search for: voipa

hello, I've callcenter and our queue members want to see on their IP phone's display queue's name , from which incoming call was originated, for example "<client's_number> -> Sales". This problem appears when one member can belong to couple queues. Work around would be setting calling name with such information. Maybe there is another way (setting SIP

Hello, I have asterisk 1.6.0.20 and Is it possible to add Reason header on Hangup: Reason: q.850;cause=17 Thanks -- Best Regards, Giedrius -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100221/d29c02b8/attachment.htm

...By looking in the zaptel code, this error value (0x00103017) means cOCT6100_ERR_OPEN_EXTERNAL_MEM_BIST_FAILED. Is anyone familiar with that problem ? Thanks for your help. --- TE212P card: jumpers are set to E1 mode and nothing is connected to that card at the moment. # uname -a Linux ditti-voipa-serv-1 2.6.18-4-amd64 #1 SMP Fri May 4 00:37:33 UTC 2007 x86_64 GNU/Linux # cat /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 span=2,1,0,ccs,hdb3,crc4 bchan=32-46 dchan=47 bchan=48-62 # cat /proc/interrupts CPU0 CPU1 0: 42385 0 IO-API...

Hi, I have cable modems Arris with MGCP protocol. And I need PacketCable NCS patch for Asterisk. http://asterisk.urtho.net/ doesn't work! -- Pagarbiai, Giedrius Augys Siauliu Universitetas, IST IP telefonijos inzinierius Tel. 8 41 590408 Mob. Tel. 8 678 05790 el. pastas voipas@gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060614/27aef641/attachment.htm

Hi, I have problem installing spandsp-0.0.3pre24 on FreeBSD 6.1. I get error: configure: error: "Can't build without libtiff" . But I have installed tiff from port tiff-3.8.2. I understand that the problem is about libtiff, and spandsp can't find these libs. So how to fix the problem? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am testing Nokia E60 with Asterisk. And I noticed that if another side is busy, nokia is still calling (I hear alerting), it do not show that another side is busy. Maybe somebody has noticed the same problem too adnd solved this one. I made the same tests with Xlite and don't have any problems like nokia. Please help me -------------- next part -------------- An HTML attachment was

Hello, I want to create php rate script and I'm using Deadagi. But I allways get billsec 0 , or nothing. Can you help me to solve this problem... My extension.conf: exten => _123,1,DeadAgi(rate.php) exten => _123,2,hangup And my simple test php script rate.php #!/usr/local/bin/php -q <?php include_once (dirname(__FILE__)."/phpagi.php"); $AGI = new AGI();

Hello, I've asterisk 1.4.22. I need to that the first conference user hears "You're the only conference user..." . When the second user joins (without recording his name) , the first user only hears "new user have join" , when the third user joins to conference, others hear "new user have join" and so on. I'll try to do this with meetme, but it always

Hello, Is it possible, that during the call one side , for examples clicks the button on the web, and this call starts recording? It's possible with asterisk feature automon and DTMF. So it is possible to start recording the channel using AMI or ... ? Thanks -- Pagarbiai / Best Regards, Giedrius Augys -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I've a problem. I've asterisk 1.6.0.5 version. And I've created callcenter, but agents registers to another SIP server. When agent tries transfer a client to another operator , pressing flash, I get this: [Aug 18 16:06:37] WARNING[5259]: chan_sip.c:5349 sip_indicate: Don't know how to indicate condition 9 [Aug 18 16:06:37] WARNING[5259]: channel.c:2858 ast_indicate_data:

Hello, I have one problem and I need to disable dtmf (disable rfc2833, info and inband) on one (other peers must support dtmf) SIP peer . Is it possible? Workaround would be use g729 codec with dtmfmode=inband. Maybe there is better solution? Thanks for help. -- Pagarbiai / Best Regards, Giedrius Augys -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, Is it possible use asterisk curl function with ssl sertificate? Thanks -- Best Regards, Giedrius -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100225/37ceb99e/attachment.htm

Hello, Does asterisk func odbc support multi query? I'm executing stored procedure which returns two tables. With tsql command I can see both tables. But asterisk only shows the first. My database is MSSQL. Maybe there is workaround... Thanks -- Best Regards, Giedrius -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I want that, when call is answered , callee and calling would hear different prompts and after promts the calls would be bridged. I've tried this situation: exten => s,1,Set(LIMIT_CONNECT_FILE=hello-world) exten => s,2,Dial(SIP/trunk-out/37052390920|60|rL(10000000000000)A(conf-enteringno)) But these prompts play not in the same time: just after conf-enteringno prompt

Hello, My situation is that , I can't make calls with asterisk, but with x-lite works fine. Asterisk shows , that successfully registers with another SIP server, asterisk sends invite, gets trying, and after 30 secs asterisk gets 408 Request timeout. And as I said , with x-lite no problems. I heard that for comercial purposes, this SIP server detects asterisk , and ignores him. Or maybe it

Hi, I have problem with Asterisk 1.6.0.1. I need to change language for playing prompts in Lithuanian. But in Asterisk 1.6.0.1 version everytime plays in English, but in Asterisk 1.4.x I haven't any problem. Maybe it is a bug ...? So I paste my test dialpan and prompt's locations. I hope this helps you. Files are: [root at voip asterisk]# find /var/lib/asterisk/sounds/test -name

Hi, I have Sipura SPA-841 with two lines. And I have some little problems with it: 1) How to turn off alerting tone in Sipura, cause when I'm trying to call , I hear two alerting tones (I also have audiocodes product and I don't hear two alerting tones, just tone)? 2)The second problem: How to enable two lines to work with one number. For examle, if I'm talking with

Hi, I 'm using spandsp-0.0.2pre26<http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.2pre26/>, and thereis a bug adding headers: LOCALHEADERINFO and LOCALSTATIONID (I can't see them ). But faxes goes using rxfax and txfax fine. I also have tried development versions, the bug is fixed, but I get bad faxes (I get one page, but my tiff consists of three pages, and I get just

Hi, I'm analyzing freepbx extensions. When creating ivr with freepbx, it writes like this: exten => 1111,1,Answer exten => 1111,n,GotoIf($["${CONTEXT}"="from-internal"]?USERCID:SETCID) exten => 1111,n(USERCID),Macro(user-callerid,) exten => 1111,n(SETCID),Set(CALLERID(name)=${CALLERIDNAME}) exten =>

Hi, I'm writing asterisk application in C language. I need to know what is state of my asterisk user, so I have found command: ast_device_state(data); . So if my IP phone is reachable I get status 1 (AST_DEVICE_NOT_INUSE<http://www.asteriskpbx.com/doxygen/1.2/devicestate_8h.html#42ea804da1426b4117686332400b27c2>). But when I have unplugged my phone's cable , and sip show peer