similar to: external RTP IP address

Displaying 20 results from an estimated 8000 matches similar to: "external RTP IP address"

2009 Sep 25
3
disable dtmf on SIP peer
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:
2008 Nov 17
1
asterisk conference
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
2008 Dec 16
2
starting call recording using AMI or other stuff
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:
2009 Aug 18
1
avoid indicate condition 9 and starting music on hold
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:
2009 Feb 27
1
change language and playback issue
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
2008 Nov 26
1
language and meetme issue
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:
2008 Dec 17
1
ael queue gosub already has PBX structure??
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(${Queue},wr,,,60,,,check-record); Hangup(); }; }; macro check-record() {
2006 Jun 14
0
NCS patch
Hi, I have cable modems Arris with MGCP protocol. And I need PacketCable NCS patch for Asterisk. http://asterisk.urtho.net/ doesn't work! -- Pagarbiai, Giedrius Augys Siauliu Universitetas, IST IP telefonijos inzinierius Tel. 8 41 590408 Mob. Tel. 8 678 05790 el. pastas voipas@gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL:
2007 Oct 17
2
asterisk hylafax iaxmodem
Hi, I have problems with asterisk and hylafax+ iaxmodem. I can successfully send faxes to Panasonic KX-FT932 fax, but with Xerox WorkCentre M20i I have problems: No carrier. This is hylafax log, maybe you can suggest me where to find ... Oct 17 07:38:48.22: [22428]: SESSION BEGIN 000000041 180037052390906 Oct 17 07:38:48.22: [22428]: HylaFAX (tm) Version 4.4.2 Oct 17 07:38:48.22: [22428]: SEND
2008 Oct 05
5
asterisk, phpagi and singleton
Hello, I've this situation: 300+ simultaneous calls and dialplan like this: exten => _X.,1,Answer() exten => _X.,2,DEADAGI(check_status.php) exten => _X.,3,Dial(SIP/other/${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
2009 Nov 06
1
app read accept # sign
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
2008 Nov 24
1
play sound while executing agi script
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:
2008 Dec 01
1
func_odbc questions
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} ${REALNUMBER1},
2008 Dec 02
1
func_odbc and hash problem
Hello, Now I'm testing func_odbc and hash. My configurations are: func_odbc.conf [GETNUMBER] dsn=sqlserver ;mode=multirow ;rowlimit=10 readsql=SELECT number,real_number1,real_number2,status FROM ivr.dbo.numbers WHERE number=${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 =>
2010 Jan 07
1
compile one additional module without recompiling all asterisk
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
2010 Jan 21
1
odbc question
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
2009 Dec 07
3
show queue's name and other info in incoming call to queue member
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
2008 Oct 08
0
fastagi example
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,
2008 Dec 16
0
realtime odbc queue member cache problem
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
2008 Dec 19
0
realtime queue change ring strategy
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