similar to: Defining a call

Hi list, someone has successfully change different ringtone from dialpan with asterisk with this model Granstream? for example: exten => 0,1,Playback(pls-wait-connect-call) same=> n,SIPAddHeader(Alert-Info:;info=ring3) same=> n,Dial(SIP/310&SIP/318,30,t) can not get it to work any idea o tips? regardss -- rickygm http://gnuforever.homelinux.com

Hi All I have just upgraded from Asterisk 1.2 to 1.4 and am having trouble with with one of my ATAs not ringing. Basically, when I execute the Dial command, an error occurs: "Got SIP response 400 "In alert-info header: Empty value expected" Now in 1.2, I just issued the following command to overcome this problem: Set(_ALERT_INFO=). Now in 1.4, _ALERT_INFO is deprecated, so I

Hi guys I'm using asterisk 1.4 and going on to Snom phones. I'm trying to add a sip header to make the Snom phone use a different ring tone on one particular incoming number. I have added the following to the dial plan of the incoming context +------+------------------+-------+----------+--------------+-------------------------+ | id | context | exten | priority | app

Hi, I am trying to write dial plan for sip to auto answer (auto attend) the incoming call to the sip phone. - If i call from sip1 to sip2 then sip2 should automatically answer the call and play some sound file. I am trying to do this but as new to the asterisk dial plan configuration , so not able Todo this. help me if anyone already done this setup. Regards Upendra. -------------- next part

Have someone running fine Alert-Info with a Snom 370 ( System Information: Phone Type: snom370-SIP MAC-Address: 0004132661BD IP-Address: 192.168.10.170 Firmware-Version: snom370-SIP 7.3.14 14961) i've tried exten => 200,1,SIPAddHeader(Alert-Info:<http://www.notused.com>\;info=alert-external) exten => 200,n,Dial(SIP/${EXTEN},30) Can see into the phone SIP trace is

Hello, would like to have distintive ringtone for internal calls, google gave me blurr answer. My extensions are 46**, any calls made within 46** I want to ring differently than external calls. Thanks in advance.

Hi - Just wondering if anyone has gotten a Polycom VVX phone to successfully do an Auto Answer with asterisk. I have an older generation of Polycom phones that do this just fine, but I can't seem to make the VVX phones work. I tried the guide here: http://community.polycom.com/t5/VoIP/FAQ-How-can-I-change-my-Ringtone-or-Ring-in-a-special-manner-for/td-p/5167 And I have this in my diaplan:

Ok so I added to my Freepbx config running Asterisk 1.2.4 in extensions_custom.conf ; intercom exten => _7XXX,1,SIPAddHeader(Alert-Info: Auto Answer) exten => _7XXX,2,Dial(SIP/${EXTEN:1},12,Tt) and configured my Polycoms via this page http://www.voip-info.org/wiki/view/Polycom+auto-answer+config for auto answer and that works fine if I dial 7 then the 3 digit extension. No

Hallo Group, I have basically set up a small asterisk system, which ahs 4 peers: * registers at 2 Sipgates * 2 hardware phones connected to it Both Hardware phones can phone outwards(cheaper sipgate is selected with dialplan) Calls from both sipgates make my hardware phones ring But here comes the challenges: Is it possible to configure asterisk in such a way that in the phone: * there are

Hi Everyone, Using Elastix (FreePBX + Asterisk 1.4.2x combination) with Aastra phones, how can one receive distinctive ring tones for INTERNAL calls ONLY? Even though FreePBX Inbound has an option for Alert_INFO but that doesn't work when the call comes into an IVR or Queue. The calls has to go directly to extension for external ringtone to be different. So, I am looking for internal calls

I have been looking for the specs (format, bit rate, ect) on custom ringtones for digium phones. Using the DPMA how would I deliver the ringtone to a digium phone? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120820/cb4927d0/attachment.htm>

Would it be possible to set the ringtone based on the number that was dialed? Example of what the goal is: Dial Denver number Incoming calls ring with ringtone 1 Dial main number Incoming calls ring with ringtone 2 We are currently using Digium D40, D50, D70 phones. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello list, I notice that the dialplan method SIPAddHeader is not working : in dialplan : /exten => s,n,SIPAddHeader(Privacy: id)/ in SIP invite no trace of this header : /INVITE sip:0473 at sip.domain.be SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97 From: "VC" <sip:voip2 at sip.domain.be>;tag=729476652f511c67o2 To: <sip:0473 at sip.domain.be>

Hi, I'm using the new SipAddHeader application on Asterisk 1.2.1, here's a snip of my extensions: exten => _9XXXXXXX,1,SipAddHeader(P-Asserted-Identity: <sip:${CALLERIDNUM} exten => _9XXXXXXX,2,SipAddHeader(P-Asserted-Identity: tel:${CALLERIDNUM}) exten => _9XXXXXXX,3,Dial(SIP/${EXTEN}@${SIPTRUNK},,tT) exten => _9XXXXXXX,4,Congestion The problems is that Asterisk

i could add "r" option in dial command. this will generate a ringtone during connection. could i change this default ringtone? i tried indications.conf but not success. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120103/77a4d48f/attachment.htm>

How to add SipAddHeader in outgoing call file. I am implementing a Callback scenario, in which a user makes a call to Local Access Number. The system have to callback to the user. During callback a call file is generated. All I want, is to add SipAddHeader("pchargingvector","val") in outgoing Invite. How can I achieve this? regards, Asif

Hi, i have a problem with iax2 and ringtone. Here is the call path pstn -> asterisk -> iax -> firefly or any iax phone. My problem is when i receive a call on my iax phone, the ring sound is very distort and bad. If i open my sip phone, and receive a call from my pstn, the ring is like dring dring, very normal. Otherwise, it is like a machine gun with iax Help would be really

Hello all, I'm struck with a very strange problem today. I've an AGI with some code subroutine snippet as follows: sub enable_sbc($) { my $carrier = shift; my $tmp = substr($carrier,1); my $jkh = $tmp; $server_port = $ast_agi->get_variable("SIPPEER($jkh,port)"); $ser_ip = $ast_agi->get_variable("SIPPEER($tmp,ip)");

Hi Friends, I am facing a strange problem with DISA. I have installed and configured Trixbox. I've created a secret extension i.e., 555 and called this extension in Digital Receptionist using custom extension i.e., created in extensions_custom.conf file. When I call from my mobile phone to my PSTN number, which is connected to FXO port, my IVR is responding. After entering my DISA

Hi, Snom phones firmware 5.0 is now out. Try it if you like: http://www.snom.com/wiki/index.php/Main_Page. Regards, --------------------------------------------------------------------- Usman Tahir snom technology AG www.snom.com --------------------------------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: