similar to: set box IP from which send sip traffic

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

Hi guys, I have a mail server running qmail with two network interfaces. All packages comes from the first one and I want to mark outgoing packages to route them using the second interface (ADSL). I have write this script, but it ins''t work. All packages are returning by the first interface... :-( x-x-x # Tables, networks, interfaces, addresses and gateways IF1=eth0 T1=100

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 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 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'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

Hi All, I''ve configured a mutlipath as detailed in the HOWTO. For the most part it''s great, but when the cheap DSL line drops it''s connection, as it regularly does, the outbound routing doesn''t automatically use the remaining path for internal outbound traffic. Ip route show lists no default route once the default path dies, and if the ppp connection is

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},

Hi. I'm in desperate need of some good advice. I have a tinc network with 16 nodes. It's a star topology where all nodes are connecting to the one node (Node1) that have a static IP. Node 1 accepts incomming connections Node 2 through 16 connects to Node1 One of the nodes (Node5) stopped working a while ago (2 - 3 weeks or so), other than that everything was working fine. Today I

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 all, I need to put an IP alias to the same ethernet card, like this: eth0 - IP1/MASK1/BROADCAST1 eth0:1 - IP2/MASK2/BROADCAST2 In this box I will have another card that conects to a LAN netowork. I need this two alias in eth0 becouse I have two routes do take, but it seens to do not work. This is the script I am using (this scripts works fine when I have two ethernet cards

Hi, I configured a router box to use 2 providers, as described in the HOWTO. (Apendix 1) I want to use both links to reach a single smtp server. As I read in the kptd and in some old messages of this list, doing a SNAT in the postrouting chain comes _after_ the routing desision. So I guess the following lines I''m trying to use are wrong. (See Apendix 1) What can I do to have multiple

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I''m trying to setup a central IPSEC-Gateway with several ipsec tunnels. Some are to be routed over one leased line, some over the other leased line. Both leased lines have their own public ip adress. The setup looks kinda like this: eth1(ipsec0)--ISP0--Internet--eth1-Linux1-eth0--Subnet1 /