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Asterisk Users Mailing List - Non-Commercial Discussion
November 2007 - page 1
hostname in MySQL CDR records (6 Replies)
Jim Gottlieb — 119387816401 Nov 2007*
I would like to send the CDR records from all our machines around the world to a single database. But I need the hostname included with each record fo...
Connection astrisk to a RAS (portmaster) (2 Replies)
Nicolas Ross — 119387866901 Nov 2007*
Here's my planed setup : PRI from telco <--> (port 1 of A104d) * (port 2 of A104d) <--> PM3 The PM3, for those who don't know is l...
Call Failed (1 Reply)
Robert La Ferla — 119387901001 Nov 2007*
After so many rings when the party does not answer, my SIP phone says Call Failed. Why doesn't it just keep ringing? Here's the dial plan rule...
Asterisk 1.4.13 -- issue with parked calls (3 Replies)
Barry D. Hassler — 119388660901 Nov 2007*
I've tried to find other threads with this same topic, but haven't found any... Apologies if this already being discussed.... Running asterisk...
AEL2 and Callbacks (3 Replies)
Douglas Garstang — 119388808301 Nov 2007*
I am originating a command via the AMI with this... Action: Login Username: xxx Secret: yyy ACTION: Originate Async: yes Timeout: 60000 Exten: callbac...
SER with Asterisk intergration (3 Replies)
satish patel — 119391645101 Nov 2007*
Dear all anybody have implement SER with Asterisk in single machine ?? i have asterisk with 200 SIP device but i voice qulity and load of asterisk is ...
Kelly Opal — 119393280901 Nov 2007
Hi Is it possible to speed up the parking process. We receive a lot of calls during the day and it gets to be painful to wait for the call to be parke...
Help (2 Replies)
Jarga Jallow — 119393437101 Nov 2007*
I need help with my grand stream GXP2000 phones they keep freezing randomly. Any ideas? Jarga...
Michelle Dupuis — 119393627301 Nov 2007
We are connecting an asterisk box to a Nortel Option 61 via a T1 with PRI. We have hit a problem we cannot overcome; specifically, the Nortel is askin...
Antoine Megalla — 119393929201 Nov 2007*
Hi, I have a client who requires an Asterisk system with 1500 SIP clients. All clients will have ATAs (mostly Grandstream), so I think a single Asteri...
Martin Smith — 119394492701 Nov 2007
Hi folks, We have a very rare problem with Asterisk 1.2 where Chanspy reports the following: Oct 31 19:53:29 NOTICE[10490] app_chanspy.c: Attaching SI...
AEL2 and Callbacks (1 Reply)
Douglas Garstang — 119394953401 Nov 2007*
Dial(${destination},60,oL(${timeout}:${timeout_warning}:${timeout_warning_repeat}));Dial("Local/16505551212 at LegA-f707,2", "SIP/16505...
Outgoing PRI CID? (9 Replies)
Turbo Fredriksson — 119395722301 Nov 2007*
We have now got our new PRI line (10 channels, 100 numbers) connected and everything is working except the outgoing caller ID. Whatever SIP phone I...
DST (10 Replies)
Joe Acquisto — 119396712402 Nov 2007*
My Polycom phones are displaying time, off by one hour. Seems they are on the old DST rules. How do I fix this? joe a....
zaptel.conf missing (2 Replies)
kitti jaisong — 119399485602 Nov 2007*
Hi all I have installed zaptel on debian and missing file zaptel.conf in /etc/zaptel.conf .my system don't have card TDM please advice what the mi...
asterisk as a gateway (1 Reply)
Bincy K. Philip — 119400785702 Nov 2007*
Hello,Could anyone please give some information on configuring asterisk as a gateway. What contents have to add in h.323 .conf and extensions.conf fil...
ring group containing external 10-digit numbers (5 Replies)
Ryan Stille — 119401366702 Nov 2007*
I have a ring group setup that I'd like to ring a bunch of local extensions, plus a few outside lines. I want recipients to confirm the call by pr...
Two PRI setup questions (4 Replies)
Lutgring, Sam — 119402352602 Nov 2007*
I am in the process of implementing a new ISDN pri and have a couple of questions. This is a full 24 channels (23 B and 1 D) delivered over a T1 inter...
Arpit Mehta — 119402504902 Nov 2007*
Hello Asterisk Users, I wanted to know a simple way in which I could read some file from a console (say by using system command) and based on that eit...
Baji Panchumarti — 119402511802 Nov 2007
http://nooss.org/wiki/Installing_Asterisk_on_... Thank you everyone for all your help....
Costa Dinoteli — 119403885802 Nov 2007*
Hi, We created a dial plan which performs and outbound dial out and deliveres a message to a receipient What call method/option in extensions or anywh...
sip show peers in 1.4.13 (3 Replies)
Jerry Geis — 119403887102 Nov 2007*
What happened to "sip show peers" in 1.4.13? Jerry...
res_mysql versus res_odbc (3 Replies)
Tony Plack — 119404119202 Nov 2007*
...
bilal ghayyad — 119404858803 Nov 2007
Dear Amit; Special thanks for your greate help and support. Sorry for delaying in reply, I was busy during this week. It worked with very poor and noi...
Asterisk SIP Channels Bridge (2 Replies)
Asterisk — 119406966403 Nov 2007*
Hello everyone, I'm trying to bridge 2 SIP channels together via AGI script. The AGI Script is written in C#. I'm able to get the unique name ...
CSB — 119407625503 Nov 2007
I wish to implement a jajah type service using Asterisk to call two numbers and join them together. I have seen various click-to-dial scripts and have...
one way RTP using NAT (1 Reply)
Hector Quiroz — 119407824703 Nov 2007*
Hi, I'm having a problem with my asterisk, trying to connect to a CISCO 2840 IOS12.x ASterisk is behind firewall NATing, when it do the handshakin...
Polycom Park Button (2 Replies)
Kelly Opal — 119409269303 Nov 2007*
Hi I have a Polycom 501 phone. I set the park feature to 1 in sip.cfg and the button shows up just fine. However when you press it it does nothing. I ...
Spam Filter News (3 Replies)
Thomas Kenyon — 119409338603 Nov 2007*
Is there any news on getting the Spam Filter fixed for this mailing list?...
Jitterbuffer issues (1 Reply)
Tony Plack — 119409478903 Nov 2007*
...
Mike Dent — 119409543703 Nov 2007
Hi, I've had a Snom 300 connected to my Asterisk box at home for 12 months or so now. Recently it lost all its settings and I had to reconfigure i...
Lyle Giese — 119409707803 Nov 2007
The orginal did not make it to the list... Spam filter issue??? No repeat of the lockup yet. Lyle Original Message Subject: voicemail locked up Asteri...
Jerry Geis — 119409872603 Nov 2007
I have a grandstream 488 using FXO port. With asterisk 1.2.23 When I have DTMF mode set to RFC2833 (asterisk and grandstream) and I use a call file an...
OpenSER for Asterisk Load balance (2 Replies)
Edgar Guadamuz — 119410937503 Nov 2007*
Hi, guys I?ve just seen thta OpenSER can be coupled with Asterisk for load balance, with the dispatcher module, something like this: dispatcher.cfg fi...
Dovid B — 119410990603 Nov 2007
Hi, I have a UTStarcom F3000. For some reason it won't connect to my Netgear WPNT834 Range Max router ? Anyone know of a fix or new firmware updat...
Dovid B — 119412031303 Nov 2007
Hi, I have a UTStarcom F3000. For some reason it won't connect to my Netgear WPNT834 Range Max router ? Anyone know of a fix or new firmware updat...
Asterisk versions and H323 (1 Reply)
bilal ghayyad — 119413144803 Nov 2007*
Hi List; Is there an Asterisk version that contains H323 module, or still I have to download the h323 alone and compile it? Regards Bilal...
Asterisk as a SIP to SIP Gateway (2 Replies)
Tomasz Zieleniewski — 119416633604 Nov 2007*
Hi, It is my second time when I try to use asterisk :) I am starting with the following issue. I want asterisk to behave as a gateway between two sip ...
Off-Topic: add GSM codec to X-Lite (4 Replies)
Alejandro Cabrera Obed — 119419363504 Nov 2007*
Dear all, sorry for the Off-Topic but I have an Astreisk 14 voip server connected to Twinkle and X-Lite clients. I have to use the GSM codec for all o...
Jean-Yves Avenard — 119420199404 Nov 2007*
Dear all I am trying to upgrade our asterisk from 1.2 to 1.4.x There is something that now fails to work, reading the various documentations, I can no...
Restart when convenient (5 Replies)
Doug Lytle — 119420953604 Nov 2007*
I've moved 1 of our facilities over from 1.2 to 1.4 two weeks back. So far, the only issue that I've encounted is. I have a scheduled CRON job...
Frank Church — 119422562105 Nov 2007*
Are there ATAs that allow different phone numbers from one network connection? Such as supporting multiple IP addresses so that each RJ11 has a differ...
Fax Problems with SpanDSP (25 Replies)
Christian Peter — 119423208205 Nov 2007*
Hi list, I'm running current SpanDSP http://www.soft-switch.org/downloads/spandsp/... with Asterisk 1.2.22 somewhat successfully. Most Fax machine...
Rilawich Ango — 119424496405 Nov 2007
Hi all, I have seen a lot of message talking about asterisk crashed when using queue and mixmonitor together. I do use both in our system and also get...
Chris Nestrud — 119424525005 Nov 2007
I am using the Debian package of asterisk which is version 1:1.4.13~dfsg-1. I have created an AGI script which plays files in response to external eve...
asterisk-users Digest, Vol 40, Issue 5 (1 Reply)
Bincy K. Philip — 119425864905 Nov 2007*
Hello Thanks for the reply.. I could use Asterisk as SIP server and establish call using two SIP phones. But I need H323 support also. For that I have...
Bincy K. Philip — 119425987605 Nov 2007
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: Bincy K. Philip Subject: R...
Tomasz Zieleniewski — 119425992405 Nov 2007
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 INVIT...
Dynamic Queue Members - Auto Logoff (2 Replies)
Nick Brown — 119427237505 Nov 2007*
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 ab...
bilal ghayyad — 119427689605 Nov 2007
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...