similar to: play sound while executing agi script

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, 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:

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 problems with asterisk and hylafax+ iaxmodem. I can successfully send faxes to Panasonic KX-FT932 fax, but with Xerox WorkCentre M20i I have problems: No carrier. This is hylafax log, maybe you can suggest me where to find ... Oct 17 07:38:48.22: [22428]: SESSION BEGIN 000000041 180037052390906 Oct 17 07:38:48.22: [22428]: HylaFAX (tm) Version 4.4.2 Oct 17 07:38:48.22: [22428]: SEND

Hello, I have created a dynamic conference into two languages (english and russian). Client calls to confrence number and interactive choose the language. Meetme runs with 'dMi' options. Everything works perfect if one conference room clients have choosed the same language. If clients had choosed different language , there is a problem with user join/leave announcements. For example:

Hello, I've this situation: 300+ simultaneous calls and dialplan like this: exten => _X.,1,Answer() exten => _X.,2,DEADAGI(check_status.php) exten => _X.,3,Dial(SIP/other/${NUMBER}) exten => _X.,4,Hangup exten => h,1,DEADAGI(cdr.php) When project is running , I had a lot of defunct php scripts (I've exceed mysql connection limits and so on, deadagi help a bit). The

Hello, I want that after client and queue member call would be established, cmd queue runs some 'procedures' . So I am using cmd Queue option 'gosub'. This is my example of ael : context QUEUE { _X. => { Ringing(); Wait(4); Answer(); Queue(${Queue},wr,,,60,,,check-record); Hangup(); }; }; macro check-record() {

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'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'm working with asterisk 1.6. And I have success using func_odbc with one row query results (SELECT source,destination from cc WHERE ... ): exten => s,1,Ringing exten => s,n,Wait(4) exten => s,n,Answer exten => s,n,Set(ARRAY(NUMBER,REALNUMBER1,REALNUMBER2,STATUSAS)=${ODBC_GETVARIABLES(${NUMERIS})}) exten => s,n,Verbose(1| ${NUMERIS}, ${REALNUMBER1} ${REALNUMBER1},

Hello, Now I'm testing func_odbc and hash. My configurations are: func_odbc.conf [GETNUMBER] dsn=sqlserver ;mode=multirow ;rowlimit=10 readsql=SELECT number,real_number1,real_number2,status FROM ivr.dbo.numbers WHERE number=${SQL_ESC(${ARG1})} extensions.conf exten => s,1,Ringing exten => s,n,Wait(4) exten => s,n,Answer exten => s,n,Set(NUMERIS=37037210602) exten =>

hello, I'm using Asterisk 1.6.0.5 . And I'm creating IVR, and I need that Read application accepts # sign, So is it possible? And maybe there is a workaround? Thanks -- Pagarbiai / Best Regards, Giedrius -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091106/2f2b443c/attachment.htm

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

Hello, Maybe there is the easiest way to compile additional my module without recompiling all asterisk? Thanks -- Pagarbiai / Best Regards, Giedrius -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100107/cfe8f0b7/attachment.htm

Hello, I want to know what is timeout for MS SQL connection? My config is: [mydb] enabled => yes dsn => MYDB pooling => yes limit => 200 share_connections => no username => login password => password pre-connect => yes backslash_is_escape => no In the peak , I can see : ODBC DSN Settings ----------------- Name: mydb DSN: MYDB Pooled: Yes Limit: 200

Say there! This is a new Asterisk install on Fedora; I downloaded and installed Asterisk 1.2.1 and it runs fine. I downloaded the 20051217 version of chan_sccp, compiled and it installs ok. When I start Asterisk, it dies on startup at the activation/recognition of the chan_sccp module... A snippit from the startup: Asterisk Dynamic Loader Starting: [res_musiconhold.so] => (Music On Hold

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:

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:

Hello, I have a strange h323 issue. After executing command "Dial("SIP/333-0d1dfe00", "H323/361737052390920 at ccg|5|tT")" at Oct 18 22:32:23. Meanwile I have sniffing traffic on port 1720. The call was established just at Oct 18 22:33:03 (New H.323 Connection created.) and also packet sniffer grabs the h323 invites at this time also. So my question is what