similar to: originate and local channel problem

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:

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:

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

Hello, I've created Call Center with Asterisk (1.6.0.5). Call Center's agents are not Asterisk SIP user's, but other's voip gw SIP 5 class users. Everything works fine, except when one agent wants transfer call to other agent. They do it with flash hook. So and two voip gws (Asterisk and other gw) detects this flash and both starts playing MOH. The transfer is unsuccessfully,

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:

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 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() {

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'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, Is it possible to do like this: play a sound file (if needed play in loop) while php agi script finishes work ? And how to do this? When on my server is huge load , I don't want that client hears silent , but hears music. Thanks -- Pagarbiai / Best Regards, Giedrius Augys -------------- next part -------------- An HTML attachment was scrubbed... URL:

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'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, maybe somebody has fastagi examples, or can advice how to do. I just want to do a single ton connection to mysql server. Cause now I'm using AGI, and each call creates mysql connection and so on. I just want alleviate CPU load ... Asterisk and mysql servers are on the same box, and is it a good idea use fastagi if i have only one server. thanks -- Pagarbiai / Best Regards,

Hello, I have asterisk 1.6.0.1. I'm using realtime asterisk with MS SQL. Everything is OK, but I have noticed strange thing with queue members. If I modify just 'membername' - asterisk do not refresh this info. But if I make changes also in 'interface' column, and after executing command `queue show my-queue` and I see changes. So is it possible that asterisk after

Hello, I'm using asterisk 1.6.0.1 and realtime queues. But when I make changes in database (for example: change strategy from ringall to random), but asterisk shows old strategy, doesn't update this parameter. My question is, how I can dynamically change ring strategy. Thanks in advance. -- Pagarbiai / Best Regards, Giedrius Augys -------------- next part -------------- An HTML

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