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Asterisk Users Mailing List - Non-Commercial Discussion
May 2010 - page 1
No change in payload. (SDP) (2 Replies)
Aditya Kumar — 127267538301 May 2010*
re-posting the question. --- use case: when some one in my pbx calls 100.200, I have translations well defined, Media also (media via asterisk) --Work...
Portech MV-374 does not register (14 Replies)
Jonas Kellens — 127279048702 May 2010*
...
Calls Dropping (3 Replies)
Dan Journo — 127280535202 May 2010*
Hi, I'm having a major problem with random calls dropping. After spending weeks trying to figure it out, i've finally spotted the issue but do...
Security tests (5 Replies)
Daniel Bareiro — 127280774202 May 2010*
Hi all! In the network of my house I was testing the security with my Asterisk installation. The first test that I'm doing is an man in the middle...
ATA shootout: PAP2T versus Grandstream Handytone 286 (12 Replies)
David Backeberg — 127280881202 May 2010*
I'm considering a situation where I buy about twenty ATA devices. I've played with the Linksys / Cisco PAP2T, and got that working fine with s...
AGI <==> DeadAGI (8 Replies)
Redouane Zerargui — 127285180503 May 2010*
Hello, i have this problem : i phone person B . *if i hang up*, i have this "h" extension : exten => h,1,AGI(ende.agi) *if the person B h...
Asterisk stopping for no reason (3 Replies)
Alexandre Vézina — 127286662603 May 2010*
Hi, Few days ago, my asterisk began to stop unexpectedly What I did: - Added a mp3 to the musiconhold directory - Adjusted the permissions (chown aste...
Calling a RESTful Web service from Dialplan???? (2 Replies)
DHAVAL INDRODIYA — 127287066503 May 2010*
Dear All, Last Week i tried and goggling more on how to call RESTful webservice from Dialplan? i found *CURL* function but while i tried to use it ,it...
matthieu Nicaise — 127287725503 May 2010
Hi everybody, I have a problem using parking for outgoing call. A is an local sip phone. A is using the local extension : [local] exten => _XXX.,1,...
B400P card crashes conncection (1 Reply)
Rudi Oosthuizen — 127287987003 May 2010*
Had a similar problem with a B410p BRI card. Had to enable (or disable) the 100ohms termination jumper on the card, because the telco provider uses di...
Shariq Khan — 127288106703 May 2010
Is there any way, i can detect in asterisk that which party hanged up the call either from A side or B. Both parties are using SIP protocol. I am usin...
Torintino T — 127289296503 May 2010
someone has Asterisk 1.2 (upgrade is not possible), and wants to spy on specific extensions he can specify while dialing a code, could you please kind...
frangky robert — 127289351003 May 2010
Hi all... I'm sorry for repeating my message.I have a problem with caller id on my asterisk server with xorcom astribank. here is my configuration...
RTP ports (4 Replies)
voip crazy — 127289735603 May 2010*
Hello, I need to limit the RTP ports used by an asterisk in a client, Actualy the range defined is from 10000 to 20000 udp ports. If I only have 10 lo...
Run a script after Page application (1 Reply)
Andy Swing — 127291874503 May 2010*
I am trying to run a script before and after the Page application in order to mute/un-mute my whole house audio when my phones are being used as an in...
François BERGANZ — 127292696203 May 2010*
Hello, I saw that Asterisk don't calcultate fine the ANSWEREDTIME. I want that when ANSWEREDTIME =~ 5.6 become 6 and if =~10.3 become 10 because, ...
Eddie Mikell — 127295147404 May 2010*
All: My company has an existing ESI IVX E-class system with 45 phones. I can add one more card, to expand it another 6 phones, but it's $8000, and...
Asterisk and Patton (8 Replies)
A.Santoro — 127295844604 May 2010*
Hi, we have and Asterisk server connected to a Patton Smartnode 4638 with 4 BRI. We configured 4 SIP account on Patton (1001, 1002, 1003, 1004). The s...
Jonas Kellens — 127296090004 May 2010
Hello list, I need to set Voice QoS and SIP QoS for YeaLink. The possible values are 0 ~ 63. With Grandstream I can fill in DiffServ 46, which is EF. ...
Reading the CDR (5 Replies)
Dan Journo — 127296142604 May 2010*
Hi, I am diverting an incoming call to a mobile phone and a landline using the following:- exten => 0203000000,3,Dial(SIP/442080000000 at sipprovid...
Lenz Emilitri — 127296150004 May 2010
Hello list, I was wondering if there is a way to see if a given piece of dialplan is loaded through AMI. I have seen the GetConfig command, but it see...
Leo Burd — 127298050304 May 2010
Hello there, How to retrieve the "failure reason" when calling AMI Originate with Async = 0? The system seems to return the following no mat...
Vieri — 127299023704 May 2010
Hi, ZAP/DAHDI extension 3210 calls an Asterisk queue 4050 with one SIP agent 4053 added via ->QueueAdd("4050", "Local/4053 at from-i...
Interesting email project. (6 Replies)
mike mosier — 127299424104 May 2010*
Hey all. My boss asked me to implement the following When DID 713xxxxxxx is dialed send an email to mmosier at xxx.com. with the time date and CID inc...
Eddie Mikell — 127299580204 May 2010
All, Thanks for the suggestions, but the system is a plan non-sip, non-ip, non pri setup. It's pretty much a closed box setup. And the prices for ...
Asterisk Development Team — 127299584104 May 2010
The Asterisk Development Team has announced the release of Asterisk 1.4.31. This release is available for immediate download at http://downloads.aster...
Asterisk Development Team — 127299587604 May 2010
The Asterisk Development Team has announced the release of Asterisk 1.6.0.27. This release is available for immediate download at http://downloads.ast...
Asterisk Development Team — 127299590804 May 2010
The Asterisk Development Team has announced the release of Asterisk 1.6.1.19. This release is available for immediate download at http://downloads.ast...
sending T.38 fax negotiation problem (6 Replies)
Ilmars Knipshis — 127300288704 May 2010*
Hi there. I have the similar problem ("Digium fax - sending fax call file vs manager originate") sending faxes with Asterisk 1.6.2.6 and Dig...
Productivity Suite on Polycom IP7000 (2 Replies)
Karl Fife — 127301728304 May 2010*
Has anyone here ever actually truly successfully gotten a Polycom IP7000 to take a productivity suite license and enabled the bonus features like 4-wa...
Renato bianchini — 127303019305 May 2010
Hi Anyone, I have a server with asterisk 1.6.2.1 working in Realtime with PostgreSQL, but I'm having problems when happened any error in a table, ...
Julian Lyndon-Smith — 127304034905 May 2010
We have a need for up to a dozen UK mobile numbers to be forwarded to a UK landline. I know that I can just forward them, but was wondering if anyone ...
François BERGANZ — 127304415605 May 2010
Hello, I saw that Asterisk don't calcultate fine the ANSWEREDTIME for me. I want that when ANSWEREDTIME =~ 5.6 become 6 and if ANSWEREDTIME= 10.3 ...
Confirm answering a call (1 Reply)
Mark Scholten — 127306043405 May 2010*
Hello, I am working on getting the following to work and I couldn't find it in the documentation I did read. Where should I look or does someone h...
HDLC Receiver overrun on Wildcard TE410P (1 Reply)
Łukasz Krzyżak — 127306113405 May 2010*
Hello I've got small PBX (30 simultaneous connections) based on asterisk (1.6.2.6), which uses Stargate 2N ISDN to GSM gate. It runs ok for day or...
Sebastian Denz — 127306907505 May 2010
Hello list, as I am trying to write a complex macro for my users i have the problem, that the appdata field in the extensions table is to small for al...
run script after completed (6 Replies)
Necati Demir — 127308149405 May 2010*
Hello, I am creating a call file with parameter "Archive: yes". When it is completed it is moved to directory outgoing_done. It works. Now i...
Miguel Amez — 127309340905 May 2010
Hi list! I have this configuration for sending T38 faxes to my T38 fax termination provider: T38modem --> hylafax --> Asterisk-SIP-Extension --...
sean darcy — 127309949205 May 2010
I've got two 1.6.2 asterisk boxes. I'd like to be able to set up two separate sip connections. But when I try that I get: chan_sip.c:12671 che...
Registering a Cisco 7965 on 1.4.26 (4 Replies)
James Lamanna — 127310010705 May 2010*
Hi, I'm having a problem trying to get a Cisco 7965 phone registered on Asterisk 1.4.26. As we know, Cisco now, for security reasons, has made the...
Jesse Cloutier — 127315080906 May 2010
Hi everyone, I am trying to figure out the behavior of trustrpid Basically its not behaving the way I expected it to or maybe I am missing a configura...
Jason Walker — 127315345306 May 2010
I am trying to change a 1.6 realtime statement into a 1.2 realtime statement and I know much has changed. I wish I could just upgrade, but alas not ri...
Make the call finish after executing Dial(G()) (2 Replies)
Motiejus Jakštys — 127315476006 May 2010*
Dear List, My Dial command: exten => _X.,n,Dial(SIP/PBX2/1234,60,G(connect-jack^${EXTEN}^1)) exten => h,1,.... [connect-jack] exten => _X.,1,...
SIP - SIP over PBX no audio when canreinvite=no (4 Replies)
RG — 127316122406 May 2010*
Hello list, I am trying to solve a problem and after unsucessfully chasing forums and google for some hours, I turn to you in hope of a solution. I fe...
OT: NAT in SPA922 (6 Replies)
Sebastian Milioto — 127316625306 May 2010*
Hi all, I've just bought some SPA922. First time with this hardware for me. I see no LAN tab in its web GUI where I can setup NAT for PC conected ...
Alexandre Rodrigues — 127316669106 May 2010*
Hi all, I am trying to connect to a softphone application using an Iax channel on Asterisk 1.4.30. I can do outbound calls, from softphone to asterisk...
Asterisk 1.6.2.7 Now Available (7 Replies)
Asterisk Development Team — 127316725206 May 2010*
The Asterisk Development Team has announced the release of Asterisk 1.6.2.7. This release is available for immediate download at http://downloads.aste...
David Backeberg — 127316997106 May 2010*
I've just upgraded to 1.6.2.6 on one of my test systems. I started out happy, with some improvements in transfers to Local() channels from a SIP c...
Tom Browning — 127318860806 May 2010
I'm migrating an application running on a fairly old 1.4 (or 1.2?) version of Asterisk to some boxes running 1.6.0.27 The application takes an inb...
Channels In Use (2 Replies)
dotnetdub — 127318875906 May 2010*
Hi List, If we have a scenario where a customer is using a telephone and their WAN link goes down for example the channel in asterisk stays marked as ...