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Asterisk Users Mailing List - Non-Commercial Discussion
January 2010 - page 1
Joseph — 126231224801 Jan 2010
When I configured AudioCodes MP-114 to MWI it keeps complaining bout subscription without mailbox: chan_sip.c:15450 handle_request_subscribe: Received...
Patterson — 126235384801 Jan 2010
An great an happy new year 2010 for all of you ......
Monitoring SIP & Skype connections (1 Reply)
Myles Wakeham — 126235592801 Jan 2010*
I have an Asterisk 1.4.2 server with 3 different SIP providers and Asterisk for Skype gateway installed. Periodically the SIP providers go offline for...
Joseph — 126238304301 Jan 2010
I have AudioCodes 2xFXO / 2xFXS but can not make the FXO port to work correctly; I can dial out on one FXO port or the other FXO, but not on both. It ...
PBX Extension Help (6 Replies)
Saeed Akhtar — 126238595001 Jan 2010*
hi all, I have a little problem. I'm trying to configure a2billing (asterisk2billing) with asterisk. Everything done successfully but when I try t...
Yuval Yogev — 126245047102 Jan 2010*
I installed an Elastix based system and changed it to work in Device-Mode since there is a call center and users has to login. As requested, I made ...
Yuval Yogev — 126246368602 Jan 2010
Dear Tzafrir, There are about a few hundreds of calls every day, so we get a few thousands of files (corrupted files ..) Please instruct me what to do...
Help getting info from caller (7 Replies)
Landy — 126250877303 Jan 2010*
Hello. Happy New Year to everyone. I have a small WISP and would like to have customers to call our number to check their balance. I am planning on wr...
identifying channel for softhangup (12 Replies)
Jeremy Kister — 126251422803 Jan 2010*
When I place an outbound call from asterisk 1.6.1.12 to a FXO port on my Cisco 1760V 12.4, the channel changes - seemingly incrementing: e.g., in the ...
verifying correct loading of VPMADT032 (9 Replies)
ramadasan at amachu.net — 126251520103 Jan 2010*
Hi, I have two Digium Cards http://www.digium.com/en/products/digital/te1... on connected to PRI & the other to EPABX. We have felt problem simila...
hadi motamedi — 126252025603 Jan 2010
Dear All Can you please give me guidelines and the link to join Asterisk real time chat to have your online technical support? Thank you...
SIP Listen Multiple Ports (5 Replies)
Shariq Khan — 126255082103 Jan 2010*
Is there any way to listen SIP on multiple ports on asterisk. Is is possible to define in sip.conf in the following way. sip.conf [general] port = 506...
J Smith Thomas — 126255292703 Jan 2010
"asterisk-users-request at lists.digium.com" wrote: Send asterisk-users mailing list submissions to asterisk-users at lists.digium.com To su...
Joseph L. Casale — 126257479704 Jan 2010
Not sure how to go about troubleshooting this, did a fresh install of CentOS 5.4x86 with a netinstall iso off the base and update repo followed by a i...
Giedrius Augys — 126258540504 Jan 2010
Hello, Firstable, happy new year everybody! I have couple problems with asterisk 1.6.0.5 (meetme and ztdummy - kernel panic). And I want upgrade to ne...
lesly dorval — 126262070604 Jan 2010
[ArchiveOrange]: There doesn't seem to be anything here...
Script to show asterisk stuff (2 Replies)
Tiago Geada — 126262628204 Jan 2010*
Hello folks. I'm looking into having a web page displaying asterisk callers. We are a call centre, and having operators answering calls at home or...
hin lee — 126262785204 Jan 2010
I have posted my problem on the link below, but didn't get any answer. I am hoping someone here can help me with this issue. Here's my problem...
Joseph — 126263089704 Jan 2010
How to register/configure Sip accounts to register per gateway? All the accounts I have are registered individually but with mix (FXO/FXS) AudioCodes ...
shrikant.soni at globussoft.com — 126263578904 Jan 2010
I am implementing one dialer type of application. In which i am first dialing one source number and sending it to conference and then starting dialing...
Realtime Queue Members Not Ringing (1 Reply)
Robert Broyles — 126263797804 Jan 2010*
Hi, So I'm using Asterisk Realtime Queues and Queue members on 1.4.28. I've noticed if there are no people in the queue when a call enters, ev...
Qurba Joog — 126263841604 Jan 2010*
Hello, Please forgive me if I'm repeating this post. I have searched and looked for similar problem with a solution but have not see a similar one...
Jerry Geis — 126263900304 Jan 2010
Is there a way to not compile in lpc10 support using the ./configure command? ./configure --disable-lpc10 or something like that? if that is not avail...
cheap ip phone with auto-answer (3 Replies)
Leif Neland — 126264358704 Jan 2010*
I want some cheap ip-phones with auto-answer, to work as paging system at dinnertime. Options, please. Leif...
hadi motamedi — 126266722405 Jan 2010
Dear All Further to my previous inquiry regarding Asterisk sending dialed digits in one-by-one digit format when we had ISDN PRI link with the PSTN sw...
shameem Banu — 126267124305 Jan 2010
Hi , Can any one tell me that how to automatically dial a list of numbers from database .I have seen a methodology in the post but am not clear vth th...
DNS reload on trunks for outgoing calls (3 Replies)
Remco Barendse — 126268180905 Jan 2010*
Is there any fix or workaround for the DNS problem (old standing bug that when the box starts and domain names do not resolve quickly enough from DNS ...
Leif Neland — 126268383605 Jan 2010
It seems dahdi is needed for meetme, but not available under FreeBSD. So what do I do then? Asterisk has only SIP-channels....
Daniel Stefanus — 126268431105 Jan 2010
Hi, I have a difficulty on my Asterisk's database.How can I get the info about list of ringing agents on my queue In console : -- Started music on...
Oscar Atienza — 126268462805 Jan 2010
Hi, That model HP or Dell server that I recommend for a TE412P card for about 200 users? Thank you very much....
ZapRAS priviledge error (5 Replies)
Will Payne — 126268877405 Jan 2010*
Hi, I'm trying to get ZapRAS working but not getting very far.. Asterisk CLI shows: WARNING[3355]: app_zapras.c:173 run_ras: wait4 returned -1: No...
phiroc at free.fr — 126270099905 Jan 2010
Hello, can the Asterisk API be used to automate a MITEL 5330 telephone? If not, are there any other API which can used to do that? Many thanks. phiroc...
Dialplans & Holiday Dates (3 Replies)
Myles Wakeham — 126270183505 Jan 2010*
I have a working dialplan for our phone system with Mon-Fri, business hours identification, etc. But what I'm lacking right now is support for com...
Inquiry:Asterisk different codec schemes? (8 Replies)
hadi motamedi — 126270997905 Jan 2010*
Dear All Can you please let me know if we can have different codec schemes for audio codec in & audio codec out ? I mean , in one application , we...
T.38 ITSP? (2 Replies)
Karl Fife — 126272111305 Jan 2010*
Has anyone found an ITSP that will relay T.38 fax to an asterisk 1.6.x instance AND do it reliably? If so, I can think of a number of locations with c...
achris at abacus6.net — 126272766205 Jan 2010
Hi, I have installed Asterisk with iaxmodem to send faxes with Hylafax. But I have problems to send some faxes because the receiver does not accept ...
Canadian call quality issue (3 Replies)
Max McGraw — 126272858805 Jan 2010*
hello, we have been using a couple of US based VoIP providers for outbound calls completed within the US, without any issues. We recently started maki...
AGI and embargeability (4 Replies)
Quinn Weaver — 126273087705 Jan 2010*
Hi, This is a naive question, but is there a way in my AGI script to simultaneously play audio and listen for DTMF or voice responses? I've heard ...
Matthew Edmondson — 126274737906 Jan 2010
Hi all, I an using the Originate() dialplan command but I cant get it to save cdr's. Here is the line I am using: exten => _61XXXXXXXXX,53,Orig...
phiroc at free.fr — 126277057906 Jan 2010
Hello, can any of the Asterisk API be used to dial a number on MITEL telephones? Many thanks. phiroc...
Fastagi-mapping problem (2 Replies)
ahmed magdy — 126277685906 Jan 2010*
Hello I am new in Asterisk Java, i am working on Asterisk 1.6.2.0 , i started the first example Hello AGI in this tutorial http://asterisk-java.org/de...
mosleh at infolog.mr — 126277821606 Jan 2010
Hi all, I need Help. I want to compile zaptel in data mode but i got this errors: /usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c: In function ?zt_xmit?...
Inquiry:How to define incoming route for sip? (2 Replies)
hadi motamedi — 126277985506 Jan 2010*
Dear All Can you please let me know how can I define incoming route to accept incoming calls from an external sip server? I have defined the following...
Outgoing Calls Only -- Firewall Rules (4 Replies)
Nicholas Blasgen — 126278524106 Jan 2010*
I'm trying to move my Asterisk deployments under a Virtual IP address and now remember why I dislike this. My primary Asterisk system is now behin...
Will Szopko — 126280532106 Jan 2010
We have recently pulled an ancient Fujitsu-branded Centigram voicemail system out of production use and replaced it with an Asterisk box, which is now...
Shane Brath — 126280654506 Jan 2010*
Folks, I have a Merlin Legend R7 V10.0 with a 2 100D cards. I have 1 card in slot 4 going to CenturyTel, and the card in slot 10 going to a flip cable...
question on makefile (3 Replies)
Jerry Geis — 126281233106 Jan 2010*
There is a line like in codes/Makefile $(if $(filter codec_lpc10,$(EMBEDDED_MODS)),modules.link,codec_lpc10.so): $(LIBLPC10) What is filter? Where is ...
Tiago Geada — 126281272806 Jan 2010
Hi We have an operator that his device state on all queues is "In use" where it should be "Not in use". how can we manually change...
Olivier — 126281835606 Jan 2010
Hi, I need to install (within the next couple of hours) a 1.6.1.11 server with a Digium B410P board. One of this system's DID is dedicated to Fax ...
Jerry Geis — 126282645607 Jan 2010
I download the x-lite software for windows. Put it on two laptops. Using asterisk 1.4.28 set videosupport=yes in sip.conf general. disallow=all allow=...