Re: [Asterisk-Users] Installing the none commercialintelg729codecsinto [EMAIL PROTECTED] 2.2?

ram wrote: Hi as per the list people guidence i have downloaded the Codec and installe my Pc is P4, but i have downloaded the P2.so file and copied in specific directory whe i see show translation i could able to see 30 i have configure AAH for VOIP JET connection when i try to make

Re: [Asterisk-Users] DeadAGI and Hangup on channel

Hello, GP> I'm trying to catch channel hangup in DeadAgi script. Googling didn't help. Channel status AGI command returns 6 - "line is up", because hangup has been requested only (and not completed). I found the following solution. In res_agi.c source I found usage of ast_check_hangup(chan) func

RE: [Asterisk-Users] HandyTone 488 ata?

On Mon, 2006-01-30 at 23:46 +, Chris Bagnall wrote: > Has anyone ever gotten that working? I've tried it on every Granstream > device I've had (budgetone, HT486, GXP-2000) and it's far from reliable on > any of them. Seems that when dialling an external number, the phone accepts > the first 3 d

Re: [Asterisk-Users] Re: [asterisk-biz] iDEFISK (mac iax2 softphone) release

On Tue, 2006-01-31 at 14:33 +0800, Dinesh Nair wrote: > > On 01/31/06 14:24 Tzafrir Cohen said the following: > > 1. FreeBSD can run Linux binaries, IIRC. Have you tried the Linux > > version? > > nothing beats a native version, no ? FBSD linux stuff isnt really that different from native. Ther

Re: [Asterisk-Users] Installing the none commercialintelg729codecsinto [EMAIL PROTECTED] 2.2?

Hi   as per the list people guidence i have downloaded the Codec and installe   my Pc is P4, but i have downloaded the P2.so file and copied in specific directory   whe i see show translation i could able to see 30   i have configure AAH for VOIP JET connection   when i try to make call out, its us

Re: [Asterisk-Users] Set caller id on Swedish PRI (euroisdn)

On Mon, Jan 30, 2006 at 03:43:18PM +0100, [EMAIL PROTECTED] wrote: > Hi, > > I have a problem with setting outgoing caller id to "nothing" (secret) > on our Wildcard TE405P connected to a swedish euroisdn line. Caller ID > seems to work fine when connecting the same line to a Ericsson PBX - so >

[Asterisk-Users] Polycom IP301: Pass-through ethernet port unusable?

Have just done a deployment of 45 of these puppies. They are doing their main job quite well, but of course there are minor kinks. A not-so-minor one is that if one attempts to plug a PC into the 2nd RJ-45 jack, as soon as you send any reasonable amount of traffic (even casual web surfing) the

Re: [Asterisk-Users] Re: [asterisk-biz] iDEFISK (mac iax2 softphone) release

On 01/31/06 14:24 Tzafrir Cohen said the following: 1. FreeBSD can run Linux binaries, IIRC. Have you tried the Linux version? nothing beats a native version, no ? 2. stick to free software ;-) i'll ignore this in the interest of avoiding silly my license is better than yours type thread

RE: [Asterisk-Users] dialing 2 channels at the same timewithdifferentcaller ID number?

> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Damon Estep > Sent: Monday, January 30, 2006 11:20 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] dialing 2 channels at the same > time

Re: [Asterisk-Users] Re: [asterisk-biz] iDEFISK (mac iax2 softphone) release

On Mon, Jan 30, 2006 at 04:11:12PM +0800, Dinesh Nair wrote: > > is there any chance of a FreeBSD port of idefisk ? more often than not, > most of us in *BSD-land get left out in things like this. we'd be willing > to help in the port wherever we can. 1. FreeBSD can run Linux binaries, IIRC. Ha

RE: [Asterisk-Users] dialing 2 channels at the same timewithdifferent caller ID number?

>Exten => _NXXNXX,2,Set(__ORIGCID=CALLERID(number)) >exten => _NXXNXX,2,dial(sip/${EXTEN}&local/[EMAIL PROTECTED]/n,r} > >[alternate1] > >exten => _NXXNXX,1,macro(alternate-number|${__ORIGCID}) > >[macro-alternate-number] > >exten => s,1,set(CALLERID(number)=${ARG1}) >exten => s,2,dial(

Re: [Asterisk-Users] Grandstream Budgetone BT-101 audio problems

Are you using the same NTP server for both phones? Are you using NTP at all? Is jitterbuffer enabled on the asterisk server? Not sure about SIP, but on IAX if the timestamps go haywire, you can loose audio from one side. hth, c On 1/31/06, jurgen <[EMAIL PROTECTED]> wrote: > Hi all, > > I'm h

RE: [Asterisk-Users] dialing 2 channels at the same time withdifferent caller ID number?

> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Damon Estep > Sent: Tuesday, January 31, 2006 12:38 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] dialing 2 channels at the same > time withdiff

[Asterisk-Users] Grandstream Budgetone BT-101 audio problems

Hi all, I'm having a really frustrating time with a bunch of BT-101 phones. They've been trouble-free and working very well for the past several months. A couple of days ago, some of the phones (but not all of them, yet) have started acting very strangely. All phones are running firmware 1.0.6.7,

RE: [Asterisk-Users] dialing 2 channels at the same time with different caller ID number?

> > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of > > Damon Estep > > Sent: Sunday, January 29, 2006 11:09 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: [Asterisk-Users] dialing 2 channels at the same time > > wit

[Asterisk-Users] Re: [asterisk-biz] iDEFISK (mac iax2 softphone) release

On 01/21/06 02:02 Zoa said the following: ] Hey ho, A few days ago we released the linux version of the phone, today we are very happy to have the mac version ready for a little field test. Freely downloadable from http://www.asteriskguru.com/tools/idefisk_mac.php is there any chance of a

[Asterisk-Users] Meetmee weirdness

I have several instances where conference calls are not being torn down appropriately.  My CDR shows 3000 minute calls, which are coming in on PRI.  I know that the calls aren't really  lasting that long.  What could be causing this?  IN fact, here is what shows now:     asterisk*CLI> meetme

RE: [Asterisk-Users] AAH out bound routing problem

Ram-   You  are three steps ahead of where you need to be.  You need to figure out what to send before you figure out how to send it.   Add a test extension in your extensions_custom.conf:     exten => 3852,1,Dial(sip/easycall/19197543700,30)   dial 3852 and aee if it works.  If not, tr

[Asterisk-Users] RE: [Asterisk-Announce] Asterisk 1.2.4 and Zaptel 1.2.3

Does anyone know what date this memory leak was introduced and/or how to check source code for it? I am running a pre-1.2 CVS head version and would like to know if the potential problem exists. > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk- > [EMAIL PROTECTED] On Behalf

[Asterisk-Users] Asterisk 1.2.4 and Zaptel 1.2.3

Asterisk 1.2.4 and Zaptel 1.2.3 have been released! This update of Asterisk includes a fix for a significant memory leak in the expression parser that is present in all previous releases of Asterisk 1.2. This version of Zaptel includes support for the new generation of VPM100M echo cancellation mo

RE: [Asterisk-Users] Urgent: Unable To Execute after updating from SVN

Delete /usr/lib/asterisk/modules/app_md5.so and update your dialplan if you use MD5. It is now done in functions. /usr/lib/asterisk/modules/app_md5.so is a leftover from your previous installation.  [app_md5.so]Jan 29 02:49:10 WARNING[32424]: loader.c:326 __load_resource: /usr/lib/asterisk/mo

[Asterisk-Users] cdrtool

anyone having weird problems on latest cdrtool? #!/usr/bin/php4 Fatal error: Class webservice_ngnprocdrtool_ngnprocdrtool: Cannot inherit from undefined class soap_client in /var/www/CDRTool/SOAP/client_lib.php on line 2 always get weird error like that ___

RE: [Asterisk-Users] Asterisk SIP phones to Cisco Unity via CCM4.0SIP Trunk

It can be done.   1.  Setup a new Vm profile on CCM with a mask of 2.  Setup a CTI route point:     a. Set the directory number to a pattern.  I use *27XX     but any pattern that you can send from * is good, ie. 88XXX     b.  Set the VM profile to the newly created profile     c.  Set

Re: [Asterisk-Users] adress book

This all depends on what your existing setup is, mainly where you are storing your sip users in the first place. No point duplicating your user list. The Ciscos consume XML, so just parse out the list of users via some scripting language from the DB/directory/flat-files or whatever you're usi

[Asterisk-Users] TDM400P FXO port problem.

I have a digium TDM400P with 1 FXO and 1 FXS port. I have a standard analog phone connected to the FXO port and place calls to the PTSN phone line. The analog trunk is accessed via the standard "9" and area code (if needed) and of course the phone number. The error is as follows. I dial 9,866-XXX-X

Re: [Asterisk-Users] Asterisk Evening in Melbourne: Feb 2!

On 31/01/06, Skeeve Stevens <[EMAIL PROTECTED]> wrote: > Any idea if there will be a Sydney one? There was some talk about Sydney doing something, but as far as I know, no one has done anything about it yet. There's a page in the wiki about Sydney, with some contact information of the person who'

Re: [Asterisk-Users] polycom ip601 attendant console

Damon Estep wrote: Anyone successfully set up one of the polycom soundpoint ip sidecars with asterisk to monitor and allow transfer to monitored extensions? How does it work? Any issues? ___ --Bandwidth and Colocation provided by Easynews.com -- Aste

[Asterisk-Users] polycom ip601 attendant console

Anyone successfully set up one of the polycom soundpoint ip sidecars with asterisk to monitor and allow transfer to monitored extensions? How does it work? Any issues? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing lis

Re: [Asterisk-Users] Asterisk Evening in Melbourne: Feb 2!

Sounds great - be there and be square. PaulH > jurgen <[EMAIL PROTECTED]> wrote: > > Hi all, > > Come one come all! We're having the next Asterisk evening at the > Fujitsu Centre for Excellence! This is Fuji's state of the art > show-off centre - they're promising lots of interesting toys to

RE: [Asterisk-Users] HandyTone 488 ata?

> The only significant feature that the SPAs seems to be > missing compared to the HTs is the "Early Dial" thing (where > it sends each digit to Asterisk until it gets something other > than a 484 response back). Has anyone ever gotten that working? I've tried it on every Granstream device I've

RE: [Asterisk-Users] HandyTone 488 ata?

> I played with FXO on the HT488 a bit, but didn't have a whole > lot of luck. We had a bit of a problem with echo, but more > seriously the thing kept getting itself into a variety of > wedged states: sometimes it would lock up altogether (usually > with its button lit up), and sometimes it w

[Asterisk-Users] Grandstream Budgetone mass deployment?

On Mon, 30 Jan 2006, Phil Blundell wrote: / On Mon, 2006-01-30 at 11:51 -0800, asterisk at anime.net wrote: />>>/ does your python script generate the binary format grandstream files or do />>>/ you still need to use their closed-source

Re: [Asterisk-Users] Grandstream Budgetone mass deployment?

On Mon, 30 Jan 2006, trixter aka Bret McDanel wrote: I looked and it doesn't seem anyone has cracked the checksum yet. Depending on what you want, there is a perl script called 'gsutil' that will configure granbdstream stuff without using tftp or the web interface. It doesnt create cfg files, b

RE: [Asterisk-Users] Asterisk Evening in Melbourne: Feb 2!

Any idea if there will be a Sydney one? ...Skeeve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jurgen Sent: Tuesday, 31 January 2006 9:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Commercial and Business-Oriented Asterisk Discuss

[Asterisk-Users] mISDN errors on asterisk CLI

Hi there guys, Does anyone know what this is?? Every time a mISDN channel connects to anything, I get this message on the CLI of asterisk. Unhandled Message: prim 281 len -22 from addr 51400101, dinfo 0 on port: 1 Thanks Pedro Nunes ___ --Bandwidt

RE: [Asterisk-Users] Cisco 7940 not reading SIP image file

Do you have the file in the SEP.cnf.xml file? i.e. P0S3-05-1-00 Make sure you have SIP.cnf, RINGLIST.DAT, and dialplan.xml too. Thanks, Greg > -Original Message- > From: Jerry Geis [mailto:[EMAIL PROTECTED] > Sent: Monday, January 30, 2006 12:57 PM > To: asterisk-users@lists.digium.

[Asterisk-Users] Asterisk Evening in Melbourne: Feb 2!

Hi all, Come one come all! We're having the next Asterisk evening at the Fujitsu Centre for Excellence! This is Fuji's state of the art show-off centre - they're promising lots of interesting toys to play with. As usual, we'll be discussing developments in Asterisk land over the past couple of mon

Re: [Asterisk-Users] Connecting the two servers

satish Ahalawat wrote: Hi All, I want to setup the interconnectionm between two servers, both having sip clients behind firewalls. I want the calls from any of the servers to land on any of SIP clients on the other. I am looking for dial out plans with the sample configuration files . Thanks

Re: [Asterisk-Users] Live CD?

Thczv F. Thczv wrote: I would love to run Asterisk on an old laptop, in a mostly solid state configuration, with no HD. The laptop is slow (Pentium 233), and I need PCMCIA support (for my network card). Are any of you aware of a live CD that might work? Thanks, Dave Take your pick: ht

RE: [Asterisk-Users] How many TDM2400P's will a server take?

> Juan Carlos Castro y Castro wrote: > > How many TDM2400P cards can I safelly install in one PC? I'm loking for > > answers from whoever has a working scenario with * and a number of cards > > higher than one. > > > Depends on the specs of the server. For example, a quad Xeon will be > able to s

Re: [Asterisk-Users] Grandstream Budgetone mass deployment?

On Mon, 2006-01-30 at 13:33 -0800, [EMAIL PROTECTED] wrote: > On Mon, 30 Jan 2006, Phil Blundell wrote: > > On Mon, 2006-01-30 at 11:51 -0800, [EMAIL PROTECTED] wrote: > >> does your python script generate the binary format grandstream files or do > >> you still need to use their closed-source tool

Re: [Asterisk-Users] Grandstream Budgetone mass deployment?

On Mon, Jan 30, 2006 at 03:14:02PM +0200, Dmitry Ivanov wrote: > I have created dynamic "CGI-like" TFTP server so I will create config > files on-the-fly. Now we use this system (dynamic tftp server and Perl > "CGI" script) for country-wide Sipura 3000 configuration. BTW, if > anyone is interes

Re: [Asterisk-Users] Cisco 7940 not reading SIP image file

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Jerry Geis wrote: > I have a 7940 trying to connect to an existing running system. > tftp is configured and running normal. > > (NOTE: I know there is a later SIP version but this is the one I have) > > I see the phone bootup and ask for OS79XX.TXT w

RE: [Asterisk-Users] Web interface

I don't think I received the whole thread, but I just wanted to mention that the language selection has been added to the preferences page of PBX Manager. As Stefan mentioned, it is normally done in Webmin, but since some users may not be allowed to change anything outside of the module we added it

Re: [Asterisk-Users] Grandstream Budgetone mass deployment?

On Mon, 30 Jan 2006, Phil Blundell wrote: On Mon, 2006-01-30 at 11:51 -0800, [EMAIL PROTECTED] wrote: does your python script generate the binary format grandstream files or do you still need to use their closed-source tool? Right now I'm still using their Java thing, but it's slow enough that

[Asterisk-Users] Help configuring Asterisk server

I need to configure / migrate Asterisk server from 0.9 to the latest version with some upgrades. Please help! Thank you. Sincerely, Naren Koka (480) 829-0479 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To

[Asterisk-Users] sip domain

hi, where can i change the sip domain?, i dont see any reference in the docs, thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/lis

[Asterisk-Users] load balancing

Hi list, I was wondering i anybody ever tried to use asterisk on an openmosix loadbalancing cluster. Obviously, hw-related processes can not migrate from one system to another, but any other pricess else should be able. Or not? Hans -- pgp-id: 926EBB12 pgp-fingerprint: BE97 1CBF FAC4 236C 4A73

Re: [Asterisk-Users] Re: Cant compile asterisk #error "You need newer libpri"

Thanks Tony! You are (of course) absolutely correct. I feel like an idiot for doing that when I know better. Take care Steve > In article > <[EMAIL PROTECTED]>, > Steve Gladden <[EMAIL PROTECTED]> wrote: >> >> I am starting over and now trying to compile/install /trunk >> zaptel >> libpri >>

Re: [Asterisk-Users] Grandstream Budgetone mass deployment?

On Mon, 2006-01-30 at 11:51 -0800, [EMAIL PROTECTED] wrote: > does your python script generate the binary format grandstream files or do > you still need to use their closed-source tool? Right now I'm still using their Java thing, but it's slow enough that one of these days I guess I'll crack and

[Asterisk-Users] Dlink DVG-3004S ?

Anyone have one of these yet? http://www.dlink.com/products/?pid=451 -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asteris

Re: [Asterisk-Users] Asterisk 1.2.3 CentOS 4.x RPMS

Remco Barende <[EMAIL PROTECTED]> wrote: > Hi! > > I'm trying to install the RPMS, in the installation document the > following module is not mentioned: > perl-DBD-mysql-3.0002-1.RHEL4.LSE.i386.rpm > But the RPM is in the CentOS 4 directory. > > On CentOS 4 the rpm is even already present albeit

Re: [Asterisk-Users] help with iaxy - one way sound

All the phones, the iaxy and the server are on a 192.168.3.* network and the only outside interface currently is the x100p card. I do have an account with a voip provider but I haven't got that setup up yet since rebuilding the server on Sunday. So there shouldn't be any routing issues coming

RE: [Asterisk-Users] help with iaxy - one way sound

At first glance it sounds like a routing issue. Are you IAX to your phones, but SIP to your tisp provider? Any change on your asterisk box / firewall / ISP / TISP since then? My first guess is that when you replaced the card you changed a network / iptables setting etc. on your asterisk box. T

Re: [Asterisk-Users] Grandstream Budgetone mass deployment?

On Mon, 30 Jan 2006, Phil Blundell wrote: Budgetones as they do on the Handytones and the GXP-2000. Obviously you need some way to make the files in the first place: when we deployed our GXP-2000s I ended up writing a little Python script to create the Grandstream config files (and the associate

Re: [Asterisk-Users] Grandstream Budgetone mass deployment?

On Mon, 30 Jan 2006, Dmitry Ivanov wrote: I have created dynamic "CGI-like" TFTP server so I will create config files on-the-fly. Now we use this system (dynamic tftp server and Perl "CGI" script) for country-wide Sipura 3000 configuration. BTW, if anyone is interested I can send sources of this

[Asterisk-Users] re: help with redirect from SER

hello all,  i have a problem, and i'm tearing my hair out...any assistance is appreciated. I am trying to redirect from SER to Asterisk, both on the same machine. In 1.09 I didnt need to set up a peer for SER, just autocreatepeer=yes, and rewritehostport from SER as below, and asterisk accepted the

Re: [Asterisk-Users] Web interface

Hello Zac, > There is 1 problem.. I only took 1 semester of German 15 years ago. > Looked all over the page for the English button, but I could not find one. > I did wake up 10 minutes ago, so I could still be blind. > the language of the module is influenced by the language you choosed for webm

[Asterisk-Users] Codec preference selection?

Hi;   I'm trying to implement what is known by Cisco Callmanager as regions: Specify that when phones from  zone A call to phones in zone B, use g729, but if they call to zone C, use g711. Any ideas on how to achieve this?   Thanks! Francisco Sedano   _

[Asterisk-Users] help with iaxy - one way sound

I have an iaxy the is across a 802.11b link from my asterisk server. signal strength is good and it has been working fine there for about a year. Friday night lightning took out the x100p card in my asterisk server and I just got it all working again last night. Lucky I had a spare. Since I

[Asterisk-Users] Analog with channel bank - Inbound works, outbound doesn't

I am experimenting with an asterisk setup in my office. The last bit I have to test is working with analog lines. I have TE411p digium card, with an ISDN line plugged into the first, a channel bank plugged into the second port, and the last two ports empty. I have the following setup in my za

RE: [Asterisk-Users] Gateways

Hi, I didn't have this gateway, But on welltech 4fxo gateway. You can just dial SIP/[EMAIL PROTECTED] Even the gateway didn't register to the asterisk server. Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Corne Vermeulen Sent: Monday, January 30

Re: [Asterisk-Users] Grandstream Budgetone mass deployment?

I have 3 Grandstream Budge-Tone 100 with Firmware 1.0.7.11beta, and they are working very well since more then 4 month now. Guenther Davao City, Philippines, Planet Earth, 32.1 °C Phil Blundell wrote: On Mon, 2006-01-30 at 15:14 +0200, Dmitry Ivanov wrote: On Monday 30 January 2006 13

RE : [Asterisk-Users] How many TDM2400P's will a server take?

A PICMG card equiped with Pentium M CPU permits to reduce dragsticaly the power consumption. As this, you can use more power from your PSU for the interface cards. But, for several TDM2460E/B cards with a heavy traffic charge (many simultaneous rings), I believe that it could be better to use a se

[Asterisk-Users] Re: How many TDM2400P's will a server take?

Ah -- for all intents and purposes, assume I can obtain the most kickass PC server hardware in the known Universe. So -- any real-life experiences out there? > How many TDM2400P cards can I safelly install in one PC? I'm loking for > answers from whoever has a working scenario with * and a number

Re: [Asterisk-Users] Kirk IP600

Hi! Yes, it works (sort of) but I still have some issues. When using more than 2 handsets some of them do not always ring on an incoming call. This might be because I use only 2 Kirk handsets and the rest are Siemens, maybe it's the driver I created a howto for it, you can find it here: http

[Asterisk-Users] Cisco 7940 not reading SIP image file

I have a 7940 trying to connect to an existing running system. tftp is configured and running normal. (NOTE: I know there is a later SIP version but this is the one I have) I see the phone bootup and ask for OS79XX.TXT which has [EMAIL PROTECTED] src]# cat /tftpboot/OS79XX.TXT P0S3-05-1-00 Jan

Re: [Asterisk-Users] Asterisk 1.2.3 CentOS 4.x RPMS

Hi! I'm trying to install the RPMS, in the installation document the following module is not mentioned: perl-DBD-mysql-3.0002-1.RHEL4.LSE.i386.rpm But the RPM is in the CentOS 4 directory. On CentOS 4 the rpm is even already present albeit an older version: [EMAIL PROTECTED] rpms]# rpm -qa | g

Re: [Asterisk-Users] How many TDM2400P's will a server take?

And you should consider how many FXS's you're running. More than one card with all FXS's will require a turbo fan to cool and if they all ring you'll need a decent power supply to handle the power draw.Rob On 1/30/06, Steven Ringwald <[EMAIL PROTECTED]> wrote: Juan Carlos Castro y Castro wrote:> Ho

[Asterisk-Users] Re: [asterisk-dev] SIP domain support for authentication and virtualhosting

Barry Flanagan wrote: Hi, Can someone explain to me or point me in the direction of documentation for the domain support feature in 1.2.x? Specifically I need to be able to have sip users who are authenticated as [EMAIL PROTECTED] For authentication, we only look at 1) Whether the domain is

[Asterisk-Users] Kirk IP600

Hi all, has anyone tryied to configure asterisk with Kirk IP600 Dect-IP gateway? Could it works using the skinny channel ?   Thanks   Giordano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIB

[Asterisk-Users] About Extensions

Im trying to detect before entering in Meetme , which dtmf has been entered. I did a Background(file) and go to a context where i define a exten => _X.,1,Meetme() I have detected that with (1.2.1) when 1 is entered and conference 1 must be created, extensions say it is not possible and ga

[Asterisk-Users] Re: [asterisk-dev] SIP domain support for authentication and virtual hosting

Barry Flanagan wrote: Hi, Can someone explain to me or point me in the direction of documentation for the domain support feature in 1.2.x? Specifically I need to be able to have sip users who are authenticated as [EMAIL PROTECTED] For authentication, we only look at 1) Whether the domain is

RE: [Asterisk-Users] DID over analog?

> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Ken D'Ambrosio > Sent: Monday, January 30, 2006 9:22 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] DID over analog? > > I've some DID's that I'm using for in-bou

[Asterisk-Users] Connecting the two servers

Hi All, I want to setup the interconnectionm between two servers, both having sip clients behind firewalls. I want the calls from any of the servers to land on any of SIP clients on the other. I am looking for dial out plans with the sample configuration files . Thanks,. satish _

Re: [Asterisk-Users] DID over analog?

Michael Collins wrote: Analog DID's are a bit backwards compared to normal POTS lines If you point a DID number at a regular POTS line then when a call rings in it simply rings in like a regular analog phone line. Unless the carrier can provide some form of DNIS on an analog line I belie

[Asterisk-Users] Re: Cant compile asterisk #error "You need newer libpri"

In article <[EMAIL PROTECTED]>, Steve Gladden <[EMAIL PROTECTED]> wrote: > > I am starting over and now trying to compile/install /trunk > zaptel > libpri > asterisk > > following the instructions to grab the source trees: > > # svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk > #

RE: [Asterisk-Users] Most Popular FREE SoftPhone for Windows

I prefer IDEFISK. -Kerry   From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave MorrowSent: Monday, January 30, 2006 8:35 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Most Popular FREE SoftPhone for Windows Hi all

[Asterisk-Users] Caller Holds (how to ignore "Drop Call" events from callee?)

Hi, In Brazilian POTS, the caller holds the callee line hijacked - the call is not droped until de caller hungs up. I really don't know the right english term to describe this behavior. My problem is that its most standard here, and there are a number of hacks that count on this. For example

RE: [Asterisk-Users] DID over analog?

Ken, Analog DID's are a bit backwards compared to normal POTS lines. I don't know about outside the US, but here (in California, specifically) I've done a few analog DID installs on some NEC PBX equipment. The trick is that with an analog DID line, the CPE provides the battery to the telco (i.e.

Re: [Asterisk-Users] DID over analog?

> I've some DID's that I'm using for in-bound faxing, but I'm having some > trouble with getting that working perfectly on my T1. So I'm thinking of > pointing them to an analog line. Will the DID's simply come in over the > analog, presumably sending the DID digits via DTMF? Or is that not > s

[Asterisk-Users] Live CD?

I would love to run Asterisk on an old laptop, in a mostly solid state configuration, with no HD. The laptop is slow (Pentium 233), and I need PCMCIA support (for my network card). Are any of you aware of a live CD that might work? Thanks, Dave ___ -

[Asterisk-Users] SIP domain support for authentication and virtual hosting

Hi, Can someone explain to me or point me in the direction of documentation for the domain support feature in 1.2.x? Specifically I need to be able to have sip users who are authenticated as [EMAIL PROTECTED] Thanks.. -- -Barry Flanagan ___ --Bandw

Re: [Asterisk-Users] How many TDM2400P's will a server take?

Juan Carlos Castro y Castro wrote: How many TDM2400P cards can I safelly install in one PC? I'm loking for answers from whoever has a working scenario with * and a number of cards higher than one. Depends on the specs of the server. For example, a quad Xeon will be able to service many more i

RE: [Asterisk-Users] Set caller id on Swedish PRI (euroisdn)

Your settings are fine. Debug PRI to make sure SETUP message is OK (which probably is) and then check with you PRI provider that callerid is enabled on that PRI - E1. /Vel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, Jan

RE: [Asterisk-Users] SER redirect

Check http://www.voip-info.org/wiki/view/Asterisk+at+large Or sipedu http://mit.edu/sip/sip.edu/ Plenty of examples /Vel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sharon Sent: Friday, January 27, 2006 4:41 PM To: Asterisk Users Mailing List - N

RE: [Asterisk-Users] Re: Polycom 501 horrible echo

> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > Upgrading to the correct sip.cfg fixed the problem. The Polycoms are > back to their great speakerphone-ness. A gotcha is that the new > sip.cfg now contains ntp settings. You'll need to modify these to fit > your time

Re: [Asterisk-Users] Need to recompile * after changing zap echo method?

Hi there, Don't think you have to recompile * if you've already compiled * with zaptel before. (chan_zap.so exists) Should just have to rebuild zaptel, install the module, and do a ztcfg Good luck, - Original Message - From: "Brent Torrenga" <[EMAIL PROTECTED]> To: Sent: Monday, Ja

RE : [Asterisk-Users] How many digium cards per server ?

Hi Harry, How many IRQ do you have ? Be carefull for power supply is it is several TDM2460E (all FXS ports) ! It is better to use a seconf power supply... Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de [EMAIL

Re: [Asterisk-Users] Re: Polycom 501 horrible echo

> > One thing I was pondering: you are not, by chance, using the same > > sip.cfg between version 1.4.1 and version 1.6.2 are you? The file has > > changed significantly between these versions, and certain acoustic > > settings that worked with 1.4.1 may not work with 1.6.2 (Not to mention > > tha

[Asterisk-Users] How many TDM2400P's will a server take?

How many TDM2400P cards can I safelly install in one PC? I'm loking for answers from whoever has a working scenario with * and a number of cards higher than one. Thx, Juan __ Mensagem enviada usando o Webmail da ViaLink ver. 2.7.8

[Asterisk-Users] Cant compile asterisk #error "You need newer libpri"

Trying to compile asterisk (again) from scratch. I seem to be still experiencing the effects fro Jan 25 where I get no sip to sip audio. I have tried upgrading to 1.2.3 which has made no change in the problem. I am starting over and now trying to compile/install /trunk zaptel libpri asterisk fol

[Asterisk-Users] Need to recompile * after changing zap echo method?

Dearest List, I guess I missed this point: Is it true that if you change the echo canceler in zconfig.h, and then recompile/install your zap modules, that for this to be taken into effect by * you must then recompile/install *? I would have figured that the zap echo cancellation method was indepe

[Asterisk-Users] Most Popular FREE SoftPhone for Windows

Hi all.  I am trying to find out what the most popular soft phone for Windows is for use with Asterisk. SIP or IAX?   David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com   Tel: (519) 963-3020 Fax: (519) 451-6615   < Lead, follow or

RE: [Asterisk-Users] Asterisk and LCS ?

> Also there is SER or SIP Express Router from > iptel.org, does this do what I need and how do I do it ? Yes. Converting from TCP-to-UDP is simple; in ser.cfg put: # Forward to Asterisk if (method == "INVITE") { if (uri =~ "sip:[EMAIL PROTECTED]") { log(1, "Forw

[Asterisk-Users] DID over analog?

I've some DID's that I'm using for in-bound faxing, but I'm having some trouble with getting that working perfectly on my T1. So I'm thinking of pointing them to an analog line. Will the DID's simply come in over the analog, presumably sending the DID digits via DTMF? Or is that not something th

[Asterisk-Users] Question on SIP Domains and registration

Hello, I have a situation where I need to differentiate between registrations by users where there might be clashes on the left hand side (username) portion of the SIP From URI. (for a multi-domain virtual hosting system) It seems that only the username portion is used for SIP authentication,

RE: [Asterisk-Users] How many digium cards per server ?

How many free intreuppts do you have, its hard to get more than two to work in most systems. How many PSTN lines are you trying to support? > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > [EMAIL PROTECTED] > Sent: Monday, January 30, 2006 1:00 AM

RE: [Asterisk-Users] Grandstream Budgetone mass deployment?

I had a problem with the scripts you can bulk generate, they are linked to the MAC address you initially put in, so if the phone packs in you can't just rename the file. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Phil Blundell Sent: 30 January 20

[Asterisk-Users] Unable to do anonymous outbound calling

Hi,   I'm wanted to do working anonymous calling with my sip  provider.   To do it, I use SetCallerPres(prohib).   The problem:   The "fromuser=" parameter overide the value of "CallerID(number)" and do it don't working.   Anyone had an idea?   Tank's   Loic Foucault __

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