voicemail after a specified number of days I know that there is in /contrib/scripts one script to do that) - ho

Re: [asterisk-users] Cisco GW & CID Name

I'd look into Remote-Party-ID headers to affect the type of call screening you want. I use this for caller ID blocking to/from SER but my carrier doesn't support the name in the caller ID . -Steve Peder @ NetworkOblivion wrote: Does anybody know how to enable CallerID name passing from a Ci

[asterisk-users] Asterisk with cisco 7935

Just wondering if anyone has had any luck getting the cisco 7935 working with asterisk and if so, what is the best way to go about it? on the wiki there is talk about new software images etc, but I'm thinking those are for the 7940 & 60 phones. If someone could point me in the right direction

[asterisk-users] Internal message being heard on pstn line

I am on a sip line, ext 11, checking my voicemail. While I am in the midst of it, an incoming call from a pstn line, dialing 11, caller hears the internal message: "Asterisk mail box,... password" Obviously, the caller is confused. Where should I look in the configuration to avoid this ? Th

[asterisk-users] Cisco GW & CID Name

Does anybody know how to enable CallerID name passing from a Cisco gateway (with PRI that has name and number) to an * box via SIP? Supposedly CID name is enabled, but we can't get it passed to * and I've googled and I can't find what I need. ___ --B

[asterisk-users] Branch office interconnect - IAX :vs: SIP?

Scenario:   Two Astlinux servers, main office/branch office.  Calls come in via PSTN (ZAP) or SIP VoIP provider always at the main office.  Inbound call will ring a number of extensions at main office and one phone located at a branch office site.  Calls are routed to the branch office via

[asterisk-users] [OT] Nokia E60/61/70 and SIP

For all of us using these devices, I have some good news. There is a self installable firmware update available from Nokia here (requires windows box to install): http://www.nokia.co.uk/nokia/0,1522,,00.html?orig=/softwareupdate This seems to radically improve the behavior of the SIP clien

[asterisk-users] inbound call from GSM gateway: handle_request_invite: Failed to authenticate user

Hi all, I am getting a "handle_request_invite: Failed to authenticate user" error when I attempt to receive calls from a GSM gateway (I can successfully call through the device VoIP-GSM from asterisk). I have looked for a solution to this error but most point me to adding a register line which I've

[asterisk-users] ZT_SPANCONFIG failed on span 1: No such device or address (6)

Help me please..   ZT_SPANCONFIG failed on span 1: No such device or address (6)   how can i fixed this problem.   Thank you.   JmiguelY ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update o

RE: [Asterisk-Users] Shared Line Appearance, Snom and trunk

As far as I know there is ZERO documentation and it's still too buggy to even test. I'm pretty sure the Snom will work with it when they do get it further along. -Original Message- From: Olivier [mailto:[EMAIL PROTECTED] Sent: Friday, September 15, 2006 1:19 AM To: Asterisk Users Mailing

Re: [asterisk-users] 4-wire analogue interfaces?

Quoting Tony Mountifield <[EMAIL PROTECTED]>: > Does anyone know of any 4-wire analogue interface cards that could be > made to work with Asterisk? (I'm not averse to hacking channel drivers) A T1 card to a D4 bank with something like a 4WEM or 4WTO should do the trick. --Shane ---

Re: [asterisk-users] Cisco Distinctive ring using alert-info

Rich Adamson wrote: Julian Lyndon-Smith wrote: I've got a cisco 7960, with (amongst many others) the following in the RINGLIST.DAT file Foghorn foghorn.raw I can manually select this for the ringtone. However, I was wanting to use a normal ringtone, with foghorn being used if the cal

[asterisk-users] Re: [asterisk-dev] open letter

[EMAIL PROTECTED] a écrit : Dear asterisk users, the asterisk projects and ser enabled me to learn SIP, I could be insulting sometimes . I must begin my business with communigate and a French company. I consider it regrettable that asterisk and ser could not do it. I do hope that these open

[asterisk-users] 4-wire analogue interfaces?

Hi, Does anyone know of any 4-wire analogue interface cards that could be made to work with Asterisk? (I'm not averse to hacking channel drivers) They would be used to support an always-on form of conferencing. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk P

[asterisk-users] Re: [asterisk-dev] open letter

[EMAIL PROTECTED] a écrit : Dear asterisk users, the asterisk projects and ser enabled me to learn SIP, I could be insulting sometimes . I must begin my business with communigate and a French company. I consider it regrettable that asterisk and ser could not do it. I do hope that these open

Re: [asterisk-users] mISDN versus ZapHFC with BRIstuff

I have 2 single BRI s0 cards. -1 in TE mode for the outside line -1 in NT mode for the inside phones If I dial the group with "Dial(Zap/g2/,60,t)" then all MSN's on all phones ring. But how do I dial so only MSN 10,11,12 rings? If I dial every number as "Dial(Zap/g2/10&Zap/g2/11&Zap/g2/12,60,t

[asterisk-users] [asterisk-dev] open letter

Dear asterisk users, the asterisk projects and ser enabled me to learn SIP, I could be insulting sometimes . I must begin my business with communigate and a French company. I consider it regrettable that asterisk and ser could not do it. I do hope that these open projects will help many people.

Re: [asterisk-users] Cisco Distinctive ring using alert-info

Julian Lyndon-Smith wrote: I've got a cisco 7960, with (amongst many others) the following in the RINGLIST.DAT file Foghorn foghorn.raw I can manually select this for the ringtone. However, I was wanting to use a normal ringtone, with foghorn being used if the call was coming in from

Re: [asterisk-users] Intel 945G and Digium TE110P compatibility issue

cat /proc/zaptel/1are you seeing any IRQ misses?are you also seeing any HDLC errors in your asterisk debug log?Mark.On 9/8/06, Xue Liangliang <[EMAIL PROTECTED]> wrote: Hi, I just installed a TE110P in  a supermicro server with Intel 945Gchipset,  the customer reported  the system has random drop

[asterisk-users] open letter

Dear asterisk users, the asterisk projects and ser enabled me to learn SIP, I could be insulting sometimes . I must begin my business with communigate and a French company. I consider it regrettable that asterisk and ser could not do it. I do hope that these open projects will help many people.

[asterisk-users] Polycom 501 - message waiting LED manipulation

Hi,   I'm hoping to find the answer to this here, because I believe the admin manual doesn't give it.   I'd like to change the led behavior on my Polycom 501 for the message waiting indicator.  Basically, I want to manipulate it to make the led always on.   I can't find the led pattern ref

Re: [asterisk-users] problems with Polycom 500 boot up

Dear Steve,The phone may be looking for it's specific configuration files (not phone1.cfg, but instead 0004Fcfg {or [mac].cfg}). In our past experience, if the phone was ever formatted (fully formatted), the phone will request this from the FTP server specified. Of course confirm your phone

Re: [asterisk-users] How to install HUDLite Server

As for FOP, when clients come to meet you after seeing attractive interfaces from other proprietary systems, its just embarrassing to show them such an ugly interface like FOP. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mai

Re: [asterisk-users] two safe_asterisk processes on the same PBX???

If you want to have a "safe asterisk" I would recommend using "svscan" from daemontools package, more wonderfull software of D.J. Bernstein. http://cr.yp.to/daemontools/svscan.html Regards On 9/15/06, Giorgio Incantalupo <[EMAIL PROTECTED]> wrote: Hi Julian, I know I have two process.the p

RE: [asterisk-users] Sphinx2

What sphinx documentation? All I could find was docs on the code, not on how to USE the software. > -Original Message- > From: Matt Riddell (IT) [mailto:[EMAIL PROTECTED] > Sent: Friday, September 15, 2006 1:18 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re

Re: [asterisk-users] How to install HUDLite Server

HUDLite has a very impressive interface. Works fine for me on Trixbox, but couldn't make it work on standalone Asterisk.   What kinds of bugs are there in it? Has anybody used its full version, how is it. I am thinking of upgrading one of my clients to fonality, just because of HUD's interface and

Re: [asterisk-users] Re: chan_zap.so stopped working after upgrading CentOS

Why zaptel 1.2.5 and not the newer version? ___ --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] Re: Streaming MoH Problem, starts and then stops immediately

CentOS 3.8, Asterisk 1.2.9.1, AMD Sempron(tm) Processor 3000+, mpg123-0.59r ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asteris

[asterisk-users] Issues with AGI+Dial command

Hello, I am trying to write an AGI application that will transfer the caller to a phone number on certain conditions. From what I understand (from the astcc application and voip-info wiki), I should just be able to EXEC the dial command. I'm having problems with this though. I send asterisk the

Re: [asterisk-users] Astmanproxy authentication problems

Fabian, [EMAIL PROTECTED] escreveu: > I've try to use Astmanproxy with "Asterisk TAPI line". > But login fails, astmanproxys error message: > > "Sep 13 20:06:26: [EMAIL PROTECTED] got: Response: Error > Sep 13 20:06:26: [EMAIL PROTECTED] got: Message: No variable specified > Sep 13 20:06:26: [EM

[asterisk-users] where download app_txfax?

I have gone to http://soft-switch.org/downloads/spandsp/ looking to app_rxfax and I dont see it? Where is it? Parent Directory - [TXT] app_dtmftotext.c 17-Mar-2004

Re: [Asterisk-Users] Can you explain why multiple registration is an important (missing) feature ?

Christian Mohrbacher wrote: In some cases : Yes. But we have the following situation : We re using cisco 7960 phones in each office (about 150 of them), but not every person has it's own phone. Normally there are two employees in one office and they share one phone, BUT have their own extension.

[asterisk-users] Section '12345678' lacks type

I'm testing the use of static SQL-config. Everything seems to work OK, exept these warnings: Sep 15 15:06:20 WARNING[25374]: chan_sip.c:12829 reload_config: Section '157217030' lacks typeSep 15 15:06:20 WARNING[25374]: chan_sip.c:12829 reload_config: Section 'TISP-157217030' lacks typeSep 15

Re: [asterisk-users] Modem calls

I need to pass modem calls through a TDM400 card. Conecting the modem to the FXS port (ZAP/1), it should be put through the FXO port (ZAP/4) directly. According to Digium, Fax calls (and modem calls) are not supported on the TDM400 or TDM2400. They are designed for voice only. If you get it to w

Re: [asterisk-users] RE: Asterisk 1.4 Docs

One of the providers that I use already offers this feature via a macro in the dail plan http://connect.voicepulse.com/FlexRate.aspx -Jason On Fri, 2006-09-15 at 10:21 +0200, Tomislav Parčina wrote: > In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says... > > No

RE: [asterisk-users] CDR question with SIP/IAX trunks

Ok... I got it. Someone changed the CDRs to reflect CALLERID(ANI) instead of CALLERID(number) in 1.2.10. According to the release notes this was taken back in 1.2.12. I do not know why this was not done for IAX as well so it would have been consistent at least but well... I am either going to

Re: [asterisk-users] QuadBRI and Zyxel Wifi phone stop working togetherafter 3 calls

Just tried it. When I run "sip show channels" it doesnt show any open channels. Thanks, Frederik On 14 Sep 2006, at 03:27, Bill Gibbs wrote: Make those calls then check the CLI "sip show channels" and see if the channels are stay up -Original Message- From: [EMAIL PROTECTED] [mailto

Re: [asterisk-users] 9 becomes 99 ? And other strangeness

On Friday 15 September 2006 04:20, Brian Candler wrote: > it worse: I got a failure rate of about 50%, and in one case 66611 instead > of 611. It's clear your system is possessed. Please contact your local clergyman for help with these issues. -A. (seriously though, I've had this particular pr

Re: [asterisk-users] non-technical, dealing with users giving feedback

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 stoffell wrote: >> Search Daily Asterisk News for echo: > > Yes, that's for the issue with echo, but I was more or less meaning > the social side, the communication with the users.. echo was an > example.. :) (bad choice maybe? :)) :) I kinda knew t

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