Re: [Asterisk-Users] Eicon Diva Server 4BRI

Has anyone heard of success using the EICON DIVA Server 4 BRI with chan_capi as a PSTN interface with ISDN/BRI and * ? I'm currently installing such a setup. Up till now I did not have any trouble - but I'm still learning to manage *. At first you should make shure that the CAPI is working correc

Re: [Asterisk-Users] How change default law for T100P

On Wed, 3 Nov 2004, Michael Loftis wrote: > --On Wednesday, November 03, 2004 15:40 -0700 Manuel Marin > <[EMAIL PROTECTED]> wrote: > > > I would like to know if there is a way to change default ulaw for a T1 > > card. I see in the zap show channel X that is working as ulaw. How do I > > change

[Asterisk-Users] asterisk as sip proxy registrar

Hello ALl, I have Asterisk up and running. Now I want to set it up as sip proxy registrar. I have few machines with xlite and linphone sip UA's. How do i register these UA's in asterisk ? My Asterisk server IP is 10.0.0.2 and my sip UA's IP addresses are 10.0.0.3,10.0.0.4,and 10.0.0.5 W

Re: [Asterisk-Users] MusicOnhold on Bridged calls "plain text"

Setting the right option in the Dial() Option section may solve your problem. Check out http://ww.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Dial Paulo Adriano wrote: Now that my bridged calls are working fine with ISDN I have a question ? When my customers call in and my ext is not avail

[Asterisk-Users] MusicOnhold on Bridged calls "plain text"

Now that my bridged calls are working fine with ISDN I have a question ? When my customers call in and my ext is not available the call is routed out to my mobile. Everything works but I would like to know if there is a way of having the calling sign (tone) always on . With my current config t

Re: [Asterisk-Users] Sip clients not longer registering

The REGISTER requests that your SIP UAs are sending as listed are not requests to *register*, but request to *unregister* The contacts are '*' and expirations are '0' Granted that Asterisk doesn't do registrations correctly, but it does need a proper registration request with a contact and an E

Re: [Asterisk-Users] Segmentation fault res_features.so

On Thu, 2004-11-04 at 04:57 +0100, Serge wrote: > Have anyone some idea ? > Asterisk - latest cvs, > RedHat9 > == > Asterisk Dynamic Loader Starting: > == Parsing '/etc/asterisk/modules.conf': Found > [chan_modem.so] => (Generic Voice Modem Driver) >

[Asterisk-Users] Little help here...

Hello Everybody! I live in Queens, NY and I was wondering if there is anyone from around here who sells or have an extra Asterisk startup-kit.I can pay the price on the website. I dont have a credit card to get it from online. basically all I need is the FXO card i guess. I already have a linux box

Re: [Asterisk-Users] G.729 on YDL and MacOSX

Benjamin on Asterisk Mailing Lists wrote: On Wed, 3 Nov 2004 11:57:44 -0600, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: Ok, check out ftp.digium.com/pub/asterisk/g729/unsupported/linuxppc Knock yourself out (oh, and look at the date on that file too :-) ) date? "The requested URL /pub/asterisk/g

Re: [Asterisk-Users] How change default law for T100P

Hello Everybody! I live in Queens, NY and I was wondering if there is anyone from around here who sells or have an extra Asterisk startup-kit.I can pay the price on the website. I dont have a credit card to get it from online. basically all I need is the FXO card i guess. I already have a linux box

Re: [Asterisk-Users] Voicemail Mailbox Configuration

hello, there are two options to do this: 1. if you retrieve your voicemails via your phone you will played back some option like changing your busy and unavailable message (after dialing 0 - just follow the instructions). or 2. you just record your soundfiles with your favourite recorder and ch

[Asterisk-Users] Eicon Diva Server 4BRI

Has anyone heard of success using the EICON DIVA Server 4 BRI with chan_capi as a PSTN interface with ISDN/BRI and * ? Any issues? This looks like it should work in theory, Linux drivers, CAPI 2.0 support. Any feedback appreciated. Damon ___ Asterisk-U

[Asterisk-Users] asterisk can not hangup .usrWildcard X100P

this sort of behaviour happenning all the time. such as when call 11,then I hangup. but when I recall 11,then it is busy. os redhat 9. /usr/sbin/asterisk -r Asterisk CVS-HEAD-09/10/04-21:34:12, Copyright (C) 1999-2004 Digium. Written by Mark Spenc

Re: [Asterisk-Users] Asterisk 1.0.2, Zaptel 1.0.2, Linux 2.6.9 on a PCEngines WRAP\Soekris net4801 in Compact Flash

Citat Kristian Kielhofner <[EMAIL PROTECTED]>: > grub 0.94 (serial console) > linux 2.6.9 (compiled for the Geode SC1100, with many, many modules) > zaptel 1.0.2 (with ztdummy for 2.6) > unixODBC with myODBC > mysql > perl > ncurses > full terminfo database > OpenSSH > perl5 (with modules) > glibc

Re: [Asterisk-Users] Asterisk 1.0.2, Zaptel 1.0.2, Linux 2.6.9 on a PCEngines WRAP\Soekris net4801 in Compact Flash

Citat Kristian Kielhofner <[EMAIL PROTECTED]>: > grub 0.94 (serial console) > linux 2.6.9 (compiled for the Geode SC1100, with many, many modules) > zaptel 1.0.2 (with ztdummy for 2.6) > unixODBC with myODBC > mysql > perl > ncurses > full terminfo database > OpenSSH > perl5 (with modules) > glibc

[Asterisk-Users] Voicemail Mailbox Configuration

Hi, How do I configure mailbox to play a different gsm prompt that default setup. I would like to replace unaivalable and busy message with a new sound file that I created. All comments are welcomed and greatly appreciated. Darly ___ Asteri

[Asterisk-Users] Segmentation fault res_features.so

Have anyone some idea ? Asterisk - latest cvs, RedHat9 == Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found [chan_modem.so] => (Generic Voice Modem Driver) == Parsing '/etc/asterisk/modem.conf': Found == Loading modem

[Asterisk-Users] SER-->Asterisk-->GNUGK Accounting Problem

Hello gnugk, ser and asterisk users! I am using asterisk as a SIP-to-H323 translator in order to call my itsp provider that only support H.323 so this is my scenario: --|xlite|-->|SER-SIP_Proxy|-->|Asterisk_oh323|-->|GNUGK_Proxy|-->|h323GW_MyProvider|-->PSTN x-lite its a sip client registere

Re: [Asterisk-Users] How change default law for T100P

--On Wednesday, November 03, 2004 15:40 -0700 Manuel Marin <[EMAIL PROTECTED]> wrote: I would like to know if there is a way to change default ulaw for a T1 card. I see in the zap show channel X that is working as ulaw. How do I change it in zapata.conf or zaptel.conf to alaw. Iam interconnectin

[Asterisk-Users] Re: Re: [Serusers] asterisk can not hangup .user Wildcard X100P

Flynn, this sort of behaviour happenning all the time. such as when call 11,then I hangup. but when I recall 11,then it is busy. os redhat 9. /usr/sbin/asterisk -r Asterisk CVS-HEAD-09/10/04-21:34:12, Copyright (C) 1999-2004 Digium. Written by Mar

Re: [Asterisk-Users] Asterisk X100P doesnot Hangup

On November 3, 2004 08:50 pm, wangxingtai wrote: > asterisk-usersïæåï > > Asterisk X100P doesnot Hangup So fix it. (you gave us no data, no testcases, no details whatsoever... what did you expect?) -A. ___ Asterisk-Users mailing list [EMAIL PROTE

[Asterisk-Users] Asterisk X100P doesnot Hangup

asterisk-users，您好！ Asterisk X100P doesnot Hangup 致 礼！ wangxingtai [EMAIL PROTECTED] 　　2004-11-04 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/lis

Re: [Asterisk-Users] Automatically restart asterisk if not running

Matthew Marlowe wrote: I once found a script, I think it was on the mailing list that would run in crontab and restart asterisk if not running... Does anyone happen to have a copy of that? The other option would be to run asterisk with: safe_asterisk instead of asterisk This will restart if it cras

Re: [Asterisk-Users] marginal voicemail prompt sound quality

Damon Estep wrote: --SNIP-- The issue can be made more apparent by increasing verbosity of the CLI (-c), the more verbose it is the more problems there are with sound quality. This appears to be limited to playback of included .gsm sounds. Voice channel and recorded vmail sound fine. --SNIP-- I

[Asterisk-Users] SIP registration/dialing problem.

Hello! I have a Grandstream and a Cisco SIP phone, and I'm trying to make a call between them. I added this to my sip.conf: ; Grandstream [1001] type=friend host=dynamic ; cisco phone [1002] type=friend host=dynamic It appears that they register correct: grok*CLI> sip show peers Name/usernameH

Re: [Asterisk-Users] G.729 on YDL and MacOSX

On Wed, Nov 03, 2004 at 09:58:46PM +, Martin List-Petersen wrote: > On Wed, 2004-11-03 at 20:34, Benjamin on Asterisk Mailing Lists wrote: > > > > ftp.digium.com/pub/asterisk/g729/unsupported/linuxppc > > has somebody mirrored this somewhere? I never seem to have any luck > > with digium's ft

RE: [Asterisk-Users] An anniversary and a lament for FXOs

I have been quite successful and pleased with the quality of the Cisco MC3810 and *. I have a single X100p not as an FXO but only as a timing source for conferencing and other apps that require a zaptel timer. No echo and good quality. Jojo From: [EMAIL PROT

Re: [Asterisk-Users] Cisco 79XX - Using built-in 3way conference

Matthew Boehm wrote: This works fine if caller 1 and 2 are both other phones in the office or caller 1 is a phone in office and 2 is PSTN. Just doesn't work when both PSTN. Worked before.. What codecs are involved? If you are using low-bandwidth codecs for PSTN connections, the Cisco phones will r

RE: [Asterisk-Users] What do I need to ask my T1 supplier?

I know we have sold XO integrated access voice /data to customers that used PRI interfaces on their PBX, so you should be able to order the voice provisioned as follows; ISDN/PRI NI-2 (national ISDN 2)Format (ATT 5ESS Will also work) Transmit at least 4 numbers on DID Order a block of DID numbers

[Asterisk-Users] Cisco 79XX - Using built-in 3way conference

Hey guys, This has worked before but for some reason isn't anymore and I have no clue what to check. Here are the steps I follow: 1. Place call to PSTN number. They answer and we talk. 2. I press 'Conference' button on Cisco phone. 3. Line 1 is now on hold and I get a new dial tone. 4. Place call

Re: [Asterisk-Users] What do I need to ask my T1 supplier?

>They will split the T1 line into 10 channels of voice and 14 channels of data. >From what I understand, they will terminate the T1 into a channel bank, and >then from that give is 10 POTS phone jacks and one data port (to go to an >Adtran router for our Internet access). >Any comments and/or su

Re: [Asterisk-Users] What do I need to ask my T1 supplier?

On Nov 3, 2004, at 5:32 PM, Scott Nelson wrote: My employer is switching to a new T1 supplier (it was AT&T, we are now going with XO), and sometime in the future we want to replace our PBX with an Asterisk system. What do I need to know to make sure the T1 line is "provisioned" (is that the ri

[Asterisk-Users] How change default law for T100P

I would like to know if there is a way to change default ulaw for a T1 card. I see in the zap show channel X that is working as ulaw. How do I change it in zapata.conf or zaptel.conf to alaw. Iam interconnecting a Meridian PBX but I need to configure it as alaw. __

[Asterisk-Users] getting cid from spa3k pstn to *

i am still going crazy with this one. i can not get callerid from a call received on the spa3k pstn to asterisk. THIS USED TO WORK! in order to get the cid from the spa3k to *, i need to turn on PSTN / PSTN-To-VoIP Gateway Setup / PSTN CID For VoIP CID: = YES the sip.conf entry looks like [

[Asterisk-Users] What do I need to ask my T1 supplier?

My employer is switching to a new T1 supplier (it was AT&T, we are now going with XO), and sometime in the future we want to replace our PBX with an Asterisk system. What do I need to know to make sure the T1 line is "provisioned" (is that the right term?) correctly for a Digium T100P/TE410P/TE

Re: [Asterisk-Users] Automatically restart asterisk if not running

Hi Il mer, 2004-11-03 alle 21:41, Matthew Marlowe ha scritto: > I once found a script, I think it was on the mailing list that would > run in crontab and restart asterisk if not running... Does anyone > happen to have a copy of that? I suggest you to use something like a superdaemon, ie a process

[Asterisk-Users] MusicOnhold on Bridged calls

Now that my bridged calls are working fine with ISDN I have a question ?   When my customers call in and my ext is not available the call is routed out to my mobile.   Everything works but I would like to know if there is a way of having the

RE: [Asterisk-Users] Installing X100P Asterisk - Unable to createchannel of type 'Zap'

Look at your logs in /var/logs/asterisk/. I am pretty certain it is a fault in your /etc/asterisk/Zapata.conf file. Vikas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Seth Remington Sent: Wednesday, November 03, 2004 1:01 PM To: Asterisk Users Maili

Re: [Asterisk-Users] Sip clients not longer registering

Message: 8 Date: Wed, 03 Nov 2004 22:32:46 +0100 From: "Olle E. Johansson" <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Sip clients not longer registering To: Asterisk Users Mailing List - Non-Commercial Discussion <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text

Re: [Asterisk-Users] ISDN Dialplan

It´s solved  now. You are right it  was a matter of formating the outgoing number.     Thanks   Paulo   >>>[EMAIL PROTECTED] 11/03 1:34 pm >>>   On Wed, 3 Nov 2004, Paulo Adriano wrote:     >Aft

[Asterisk-Users] Hookflash with cisco 827-4v

Hi, I have setup my cisco 827-4v to handle incoming and outgoing calls with asterisk with an isdn line. My problem is that I can't use features like call forwarding or holding because the router doesn't support the hookflash key. Is there anyone that has succeded in using the hookflash with the

Re: [Asterisk-Users] G.729 on YDL and MacOSX

On Wed, 2004-11-03 at 20:34, Benjamin on Asterisk Mailing Lists wrote: > > > ftp.digium.com/pub/asterisk/g729/unsupported/linuxppc > > has somebody mirrored this somewhere? I never seem to have any luck > with digium's ftp site. the few times that I did manage to connect, > the connection died so

Re: [Asterisk-Users] ISDN Dialplan

On Wed, 3 Nov 2004, Paulo Adriano wrote: > After trying that syntax I still have the same problem, one thing is > very strange is the number that Asterisk reports as the incoming. My > ESN*s numbers are 219898334 and 219898335 but on the console I see > 219898334,1 and 219898335,1 Are you sure it

Re: [Asterisk-Users] Sip clients not longer registering

David Filion wrote: Hi, We have been using Asterisk since version 0.9x with little or no problems. However, for an unknow reasons, our sip clients can nolonger register. We updated to Asterisk 1.0.2 hoping that would solve the problem, but no luck. There is a chance that our change NAT logic

Re: [Asterisk-Users] Sip clients not longer registering

David Filion wrote: Hi, We have been using Asterisk since version 0.9x with little or no problems. However, for an unknow reasons, our sip clients can nolonger register. We updated to Asterisk 1.0.2 hoping that would solve the problem, but no luck. Seems to me that the HT486 never receives th

[Asterisk-Users] Sip clients not longer registering

Hi, We have been using Asterisk since version 0.9x with little or no problems. However, for an unknow reasons, our sip clients can nolonger register. We updated to Asterisk 1.0.2 hoping that would solve the problem, but no luck. Here is the entry from sip.conf for one of our clients: [1001220

[Asterisk-Users] G.729 for Asterisk: new version released

Download site: http://www.readytechnology.co.uk/open/g729 Major enhancements: - accepts packets with VAD stuff at the end (please test this with your hardware and give me feedback if it still doesn't work with some devices, you will probably see error messages on the console if bad sized frame

Re: [Asterisk-Users] Installing X100P Asterisk - Unable to create channel of type 'Zap'

On Wed, 2004-11-03 at 13:02, Frank Kostin wrote: > Hello list, > I am trying to install a Digium X100P into a Redhat Asterisk. > Kernel seems to be OK, card OK. > Zaptel Configuration seems to be OK. > # ztcfg -vv > Channel map: > Channel 01: FXS Kewlstart (Default) (Slaves: 01) > 1 channels config

[Asterisk-Users] Configuring MTA-V102 through TFTP, HTTP, HTTPS for Asterisk

I have the MTA-V102 SIP Adapter and I would like to update it through the auto-provisioning feature in the device by either TFTP, HTTP, or HTTPS. The instructions state that it will take a XML configuration file, but gives no details on the structure of the XML file or what it should be named. I

[Asterisk-Users] Automatically restart asterisk if not running

I once found a script, I think it was on the mailing list that would run in crontab and restart asterisk if not running... Does anyone happen to have a copy of that? Thanks -- MBM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.co

Re: [Asterisk-Users] Dropped calls with analog lines using TDM400P

Hiya Andres, I stared to have exactly the same problem - very soon after enabling the busydetect=yes in zapata.conf. Used to work flawless with it set to 'no'. The only reason I turned it on was I was trying to busy line detection and auto redials. Ive set it back off at the mo - just to be sure

Re: [Asterisk-Users] G.729 on YDL and MacOSX

> > ftp.digium.com/pub/asterisk/g729/unsupported/linuxppc has somebody mirrored this somewhere? I never seem to have any luck with digium's ftp site. the few times that I did manage to connect, the connection died soon afterwards. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg

RE: [Asterisk-Users] SIPGate for outgoing calls

Try: register => <>:<>@sipgate.co.uk/ And canreinvite=no disallow=all allow=ulaw allow=alaw allow=gsm Simon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Thursday, 4 November 2004 6:11 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SI

Re: [Asterisk-Users] Dropped calls with analog lines using TDM400P

hi Il mer, 2004-11-03 alle 21:09, Andres Maduro ha scritto: > I am currently using busydetect=yes with busycount=5 and I have tweaked the raise busycount to at least 6 or even 8. In my experience 4 is an assurance to see dropped calls, 5 also, but less frequently, 6 never. but I prefer 8, to be

[Asterisk-Users] Dropped calls with analog lines using TDM400P

Hi, I have successfully configured and built Asterisk and now it is working fine from the functionality point of view as sometimes we are getting dropped calls. The problem I am getting with POTS lines even if I receive/make a call from a sip or analog phone is that the call may be dropped rando

Re: [Asterisk-Users] An anniversary and a lament for FXOs

On Wed, 3 Nov 2004 [EMAIL PROTECTED] wrote: > I've tried X100p cards but found them horribly unreliable. I presently > use Sipura SPA-3000s but they're only marginally better. How is it that > my Panasonic 4 line SOHO phone system (KX-TG4000B) can have four > stable, reliable FXOs with no echo a

[Asterisk-Users] SIPGate for outgoing calls

I'm trying to get * to route external calls through SIPgate, and have obtained limited success. I can get calls routed to other SIPGate users, but when I try to route calls to the PSTN, I get an error announcement back from SIPGate. The SIPGate account is fine, I can use it to make calls to the PST

RE: [Asterisk-Users] FireFly Problems

Ok. For [user_test] I specifically disabled all codecs and ONLY allowed Ulaw. This seems to have forced firefly into using Ulaw, regardless of what’s checked in the program.    Also, I left dtmfmode as rfc2833 in sip.conf as well as set rfc2833 in FireFly, and it appears to work. I didn’

Re: [Asterisk-Users] G.729 on YDL and MacOSX

On Wed, 3 Nov 2004 11:57:44 -0600, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > Ok, check out ftp.digium.com/pub/asterisk/g729/unsupported/linuxppc > > Knock yourself out (oh, and look at the date on that file too :-) ) date? "The requested URL /pub/asterisk/g729/unsupported/linuxppc was not f

Re: [Asterisk-Users] asterisk as a sip registrar and user accounts

Ashling, I have my sip.conf configured like this for Xlite UA: - ; sip.conf [6387] ; X-Lite client 6387 type=friend secret= auth=md5 nat=yes host=dynamic reinvite=yes ; Puede salir a cualquier lado canreinvite=yes ; Puede salir a cualquier lado qualify=1000 ;dtmfmode=inba

Re: [Asterisk-Users] ISDN Dialplan

Hi All, Let me also add my very simple dialplan. Thanks CLI> show dialplan [ Context 'toll-access' created by 'pbx_config' ]   Include =>    'local-access'   &___

Re: [Asterisk-Users] ISDN Dialplan

Hi ALL   After trying that syntax I still have the same problem, one thing is very strange is the number that Asterisk reports as the incoming. My ESN´s  numbers are    219898334  and  219898335   but on the console I see  219898334,1   and 219898335,1

[Asterisk-Users] FireFly Problems

How come FireFly doesn’t give me an Inband DTMF option? Only RFC2833 and Info. RFC2833 is the default, so I left it that way. I also unchecked all the codecs except g711ulaw to force that codecs usage. However, when I go to place a call, I get this:   Nov  3 13:18:44 WARNING[53641241]: ds

[Asterisk-Users] Re: Fw: Re: How far is IAX to be a Standard

On Wednesday 03 November 2004 12:55 pm, Randy Bush wrote: > > ... > > however don't think it's fair to continue this discussion on this list. > > certainly not the ad homina. > > but i think it is very important to try to understand how to get > the best possible technical solutions advanced in the

[Asterisk-Users] ASTCC - cdrs database and number-entry timeout questions

Hello, I have astcc installed and working, however I am having two problems: 1. Nothing gets written to the cdrs table, and I get an error "Failed to insert into database" in the log. The database settings are correct, as all the other aspects of astcc work fine. I really need to get a log of th

[Asterisk-Users] manager api originate doesn't give detailed information

Hi all, I am playing with Asterisk since few monts and I have some problems in mastering faiulre handling I built an application which call many outgoing calls through the manager-api command originate. I set the outgoing channel i.e zap/g1/$EXTEN and also the context, extension, priority tripl

RE: [Asterisk-Users] Speed Dial / New Context

Doh. Slaps head. Many many thanks for making me look stupid :) If anyone is interested, here is the logic that I used for our speed dial: ; custom/speed-2-digit "Enter your 2 digit speed dial number" ; custom/speed-target "Enter The destination phone number" ; custom/speed-saved "Your speed dial

[Asterisk-Users] RE: IAXys or IAX Softphones cannot call SIP phones

I've found the problem. Sip read: SIP/2.0 400 Bad Request Via: SIP/2.0/UDP 172.31.30.3:5060;branch=z9hG4bK658d8eed;rport From: ""Nik Martin IAX" ;tag=as40e79563 < To: Call-ID: [EMAIL PROTECTED] Date: Wed, 03 Nov 2004 15:21:52 GMT Warning: 399 Bad Request - 'Malformed/Missing F

Re: [Asterisk-Users] addon_mysql_cdr allows fraud by sip or iax users

Hi, > Are there any solutions to avoid cdr manipulations > by users, who prepare special caller id strings? set the callerid from asterisk. don't let others to set it. Matteo. -- Brancaleoni Matteo <[EMAIL PROTECTED]> Espia Srl ___ Asterisk-Users ma

[Asterisk-Users] Installing X100P Asterisk - Unable to create channel of type 'Zap'

Hello list, I am trying to install a Digium X100P into a Redhat Asterisk.Kernel seems to be OK, card OK.Zaptel Configuration seems to be OK. # ztcfg -vvChannel map:Channel 01: FXS Kewlstart (Default) (Slaves: 01)1 channels configured.Asterisk works fine with IP SIP but not with X100PI get the error

[Asterisk-Users] Re: Fw: Re: How far is IAX to be a Standard

> ... > however don't think it's fair to continue this discussion on this list. certainly not the ad homina. but i think it is very important to try to understand how to get the best possible technical solutions advanced in the market and in the ivtf. and iax, though maybe not perfect, has a lo

[Asterisk-Users] addon_mysql_cdr allows fraud by sip or iax users

Hi, it wasn't a fraud, just a coding error, by one of our customers: There were binary data in the caller id passed by SIP, obviously including an apostrophe. addon_mysql_cdr seems not to mask those binary data or apostrophes (') and therefore the mysql insert command failed. That's good for the cu

Re: [Asterisk-Users] G.729 on YDL and MacOSX

On Tue, Oct 26, 2004 at 08:13:07PM -0200, Marcelo Pacheco wrote: > If a PPC platform can double channel capacity, I would be pressuring Digium > like crazy to get them to work on it, I'd be in line to purchase over 400 > channels of G729 licenses for PPC. > > Marcelo Pacheco > > > However, with

Re: [Asterisk-Users] Speed Dial / New Context

Asterisk wrote: However, I would like before * makes the call is to "pretend" that the speeddial number has been entered directly from the phone. This is because I want to push the speed dial number through the same sort of checking that the sip phone's context does, without duplicating the logic.

[Asterisk-Users] Speed Dial / New Context

I've been setting up a basic speed dial for sip users that don't have a speed dial function on their phones (aastra 480i specifically, though I think that it also affects grandstream) So far so good - A) user can enter a specific extension, enter the 2 digit speed dial number, and the target pho

Re: [Asterisk-Users] Call pickup and snom phones

You need to have the pickupgroups added in sip.conf Then - in order to pick up, use *8 ( and not *8# ). Under each extension ( here in group 1 ) add the following lines to sip.conf : callgroup=1 pickupgroup=1 -- Pertti [EMAIL PROTECTED] wrote: First of all, excuse me if this is considered as OT.

[Asterisk-Users] asterisk port problem?

Hi all, As I mentioned in a previous email, I am getting an error (chan_sip.c: 7533 cant handle request) and can't register any client with asterisk.I ran the netstat command (see below)at the shell prompt to see what ports were listening and noted that the status of the asterisks port were not se

[Asterisk-Users] Maddog weighs in on the state of the Linux [Asterisk plug]

http://searchenterpriselinux.techtarget.com/qna/0,289202,sid39_gci1022137,00.html Excerpt: -begin- What's the big story in enterprise Linux that's going to impact IT shops in the next year? Why? Hall: Actually, I think the next big story is in the VoIP [voice over Internet Protocol] space, and i

[Asterisk-Users] SendDTMFthrough the manager

Title: Message I have been researching for a while and i can't seem to find a way to do this.   Is there a way to SendDTMF tones to a channel without changing the context a caller is in?   Here is the scoop,  I have a conference room with 2 channels in it.  I would like to send DTMFTones i

Re: [Asterisk-Users] Asterix-to-PBX

Hi Peter, The scenario is like this: [EMAIL PROTECTED] flux(Internet) --> Asterix --> TE410P(E1)-->PBX-->regular-office-phones. I want PBX to use another(the actual) E1 line for normal PSTN dialin/dialout calls but also with the other dedicated E1-TE410P line I want to be used for the same

Re: [Asterisk-Users] oh323 compilation error

This version is ancient. It doesn't compile with fresh Asterisk code. Use the latest version (0.7.0) of asterisk-oh323. Michael. Pavel Siderov wrote: Hi, while trying to compile oh323 version 0.5.10 I got these errors [EMAIL PROTECTED]:/usr/src/asterisk/asterisk-oh323-0.5.10

Re: [Asterisk-Users] Anyone have bristuff's zaphfc module coexisting with wcfxs for the tdm400p?

Steve, please make sure that ztcfg is not run twice! Usually modprobing the TDM400P driver will trigger ztcfg automatically. After that, running it again manually will make the zaphfc cards totally non-responsive. This is known bug that is unfortunately still under investigation... So, do this:

[Asterisk-Users] Remote MWI

Hey All, I have two asterisk servers, one local and one remote. The voicemail application lives on the remote server. Can anyone think of anyway to light a MWI on a SIP client connected to the local server? Thanks, ~c ___ Asterisk-Users mailing li

RE: [Asterisk-Users] An anniversary and a lament for FXOs

> So then the system I'm building (P4 2.66GHz) should do fine in the > encoding/decoding (which I assume accounts for most of the echos, and poor > quality). > > use Sipura SPA-3000s but they're only marginally better. How is it that > > my Panasonic 4 line SOHO phone system (KX-TG4000B) can have

RE: [Asterisk-Users] An anniversary and a lament for FXOs

Hi Rich, I've had good success with the cards in HP ProLiant ML110 servers, as well as on clones with Asus P4C800-E motherboards. I believe the chipset on both of these systems is the Intel 875/ICH5... FYI, I'm not in the US -- I'm in Canada. But that doesn't make a difference as far as PSTN si

Re: [Asterisk-Users] An anniversary and a lament for FXOs

> This week marks one year since I first setup an Asterisk server in the > hopes of transitioning my home office to a total VoIP system. I have only been doing it since May > I've tried X100p cards but found them horribly unreliable. The reason I'm answering is to weigh in with the fact that I

Re: [Asterisk-Users] Polycom Phones

Does Asterisk honour this RFC? On 3 November 2004 11:32, Michael Devenijn wrote: > I searched in the RFCs documents and found extra information in RFC3326 > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Peter > Osborne > Sent: woensdag 3 november 2

Re: [Asterisk-Users] Good ringing plans for small office

On Wed, 2004-11-03 at 16:11, Christopher TenHarmsel wrote: > At the place I work we're using Asterisk to run our in-office phone > system. We have about 15 employees and a total of about 5 hard phones. > Right now when asterisk receives an incoming call, it rings all 5 > phones, because we don

RE: [Asterisk-Users] Polycom Phones

I searched in the RFCs documents and found extra information in RFC3326 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Osborne Sent: woensdag 3 november 2004 15:28 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Polycom Phones Hi, I have Asterisk

Re: [Asterisk-Users] Good ringing plans for small office

On Wed, 2004-03-11 at 11:11 -0500, Christopher TenHarmsel wrote: > At the place I work we're using Asterisk to run our in-office phone > system. We have about 15 employees and a total of about 5 hard phones. > Right now when asterisk receives an incoming call, it rings all 5 > phones, because

[Asterisk-Users] oh323 compilation error

Hi, while trying to compile oh323 version 0.5.10 I got these errors   [EMAIL PROTECTED]:/usr/src/asterisk/asterisk-oh323-0.5.10# makefor x in wrapper asterisk-driver; do make -C $x all || exit 1 ; donemake[1]: Entering directory `/usr/src/asterisk/asterisk-oh323-0.5.10/wrapper'./check_ver

Re: [Asterisk-Users] Good ringing plans for small office

On Wed, 3 Nov 2004 11:11:42 -0500, Christopher TenHarmsel <[EMAIL PROTECTED]> wrote: > At the place I work we're using Asterisk to run our in-office phone > system. We have about 15 employees and a total of about 5 hard phones. > Right now when asterisk receives an incoming call, it rings all 5 >

Re: [Asterisk-Users] An anniversary and a lament for FXOs

Just to provide some more votes in the X100P column, I have an X100P as my telco interface and have had absolutely no echo problems at all. I've called from it and I've called into it and held long conversations and never heard an echo even once. It's not a host CPU issue as Asterisk is runnin

[Asterisk-Users] Good ringing plans for small office

At the place I work we're using Asterisk to run our in-office phone system. We have about 15 employees and a total of about 5 hard phones. Right now when asterisk receives an incoming call, it rings all 5 phones, because we don't really have a "receptionist". I was wondering if anyone has ha

Re: [Asterisk-Users] asterisk as a sip registrar and user accounts

On Wed, 2004-11-03 at 10:23, Ashling O'Driscoll wrote: > Thanks for the reply.I think I have this done but it still doesnt > seem to work. > > I set up a xlite softphone which Im attempting to register with > asterisk. However i am getting the following error on asterisks: > > Nov 3 14:25:13 NOTI

Re: [Asterisk-Users] An anniversary and a lament for FXOs

On November 3, 2004 10:42 am, David Ishmael wrote: > So then the system I'm building (P4 2.66GHz) should do fine in the > encoding/decoding (which I assume accounts for most of the echos, and poor > quality). It's not just the raw horsepower, for lack of a better term. The chipset and motherboar

[Asterisk-Users] Call pickup and snom phones

First of all, excuse me if this is considered as OT. I'm trying to use the asterisk call pickup function on the 220 Snom phones, in other phones works well. But if I dial *8# in the snom phones, the call is no picked up. In others phones this combo of keys works perfectly. Someone could give me a

Re: [Asterisk-Users] Voicemail: howto disable vm-intro.gsm at the end of message?

On Wed, 2004-11-03 at 15:30, Eric Wieling wrote: > show application voicemail > > Pay special attention to the "s" option. > Duh! Thanks for the reminder to startup my brain next time I ask something :) Me == total moron for having missed the utter obvious. Appreciate the answer. All works well

Re: [Asterisk-Users] Reading extensions from MySQL database

On Tuesday 02 November 2004 15:16, Director General: NEFACOMP wrote: > Hi list. Does anyone know of any configuration that will make asterisk > read the extensions from a MySQL database instead of reading them from > the configuration files? > > check this one http://bugs.digium.com/bug_view_page.p

RE: [Asterisk-Users] An anniversary and a lament for FXOs

So then the system I'm building (P4 2.66GHz) should do fine in the encoding/decoding (which I assume accounts for most of the echos, and poor quality). -Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Wednesday, November 03, 20

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