search for: 1&sip

...pe 'SIP' (cause 20 - Subscriber absent) if the subscriber is not registered. Is there a way from dialplan to know, *before* Dial(), if a destination Subscriber is a) not registered or b) busy ? I need to redirect a call to some other Subscriber if (s)he is not there like this: Dial(SIP/1&SIP/2&SIP/3). So if "2" is not registered (or is busy) then Dial(SIP/1&SIP/5&SIP/3). Regards -- Ethy H. Brito /"\ InterNexo Ltda. \ / CAMPANHA DA FITA ASCII - CONTRA MAIL HTML +55 (12) 3797-6860 X ASCII RIBBON CAMPAIGN - AGAINST HTML MAIL S.J...

I have the following dialplan (segment) that isn't working as I expected it to: exten => s,n,Dial(Zap/1&SIP/202&SIP/203,18) exten => s,n,Dial(Zap/1&SIP/201&SIP/202&SIP/203,42) The plan was to have SIP/201 added to the group of ringing phones after 3 or so rings. What ends up happening, though, is the Zap/1 phone STOPs ringing when the dialplan falls through to the second li...

Hi All, I'm confused! My Asterisk system has a Zap trunk and three SIP trunks. I'd like to configure the dialplan to route via the first trunk in a list and if that's not available or it's busy, fall over to the second, then to the third, etc. AIUI Dial(Zap/1&SIP/out1&SIP/out2/${EXTEN}) rings all the trunks in the list and bridges to the first to answer. Unfortunately, that's not what I want, which is (in pseudocode): if Zap/1 is available then Dial(Zap/1/${EXTEN}) elseif SIP/out1 is available then Dial(SIP/out1/${EXTEN}) else...

Hi, I intend to let several SIP-phones on my asterisk ring cascaded on incoming calls. First only phone 1 should ring, after 5 seconds phone 2 should ring in addition and after additional 5 Seconds phone 3 should also ring. How can I realize that correctly? Currently I do use Dial(SIP/1,5) Dial(SIP/1&SIP/2,5) Dial(SIP&1&SIP/2&SIP/3) But this seems not to work correctly on phone 1...

...f the mobile rejects the call (by pressing hangup while it rings), something strange happens: the following is seen in the logfile, everytime a rejected mobile call happens: ----------------- Mar 20 22:52:29 WARNING[4682]: Forbidden - wrong password on authentication for INVITE to '"0174xxxxxxx" <sip:yyyyyyy@sipgate.de>;tag=as03bffab2' Mar 20 22:52:40 WARNING[4682]: Maximum retries exceeded on call 3d4fb6381c1ddf6e17062fc03cb3f936@asterisk for seqno 102 (Non-critical Response) ---------------- on the sip phone the ringtone stops, but asterisk does not hangup th...

Dear asterisk-experts, i've got a problem with my Dialplan. The task is to get the SIP-address of the called internal phone, that answered as first. In this example, two phones are ringing: DIAL(SIP/1&SIP/2|120|m) But I want to trigger an event, if someone picks up DIAL(SIP/1&SIP/2|120|mM(answered)) The Macro function doesn't have any information about the current address of the answered phone. I'm not able to pass ARGs to the macro (i think i've to use the CVS Version), bu...

...end up with people chasing the call from desk to desk. In the example below if the caller hangs up during step 4 asterisk will continue on to step 5 and start recording a voicemail. I can't see that we we are trying to do anything unusual here, anyone able to shed any light? exten => 470,1,SetCIDName("Tech Support") exten => 470,2,Dial(SIP/1,10,tr) exten => 470,3,Dial(SIP/1&SIP/2,10,tr) exten => 470,4,Dial(SIP/1&SIP/2&SIP/23,10,tr) exten => 470,5,Voicemail(sb000)

Hi all, I have 12 SIP phones at a particular site all connected to a local asterisk server. It's in turn connected to 2 ISDN BRIs to provide up to 4 incoming calls. An IAX gateway is used for outbound calls. At the moment, when an incoming call comes in, asterisk dials every SIP phone like so: Dial (SIP/1&S...

...then like * to send the call to voice mail rather than attempting to send the call to another extension in the group. I've looked around trying to find a solution to this problem but I haven't found anything that works quite the way I want it to. I know you can use Dial(SIP/0&SIP/1&SIP/2,20,Ttr) to dial all three extension at the same time but this won't work for me. I also know that I could set up a dial plan to go from one extension to the next but I only want the phone to ring a max of 4 to 6 times. Also, I imagine I could use call queues but this is supposed...

I recently purchased a TDM11B so I could hopefully hook flash the FXO from either the FXS (on the TDM11B) or a SIP device. From the FXS, I've tried hitting # then transferring to an extension that flashes the line then dials the FXS again (3020). This seems to send me to a busy signal and the console tells me no such ho...

...if the extension is called both iPAQ and the IP phone ring and the user gets to pick up using either. Your input is highly appreciated. Thanks Walid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050109/cd73d10a/attachment.htm

I have a soundpoint 501 phone that has locked up twice now. You can make a call but when a call is sent to it, it responds with sip busy messages. You get the same message when the phone is in do not disturb. I reset to defaults the first time and it worked for a week or so and then stopped. The incoming calls are ringing...

Hi, Is there a way to limit only one registration for each user at a time? meaning if a user tries to register, but that user is already registered. i will deny? or is it possible to for a single user at the same time, and when someone calls that user, it will ring both phones? Just want something whereby a user can assign his extension on an IP phone in the office, and assign the same

...ource that will never answer but will always keep Dial() happy that it might? Thanx, b. -- My other computer is your Microsoft Windows server. Brian J. Murrell -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060206/d510574f/attachment.pgp

I have been experiencing a problem where after someone calls me from an analog line, the phone call is terminated after a period of time (anywhere from 15 seconds to 15 minutes) The phone that I use to answer the call is an Aastra 9133i SIP phone. There are several other SIP extensions on the network as well as a few analog extensions on a shared FXS line. When a call comes in the analog line on the FXO, * dials all the extensions (SIP an...

...today i just wanted to know if someone can help me to set X-Ten Lite to call PSTN line using my FX0 Currently , I am able to use X Lite to call another X lite user locally (LAN) I also has attached my setting together Thanking you all in advance -------------- next part -------------- {\rtf1\ansi\deff0{\fonttbl{\f0\fswiss\fcharset0 Arial;}} {\*\generator Msftedit 5.41.15.1507;}\viewkind4\uc1\pard\qc\lang1033\ul\b\f0\fs24 This is My Working setting for the TDM11B\par \pard\ulnone\par /etc/zaptel.conf\par \b0\fs20 fxoks=1\par fxsks=4\par loadzone = us\par defaultzone=us\par \par \b\fs24...

..._ *Technology and Quality on Information* Mauro S?rgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.brasil at tqi.com.br <mailto:@tqi.com.br> : www.tqi.com.br <http://www.tqi.com.br> ( + 55 (34)3291-1700 ( + 55 (34)9971-2572