search for: dialled

...number to the 999 service. As a matter of course, they follow up these calls in case someone is in distress. Nobody here was in distress - well, no more than normal! The Police aren't hugely happy when we tell them it must be a mistake. Thing is, I have checked both our master log, and our dialled calls log - and nobody dialled 999! Each phone has an account code applied from sip.conf, and we log all numbers dialled. Nobody dialled out. There are no phones connected in anyway other than via Asterisk, fax number is dealt with by a virtual machine, alarm system is on a different number......

Hello, I'm wondering if it is possible to ring X number of extensions simultaneously, and each answered call can be handled with some code. I can do a huntgroup-esque way of dialling, but I want all the dialled numbers to be picked up. I hope this makes sense.. If not please say.. Many thanks! Andy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100329/26fffe68/attachment.htm

I'm using DUNDi. My lookup returns 'IAX2' for the tech, and 'dundi:q9sgTFkVMBFdmp0IDX1bYQ@xxx.187.142.204/3254101' for the destination. How do I dial this? I've tried dialling it with: "Dial" "IAX2/dundi:q9sgTFkVMBFdmp0IDX1bYQ@xxx.187.142.204/3254101" passed from my AGI script, but the other endpoint (xxx.187.142.204) is returning: May 11 09:23:41

Hey all, I have an Asterisk-box with an E100P and a PRI (Euro-ISDN) coming into it from a Meridian-switch. The incoming numbers on this PRI all start with the same digit and the last part of the dialled number is signalled to Asterisk digit by digit, until Asterisk signals that the number is complete and the call rings. All works well, unless I have 2 or more numbers which start with the same digits. In that case, dialling will be signalled as complete as soon as the shortest of the numbers is d...

...mp;SIP/3254102,20,tr) and I want to check the ${DIALSTATUS} variable after the dial, how do I know which number I am getting the variable for? And, what about this? Dial(SIP/3254101&SIP/3254102@proxy1,20,tr) What happens in that case? How can I get the ${DIALSTATUS} variable for EACH NUMBER dialled? Thanks, Doug.

Hey all, I'm having problems reliably dialling out my FXO card. About 30% of the time I'll get a "your call cannot be completed as dialed". I'm thinking it might be the dialling speed, but I can't find any configs that change that setting. Any suggestions for troubleshooting? Thanks, Brad Waite

...terisk <---pri---> legacy pbx everything is fine exept that when dialling from the legacy pbx it takes about 3 seconds before the asterisk start to dial. where does this delay come from? has it to do with 'overlapdial=yes'? here is an excerpt from the logfile (i assume the number is dialled enbloc as it come with the redial function of the legacy pbx): 2005-07-28 17:23:37 VERBOSE[13873] logger.c: -- Accepting overlap call from '070314161XXX' to '<unspecified>' on channel 0/1, span 2 2005-07-28 17:23:37 VERBOSE[13997] logger.c: -- Starting simple switc...

Hello, I'm having what seems like a weird issue connecting Asterisk 13 (FreePBX 12) to a Cisco 2811 router via a chan_sip trunk. Whenever I try dialling out via this trunk, something appends '@CUBE' onto the end of the dialled number, as per the following examples; Asterisk log; app_dial.c: Called SIP/test/0429123456 at CUBE chan_sip.c: Got SIP response 500 "Internal Server Error" back from 172.22.4.12:5060 In the SIP SDP; INVITE sip:0429920437%40CUBE at 172.22.4.12 SIP/2.0. To: <sip:0429920437%40CUBE at 1...

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040728/3306e606/attachment.htm -------------- next part -------------- ? Hi, I'm dialling 1234 in the softphone or Grandstream Phone and without disconnecting the phone i want to dial "10" after dialling "1234". Is it possible to do? Regards shan

Hi Is it possible with Asterisk to tell the called party which number was dialled by the caller? Or in place of the number dialled have a description such as 'Sales' or 'Accounts'? Ideally, I would like to show a description corresponding to the number dialled followed by CIDName. How might this be set up? Currently my extensions.conf is: exten => xx,...

I'm trying to find a way in extensions.conf to match ANYTHING dialled, including characters such as *. The following works for numbers... exten => _X.,1,AGI(script) but doesn't catch when someone dialls * first. I tried this: exten => _.,1,AGI(script) which catches when someone dials say, *123 for example, but after the AGI script terminates, Asteri...

...a HFC PCI ISDN card, running in NT mode. The ISDN card is connected to a S0 bus and to a Siemens ISDN PBX. Two phones are connected to the ISDN PBX and are successfully getting calls from the asterisk box. When dialling from one of the phones, the ZAP channel seems to be missing out on some of the dialled digits everytime, i.e. if I dial 099557896, the asterisk box receives 09955896 sometimes, or 0995789, or something like that. This only happens on one of the phones, the other one is dialling fine and digits are being recognized well. I already tried setting relaxdtmf=yes in zapata.conf, but to no...

...("Accepting voice call from '11234567' to '250' on channel 0/1, span 1"). The problem however is with calls from analog lines: "Accepting voice call from '13331846' to '25' on channel 0/1, span 1" * just sees 2 digits, not the 3 digits that were dialled. so I defined some goto extension as a quick fix like exten => 25,Goto(250,1) now this does not really solve my issues, as the missing digit ("0" in this case) magically appears a few seconds late. but when it does, * is already humming away in the middle of the 250 extension and su...

...sterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of David Cunningham Sent: Tuesday, 18 August 2015 2:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number Hi Brendan, Can you attach an Asterisk log with "sip set debug on", "core set verbose 9" and "core set debug 9"? On 18 August 2015 at 10:33, Brendan Ord <bord at staff.onthenet.com.au<mailto:bord at staff.onthenet.com.au>> wrote: Hello, I?m having...

From a 'sip debug': Sip read: SIP/2.0 100 Trying From: "Evert"<sip:[username]@[my ext. IP]>;tag=as6e18534e To: <sip:[dialled number]@[SIP server of VoIP provider]> Call-ID: 6cbf41c25281f08b2e7bbc5043061975@[my ext. IP] CSeq: 102 INVITE Via:SIP/2.0/UDP [my ext. IP]:5060;branch=z9hG4bK4fd1045b Content-Length:0 7 headers, 0 lines Sip read: SIP/2.0 403 Forbidden (From header is not a Trust host or gateway) From: &quo...

When I use an analog phone connected to a Sipura SPA-2000 it takes about 3-4 seconds before the number is actually dialled. Very annoying especially if you are connecting an intercom to it. Can I change this behaviour and do I need to look at * config or the config of the SPA-2000? Thanks!

...angoma card, and asterisk installed. Incoming calls, and outgoing calls work 100%. Making an international call, results in silence, or the error message all circuits are busy Numbers being passed to the trunk for the call ? National is 020xxxxxxxx will result in 20xxxxxxxx being sent and dialled, which works ? Mobile is 07xxxxxxxxx will result in 7xxxxxxxxx being sent and dialled, which works ? International 00x[any number of digits] will result in 00x[any number of digits] which does not work I do not see why this does not work. I do know that for every call, the flag sent is...

I'm trying to use Jeremy's suggestion of dialling using just the peer name instead of user:pass@peer but I'm running into some really funky issues. It does the same thing with both VoicePulse and another * server I have. [voicepulse] type=peer host=gw5.voicepulse.com trunk=yes user=USERNAME pass=PASSWORD and in my dialplan: Dial(IAX2/voicepulse/${EXTEN:2}@VPWS,90,r) The log shows

.... I just want to make sure I understand this before doing something that might break things spectacularly for our users and customers :) We are using Asterisk 1.6.2.9 and my programming language of choice is Perl. I want, when a call comes in on someone's DDI number (which the person who dialled it can only possibly have obtained by dialling 1471 after we called them), to be able to look up the caller's details from one of our databases (where the number ought to be stored, because we already dialled it). Now, this search is going to take some time; so I'd like for the AGI scr...

...works perfectly. Outgoing callerId (set by the external provider) works as expected. ) I have dialling prefixes for each 'line', nothing special there, that side of it all works as expected. The problem is that only the last one in the sip.conf file actually accepts incoming calls when dialled from the PSTN side. (They have different PSTN phone numbers) If I swap their entries over in the sip.conf file, then the other one takes the calls. When dialling the first number, nothing seems to get through to the * box at all - nothing on the console in verbose mode, nothing in the log-file....