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Asterisk Users Mailing List - Non-Commercial Discussion
July 2011 - page 1
Olivier — 130949848201 Jul 2011
Hi, I would be very curious to know your opinion to both questions bellow. 1. As linux kernel simplified its naming recently, what do you think of tod...
Steven Howes — 130951495301 Jul 2011
Hi All, Asterisk 1.6.2.19 was released on the 28th, does anyone know if there a timescale for this reaching the RPM repository? We're badly affect...
Problem with detecting fax on PRI/DAHDI channels (10 Replies)
Sawan Vithlani — 130952175801 Jul 2011*
Hello all, I am using call files to dial out to a set of PSTN numbers. The calls are going out fine and being handled correctly by the dial plan. The ...
Olivier — 130952383001 Jul 2011*
Hi, Among TFTP, FTP, HTTP and others, which protocol would you select to provision Polycom phones. At the moment, I'm using TFTP but I'm wonde...
Bryant Zimmerman — 130952770701 Jul 2011
Hey all I am looking at some products by a company called allo I am wondering if anyone has had any experience with any of their items, phones, ata, t...
Bryant Zimmerman — 130952877001 Jul 2011
...
Javaid ITEL — 130953306101 Jul 2011
Hi All, I am doing a project in which we will have, say 5 participants in a conference talking to each other. Now suppose if a participant wants to do...
Ezequiel Lovelle — 130954223301 Jul 2011
Hi, I have a ivr, and I need to make a beep sound playback after phone when to dial sip DIALSTATUS} = $ {ANSWER example 1234,1,Answer() 1234,n,Dial(SI...
How to without GUI (4 Replies)
Danny Nicholas — 130954498201 Jul 2011*
Hey gang, I've got a CISCO SPA3102 that I want to set up. My environment is not favorable for using the Asterisk GUI interface - does anybody have...
bilal ghayyad — 130954685401 Jul 2011
Hi All; I am running Asterisk version 1.8.4 and I need to know if I am going to use it as a call center, and I have up to 6 E1s and about 150 Agents r...
Alex Vishnev — 130954903501 Jul 2011
Hello I have a small call center with about 7 queues. all agents are dynamic and they login to each queue via a dialplan. When you perform queue show ...
calleridname presentation Asterisk => Siemens (3 Replies)
Rafael dos Santos Saraiva — 130954951301 Jul 2011*
Hi I interconnect the Asterisk and the Siemens PBX with Pri QSIG. But i can't show the callerid name in the way Asterisk ==> Siemens. I realize...
Alex Vishnev — 130955105901 Jul 2011
Does anyone know why i would get this RINGNOANSWER events in queue_log when clearly the agent is busy and call-waiting is disabled. 1309550595...
error in GUI access (2 Replies)
asterisk — 130955968801 Jul 2011*
I have this error after upgrading to 1.8.4.4 on my centos 5.6 32it When using GUI to access, I got this error *** glibc detected *** /usr/sbin/asteris...
Akramul Hossain — 130956717402 Jul 2011
Dear sir, thanks your mail, before more time i request you active your list to view asterisk-user list, what is facilities, what kind of work i do the...
Kaushal Shriyan — 130961464502 Jul 2011*
Hi Please help me understand about the below issue ? [root@asterisk1 ~]# /etc/init.d/asterisk restart Stopping safe_asterisk: [ OK ] Shutting down ast...
Load Balance Trunks (5 Replies)
Abid Saleem — 130962744602 Jul 2011*
Hi All, I have 100 Trunks from my Provider. My Provider is restricting me to make only 120 minutes Call duration / trunk / day. So I want to load bala...
bilal ghayyad — 130962792802 Jul 2011*
Hi All; To be able to distribute the incoming calls on a group of extensions, is there huntgroup in Asterisk? Or what I have to use? I need first call...
chanspy spies on wrong channel (1 Reply)
steve casto — 130965962303 Jul 2011*
asterisk 1.4.32 have zapata.conf soft link to chan_dahdi.conf to use flash operator panel < 2.0 (from extensions.conf) exten=> 304,1,ChanSpy(Zap...
cnasterisk — 130976635104 Jul 2011*
Hi all, I have a server runing asterisk 1.8, and the server has 2 different ip address if i want to make a call from a sip trunk with a fixed ip from ...
Mixmonitor concept's question (2 Replies)
virendra bhati — 130978878504 Jul 2011*
[RecordPrompts] exten => 1111,1,Answer() exten => 1111,n,NoOp(WelCome to conference section) exten => 1111,n,Playback(ConfDemoWC) exten =...
stream rtp from asterisk (3 Replies)
Marcus Kvarsell — 130979135004 Jul 2011*
Hi! Anybody familiar with streaming rtp from asterisk. Preferably with the xorcom asterisk patch which streams rtp from asterisk to oreka audio server...
Alex Vishnev — 130979830404 Jul 2011
Does anyone know why i would get this RINGNOANSWER events in queue_log when clearly the agent is busy and call-waiting is disabled. 1309550595...
bilal ghayyad — 130982309004 Jul 2011*
Hi All; We know that agents can login and logout from the phone handset. But if we need the login, logout, ready and not ready to be from an applicati...
Testing Asterisk with media - sipp (2 Replies)
Daniel - Asterisk — 130983165005 Jul 2011*
I'm trying to get working SIPp with media but something is wrong (it's working well without media), please help: This is the command I send at...
James Lamanna — 130983883005 Jul 2011
Hi, I'm looking for some advice on how to solve DTMF issues. I have 2 boxes, one which is the connection to the PSTN (PSTN) through PRIs and SIP t...
Deka, Rajib IN MAA SL — 130984843905 Jul 2011
Hello all, I found a problem regarding SIP presence in asterisk (1.6.2). The scenario is not working properly for all users. Our SIP client sends SIP:...
More SQL Querys in dialplan (2 Replies)
Ulrich Meckel — 130986007405 Jul 2011*
Hi List I tried to use SQL Query in my diaplan. If i only use one or two there is no Problem but if i try to start the third one after the other it ha...
Deka, Rajib IN MAA SL — 130986398205 Jul 2011
Hi All, Following message I got in console for an extension, [Jul 5 12:10:56] VERBOSE[3729] chan_sip.c: <--- SIP read from UDP:132.186.230.70:7510 ...
Kristijan Vrban — 130986855305 Jul 2011
Hello is use realtime sip-peers in 1.8, and have the problem, that when a peer is loaded from database, the devstate is "AST_DEVICE_UNAVAILABLE...
Administrator TOOTAI — 130987798405 Jul 2011
Hi, we have 4 asterisk, versions are 1.4.35 1.4.36 1.6.2.18 and 1.4.42 One GrandStream GXV3000 is used for the tests. He is registered to asterisk 1.6...
Lee Archer — 130987966705 Jul 2011
Hi all, can someone explain what siphistory is supposed to do as it appears to record nothing to my log files. When I sip show history <callid...
steve casto — 130988339305 Jul 2011
The argument to chanspy is a pattern and not an exact match. -- Jim Dickenson mailto:dickenson at cfmc.com CfMC http://www.cfmc.com/Zap/4-1Was not awa...
realm question (2 Replies)
Hans Witvliet — 130988611005 Jul 2011*
Hi all, Trying to find where i got wrong in my config.... Is the "realm" parameter in sip.conf only used for possible autentication? The thi...
Olivier — 130988804205 Jul 2011
Hi, Using Polycom's Master configuration file, I could not find any convenient way to store 2 different versions of the same localization file on ...
Asterisk 1.6.1 Realtime SIP Users (3 Replies)
Mickael MONSIEUR — 130989518505 Jul 2011*
Hello, I just implement the SIP Peers with MySQL. In the structure mySQL missing the following fields: nat = yes notransfer = yes dtmfmode = rfc2833 c...
Couldn't call Agent and segfault (1 Reply)
Agustina Berretta — 130993098406 Jul 2011*
Hi folks! I´m having the following problem: I get the following messages, asterisk get automatically reloaded and agents log out once or twice a day,...
Tobias Steen — 130993699406 Jul 2011*
It seems that the full distro package from FreePBX with Asterisk 1.8.1.4 someway hides (deletes?) the source directory for asterisk after installation...
Lee Archer — 130994836306 Jul 2011
Hi, can anyone help with this? Thanks Lee From: [mailto:] On Behalf Of Lee Archer Sent: 05 July 2011 16:27 To: Subject: [asterisk-users] Recording SIP...
A E [Gmail] — 130995719206 Jul 2011*
hello people, I am running v1.8.4.2 on debian squeeze on a sparc platform...and for some reason I have noticed that only after a few test calls, the a...
Cassius Smith — 130995904606 Jul 2011*
Hello all I'm running Asterisk 1.8.4.4 in a new installation. I'm seeing peculiar behaviour in a context where I dispatch to different MeetMe ...
Asterisk Development Team — 130997227906 Jul 2011
The Asterisk Development Team announces the release of libpri version 1.4.12. This release is available for immediate download at http://downloads.ast...
Agents outbound calls to be recorded (2 Replies)
bilal ghayyad — 130999132906 Jul 2011*
Hi All; I know that incoming calls for the agent can be recorded, but how I can let the outbound calls for the agents to be recorded? I can determine ...
Blind Transfer Connected (7 Replies)
Nikhil — 130999163006 Jul 2011*
Hi all In asterisk if blind transfer failed ,call is not connecting back . For Eg: A make call to B through asterisk,then B transfer the call to C. If...
bilal ghayyad — 131001338207 Jul 2011*
Hi All; The asterisk version I am using is 1.8.4.2 and I compiled ooh323 channel (by selecting the add-on). But really does not work in good performan...
Thomas Hoellriegel — 131004680107 Jul 2011
Hi all, Probably my message disappeared. I updated from 1.4.42 to 1.8.4.4 and 1.8.5-rc1. The problem is the same: I generate a callfile with the optio...
Bruce B — 131005466007 Jul 2011
Hi everyone, Occasionally (with no set pattern), I get *"SIP/2.0 407 Proxy Authentication Required" *from iCall when trying to termiate to t...
Dropping Conference calls (7 Replies)
Mark Rosedale — 131005939407 Jul 2011*
Hello, I'm using Asterisk 1.8-svn branch. I'm having an issue with dropped outbound calls, particularly outbound conference calls (conference ...
Bruce B — 131006789007 Jul 2011
Hi everyone, I just lunched a CentOS VM in Proxmox and used the Digium repository to install Asterisk using "yum install asterisk16"...and i...
Bryant Zimmerman — 131006966707 Jul 2011
Here is a simple way to strip the '-' Here is a concept solution. I have not tested the code so there may be some syntax errors. It can work a...