[asterisk-users] TE220 PCI express performnace

Dear all I have going to put more PRI line in my organization theseday i have decide to put all PRI on TE220 dual span e1/t1 pci express card so what about the performnace and installation of this card is there anybody useing this card suggest me .?? PGP Signature-- Satish

Re: [asterisk-users] Video Call

Hi, Thereis any application (SIP) + Video can installed at phone, so with this application can commnication with asterisk to do video call Thanks On 11/4/07, Yann JOUANIN [EMAIL PROTECTED] wrote: Hi, A few time ago I read an article which explain how to use a 3G video phone with Asterisk.

[asterisk-users] Not Hearing hello-world Play

Hi Asterisk Gurus! My lab asterisk box has 1 FXO and 1 FXS ports in it. I connect a GSM phone to the FXO port. I connect a regular corded phone to the FXS port. The Dial() application for both incoming and outgoing calls specifies the A(hello-world) flag. From another GSM phone, if I call the

Re: [asterisk-users] TE220 PCI express performnace

On Mon, Nov 05, 2007 at 12:10:45AM -0800, satish patel wrote: Dear all I have going to put more PRI line in my organization theseday. i have decide to put all PRI on TE220 dual span e1/t1 pci express card so what about the performnace and installation of this card is there

Re: [asterisk-users] Need Reference sites

Hi, Various site available for asterisk,listed below, www.asterisk.org www.voip-info.com www.digium.com and best is search in www.google.com On Nov 5, 2007 5:22 AM, Michael Davidson [EMAIL PROTECTED] wrote: Hi, I'am comparative newbie to the world of Asterisk. I'd like to introduce an

Re: [asterisk-users] RTP Read too short

On 11/3/07, John Faubion [EMAIL PROTECTED] wrote: Am I the *ONLY* one that has this issue? John Faubion -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Faubion Sent: Thursday, November 01, 2007 11:01 AM To: Asterisk Users Mailing List -

Re: [asterisk-users] asterisk-users Digest, Vol 40, Issue 5

i dont know how to remove these errors. But i think you should try asterisk-addons package available from asterisk download site. it contains the h323 channel also. You only need to compile it. remove the asterisk-oh323 package from your system and install the asterisk-addons package. I hope this

[asterisk-users] How to disable Asterisk 407 Proxy Authentication Required Challenge response

Hi, I have an UAC registered in VoIP provider. (register command in sip.conf) When I try to make call from PSTN through this VoIP provider, when INVITE reaches asterisk is sents 407 Proxy Authentication Required Challenge response. How can I disable this, because I want to allow any external call

Re: [asterisk-users] asterisk as a gateway

Thanks once again..I will check with addon package and let you know the status.. Date: Mon, 5 Nov 2007 15:30:49 +0500 From: Rizwan Hisham [EMAIL PROTECTED] Subject: Re: [asterisk-users] asterisk-users Digest, Vol 40, Issue 5 To: Asterisk Users Mailing List - Non-Commercial Discussion

[asterisk-users] How to delete voice mail messages?

Hello all, Could I create a script to delete the first messages on my voice mail? In this script should I update any messages index file or there isn't any file to index them? Could you share any script to do that? Thanks in advance. VoipCrazy. ___

Re: [asterisk-users] How to delete voice mail messages?

On 12:15, Mon 05 Nov 07, voip crazy wrote: Hello all, Could I create a script to delete the first messages on my voice mail? In this script should I update any messages index file or there isn't any file to index them? Could you share any script to do that? Hi, Voicemails are stored in

[asterisk-users] Which Variable???

Hi Gurus! Please excuse this pesky Asterisk rookie:-) I just wanted to know which channel variable tells asterisk the number of rings before an incoming call on FXO channel is answered? I looked through zapata.conf.sample and other places and could not find something there readily. Thanks,

Re: [asterisk-users] Dynamic Queue Members - Auto Logoff

You can use RemoveQueueMember(queuename) to dynamically remove the agents. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nick Brown Sent: Sunday, November 04, 2007 11:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject:

Re: [asterisk-users] Need Reference sites

We have implemented asterisk. We are a tier one automotive tooling supplier. We have approx. 120 extensions in use plus 8 fax extensions. We also have a two port cell phone adapter so when we call out T-Mobile phone, we are using the free (included) T-mobile to T-mobile minutes. We have also

Re: [asterisk-users] Compatibility Issues with dell poweredge 195and TE110P card

2950s work fine. I have had the parity error for over a year with no noticable problems. It is working fine. I did have to make some IRQ changes to clean up the system. I did these on my Dell 1750 test machine, but have made the same changes on my production machine. The changes basically

Re: [asterisk-users] Which Variable???

Jeng Yu wrote: Hi Gurus! Please excuse this pesky Asterisk rookie:-) I just wanted to know which channel variable tells asterisk the number of rings before an incoming call on FXO channel is answered? I looked through zapata.conf.sample and other places and could not find

Re: [asterisk-users] Which Variable???

Jeng Yu wrote: Hi Gurus! Please excuse this pesky Asterisk rookie:-) I just wanted to know which channel variable tells asterisk the number of rings before an incoming call on FXO channel is answered? I looked through zapata.conf.sample and other places and could not find

Re: [asterisk-users] Dynamic Queue Members - Auto Logoff

On 11/5/07, Nick Brown [EMAIL PROTECTED] wrote: Another quick question (Spending the day trying to get this project sorted and tucked away) If I am dynamically adding queue members, they will not abide to settings within agents.conf will they? correct. Ie. I need the equivalent of

Re: [asterisk-users] Kirk IP600/3 Wireless Server SIP config

On Fri, 26 Oct 2007, Benny Amorsen wrote: RB == Remco Barendse [EMAIL PROTECTED] writes: RB Hi list! Is anyone using the Kirk IP600/3 with SIP firmware RB connected to Asterisk? Yes. RB If anyone would be willing to share the dump of their IP600 config RB file, i would really appreciate

[asterisk-users] OT: Which SIP method to use for this specific behaviour ?

Hello, Let SIP extensions 1001 and 1002 belong to an Asterisk calling group : whenever an coming call reaches this calling group, both extensions 1001 and 1002 receive a SIP INVITE message which makes these 2 phones starting to ring. When a callee picks up his phone, the other extension receives

[asterisk-users] Parameters effect on the success registeration

Hi All; nat=yes for example, it effects on the success of the registeration. What are the parameters that might let the registeration fail when I need to register Asterisk on a softswitch using register = ? Any help? Regards Bilal __ Do You

Re: [asterisk-users] OT: Which SIP method to use for this specificbehaviour ?

Search for: Reason: SIP ;cause=200 ;text=Call completed elsewhere From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Sent: 05 November 2007 15:22 To: Asterisk Users Mailing List - Non-Commercial Discussion

Re: [asterisk-users] OT: Which SIP method to use for this specificbehaviour ?

Thanks for the tip. If I may ask, do you if this signaling is support in Asterisk 1.4 ? 2007/11/5, Steve Langstaff [EMAIL PROTECTED]: Search for: Reason: SIP ;cause=200 ;text=Call completed elsewhere -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL

Re: [asterisk-users] Asterisk versions and H323

Dear Dovid; Thanks a lot for the nice reply and support. I need a document on this addon (file name to be downloaded, steps to compile, where i can find the h323 module in this addon, and the configuration for h323)? Regards Bilal There is a version in the asterisk add-ons that

Re: [asterisk-users] Which Variable???

On Mon, 5 Nov 2007, Eric ManxPower Wieling wrote: Jeng Yu wrote: Hi Gurus! Please excuse this pesky Asterisk rookie:-) I just wanted to know which channel variable tells asterisk the number of rings before an incoming call on FXO channel is answered? I looked through

[asterisk-users] Meetme - how to protect the conference?

Hi all, I am just wondering - it there any way how to protect a conference from being abused by someone? I know I can request pin, but that pin is then hardcoded in meetme.conf and normal user can not change it. I would like to establish an admin user who could set a pin for the conference to be

[asterisk-users] Problem with CDR userfield not being set

I'm trying to use the MySQL CDR records. According to dialplan show, the line in the dialplan is: 11. Set(CDR(userfield)=${billing_code}) [pbx_ael] It looks like the value is being set when I watch the console during the call: -- Executing [EMAIL PROTECTED]:11] Set(SIP/icall-0075a2e0,

Re: [asterisk-users] Meetme - how to protect the conference?

You could use meetme realtime and have the admin update the pin via a web interface instead. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ondrej Valousek Sent: Monday, November 05, 2007 09:38 To: Asterisk Users Mailing List - Non-Commercial Discussion

Re: [asterisk-users] Problem with CDR userfield not being set

On Mon, 2007-11-05 at 09:40 -0800, James Moore wrote: I'm trying to use the MySQL CDR records. According to dialplan show, the line in the dialplan is: 11. Set(CDR(userfield)=${billing_code}) [pbx_ael] It looks like the value is being set when I watch the console during the call:

Re: [asterisk-users] Which Variable???

The call is not answered until you answer it with either the Answer app, or issuing a playback command etc. On 11/5/07, Gordon Henderson [EMAIL PROTECTED] wrote: On Mon, 5 Nov 2007, Eric ManxPower Wieling wrote: Jeng Yu wrote: Hi Gurus! Please excuse this pesky Asterisk rookie:-)

[asterisk-users] Free T1 Card?

Gang, I recall several months ago that there was a company that was giving away a free 1-port T1 card, with some specific conditions. Do any of you recall who that was? My Google searches are coming up empty and now I'm wondering if I was hallucinating... Thanks, MC

[asterisk-users] Testcall

# ./testcall testcall.conf Chan 1, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025860' to '013331339767' Chan 2, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025861' to '013331339768' Chan 3, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025862' to

[asterisk-users] Help: Static and dropped calls

Does anybody know why am getting a lot of static and sometimes dropped calls from my asterisk server. Vitelity is my number provider if it matters. Thank you Jarga Jallow image001.jpg___ --Bandwidth and Colocation Provided by

[asterisk-users] Arbitrary limit on length of email address?

I'm trying to get emailing of voicemail messages to work and by and large it does... However one email address is quite long in comparison to others I am testing and it fails to get delivered. For example - this one works and gets delivered: [Nov 5 18:35:14] DEBUG[2509]: app_voicemail.c:1957

Re: [asterisk-users] AEX800 (TDM800 Express) - not detected

Mark J Elkins wrote: I have a AEX800 PCI Express card - sort of a TDM800 with PCI-Express. (or AEX844 - 4FXS 4FXO) It's not 'sort of', it *is* a TDM800 with a PCI Express bus interface. With only this card in the box Asterisknow gives me... no functional digium card found in

Re: [asterisk-users] Free T1 Card?

On Nov 5, 2007 1:14 PM, Michael Collins [EMAIL PROTECTED] wrote: Gang, I recall several months ago that there was a company that was giving away a free 1-port T1 card, with some specific conditions. Do any of you recall who that was? My Google searches are coming up empty and now I'm

Re: [asterisk-users] Arbitrary limit on length of email address?

Alan Lord wrote: Whereas this one: [Nov 5 18:36:02] DEBUG[2519]: app_voicemail.c:1957 sendmail: Sent [mail to [EMAIL PROTECTED] with command '/usr/sbin/sendmail -t' fails to get delivered and is 34 characters long. Both email accounts work otherwise and I have had no recorded

Re: [asterisk-users] Arbitrary limit on length of email address?

Per Jessen wrote: snip / Check your mail-logs. Was the email with the long address accepted and processed by your mail-server? Also look for traces of an incomplete email-address being used (or something like that). /Per Jessen, Zürich Thanks Per, I checked my exim logs and that

Re: [asterisk-users] Problem with CDR userfield not being set

On 11/5/07, Carlos Chavez [EMAIL PROTECTED] wrote: Do you have userfield=1 in your cdr_mysql.conf file? Thanks - that took care of it. - James ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing

Re: [asterisk-users] Meetme - how to protect the conference?

I am just wondering - it there any way how to protect a conference from being abused by someone? I know I can request pin, but that pin is then hardcoded in meetme.conf and normal user can not change it. I would like to establish an admin user who could set a pin for the conference to be

Re: [asterisk-users] RTP Read too short

I saw this with Grandstream GXP2000. When the phone is on a call and on mute, the phone sends SIP keepalive packets that are, indeed, too short. So asterisk is correct in this case. Grandstream said that they were just warnings and to ignore them. We have chosen to ignore Grandstream and move

Re: [asterisk-users] Testcall

You have other process using at least one of those 1-10 channels. If some other process have it, testcall cannot grab it. Other process could be other testcall instance or Asterisk itslef. On 11/5/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: # ./testcall testcall.conf Chan 1, class

[asterisk-users] PRI dialout problem with some numbers...

I have an Asterisk server (1.4.13) using PRI in Monterrey, Mexico. This is really the first server I have used with PRI in Mexico as we normally use MFC/R2. Everything seems to be working except that some numbers always seem to be busy when you dial them. All these numbers belong to

Re: [asterisk-users] Free T1 Card?

http://www.pikatechnologies.com/ -- Kristian Kielhofner Thanks, I guess I wasn't hallucinating! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)

Can anyone please point me in the right direction, provide me with OpenSER configuration, or any pointers on the subject. I tried to read all the material on how to write configuration files for OpenSER, but it is incomprehensible to me, and it is much harder that when I learning Asterisk 3

Re: [asterisk-users] Free T1 Card?

On Mon, 2007-11-05 at 10:14 -0800, Michael Collins wrote: I recall several months ago that there was a company that was giving away a free 1-port T1 card, with some specific conditions. Do any of you recall who that was? My Google searches are coming up empty and now I’m wondering if I was

Re: [asterisk-users] OT: Which SIP method to use for thisspecificbehaviour ?

No idea, sorry. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Sent: 05 November 2007 16:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: Which SIP method to

Re: [asterisk-users] Free T1 Card?

Is the offer still there? I work at a very poor college would greatly appreciate the ability to get stuff like that. Thanks! Kristian Kielhofner wrote: On Nov 5, 2007 1:14 PM, Michael Collins [EMAIL PROTECTED] wrote: Gang, I recall several months ago that there was a company that

[asterisk-users] Queue Statistics reporting

Anyone know of a good package for reporting on Queue statistics from Asterisk? Bob ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

[asterisk-users] Please explain the correct LED color for B410P

Hi. I have installed B410P in Europe and the cards works more or less ok. My question is what color should the LED's on the back of the card be when connected to the PSTN NT box? Is there anywhere some information on the expected LED color in any given state (idle, call active, cord unplugged

[asterisk-users] Two B410P cards in one machine

Hi. I have two B410P ISDN BRI cards in one machine running Asterisk on Ubuntu 7.04. One card connects to the PSTN network and is therefore in TE mode on all four ports and the other card is in NT mode and connects to a PBX. The Asterisk is used to remap features, callerid's and more from the

Re: [asterisk-users] Kirk IP600/3 Wireless Server SIP config

Where did you buy it , and how much did it cost ? ip600v3, base stations and phones ... 2007/10/26, Benny Amorsen [EMAIL PROTECTED]: RB == Remco Barendse [EMAIL PROTECTED] writes: RB Hi list! Is anyone using the Kirk IP600/3 with SIP firmware RB connected to Asterisk? Yes. RB Any

[asterisk-users] Asterisk OpenVZ

Hi All, I've got debian (etch), openvz and asterisk up and running using the openvz wiki guides. The examples use `apt-get install asterisk` and this will install 1.2.13. Has anyone gotten an VPS to compile the latest versions from source? Also, I'm unsure how the zaptel modules come into

[asterisk-users] 1.4 SIP Jitter Buffer

Hello, I'm running into a few situations on lossy network links where a SIP jitter buffer w/ some PLC would be helpful. My main TDM gateways are running 1.2 (which is solid, stable, reliable and very very very well behaved when you know it's limitations), but I'm considering upgrading them

Re: [asterisk-users] 1.4 SIP Jitter Buffer

Gregory We have many Asterisk 1.4.13 in production solid like a rock. Couples examples: a) Asterisk 1.4.13 + Unicall + 2 E1 MFCR2 Digium + Legacy PBX 60+ Extentions / IVR / 10~30 concorrent calls b) Asterisk 1.4.11 + 1 E1 ISDN PRI Digium 50+ Extentions / IVR / 5 Queues / ~2000 call/day