similar to: Call Hold

Displaying 20 results from an estimated 20000 matches similar to: "Call Hold"

2008 Feb 13
2
UK issue - Asterisk dialling 999... sort of
Hello This is a fun one for the list... Twice now, the Police have contacted us to say they have had a silent call then hangup from our landline 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
2007 Dec 03
3
Replacing Skype with Asterisk Peering Servers - and Security
Hi, I have successfully configured two OpenBSD ( 4.2 & 4.0 ) Servers to do IXA2 peering on two remote Sites. Now asterisk users on Site1 can talk to users on Site2. I just would like to know the following details. 1) Currently I have allowed all in coming traffic from "Site1 Public IP Address" on Site2 Server and vice versa. Is that really required? Is it possible to narrow down
2006 Nov 28
1
Call recording filename
I am using asterisk along with freepbx . When recording is enabled for a extension the call record file made in /var/spool/asterisk/monitor contains information like OUT(extension number)-(timestamp)-(uniqueid).wav . This can be a big mess if there are more than 1000-2000 files in that folder and very hard to locate a call recording based on call time and extension number who dialled. I need to
2006 Jan 20
2
Conversation interrupted by fax
Asterisk SVN-trunk-r7353M (will be moving to 1.2.2 this weekend) E1 connected to Sangoma A102 SIP phones (Cisco 7960) I've been making a call from my mobile to the office, when, suddenly the conversation is terminated and replaced by a "fax-type" sound. This has happened to me several times over the past year, so it's not the version of asterisk (we've had cvs trunk and
2004 Sep 14
1
What does 'Forbidden (From header is not a Trust host or gateway)' mean?
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
2003 Jul 09
1
PRI with variable length numbers
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
2006 Feb 27
3
Matching '*'
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, Asterisk executes it again with
2008 Dec 30
1
Newbie Polycom: Cannot conference with >10 digit 3rd party
Calling all Polycom gurus: I am using Polycom IP601 phones with Asterisk 1.4.21.2 In all Polycom phones, I set the following in sip.cfg. <dialplan dialplan.impossibleMatchHandling="2"> </dialplan> (I leave the digitmap unchanged because I thought setting impossibleMatchHandling will ignore the bitmap) ...so that I could dial any number by entering a variable-size
2004 Sep 14
1
Wrong ID going out...
Hi all! I'm trying to have asterisk route all outgoing calls out via my VOIP provider. exten => _NXXXXXXX,1,Dial,SIP/BYEXTENSION@VOIP seems to have them to in the direct direction. However, debug shows that my asterisk doesn't identify itself correctly: Sip read: SIP/2.0 100 Trying From: "Evert"<sip:asterisk@[my IP]>;tag=as0aca53fa To: <sip:[dialled
2009 Oct 17
3
OT - DECT SIP Phones
Hi, I have three Snom M3s at the moment but getting pretty fed up with the issues :( I am UK based and would be interested to hear of other peoples recommendations. Key features :- * VM Notification * Good Range * G729 codec support * Common/Private Address Books per Handset(s) TIA, Best Regards, -- This message has been scanned for viruses and dangerous content and is believed to be
2020 Oct 22
2
Multiple IP addresses and using same IP for outbound calls as inbound
Hello, We have an Asterisk server with two public IP addresses, let's say 1.1.1.1 and 2.2.2.2. Normally calls come in to 1.1.1.1 and are bridged with a call dialled from Asterisk to an external destination. The external destination sees the SIP packet as coming from 1.1.1.1 and the media address in the SDP is 1.1.1.1, which is great. However if we receive a call in to 2.2.2.2 then the call
2008 Jan 15
3
Meetme recording
Hello, Is there a way to change the format from the default? 'r' - Record conference (records as ${MEETME_RECORDINGFILE} using format ${MEETME_RECORDINGFORMAT}). Default filename is meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav. - requires chan_zap.so Many thanks ******************************************************************** This email and any attachments
2008 Dec 04
3
BT - ISDN30 - International Calls not working, everything else is fine :(
Dear All, Thank you for taking the time to read this post - I am *confused!* as to why my asterisk setup does not work as it should. I have an ISDN 30 connection for telephony, a Sangoma card, and asterisk installed. Incoming calls, and outgoing calls work 100%. Making an international call, results in silence, or the error message all circuits are busy Numbers being passed to the trunk for
2010 Jun 02
6
How do you hangup a call without terminating your session?
Asterisk 1.6 CentOS 5.0 All - I'd like to offer my users the ability to hangup a call by pressing **. I'm using an attendant, so when ** is dialled I'd like processing to return to the attendant so the user can place a subsequent call. I have setup features.conf to include: [featuremap] disconnect => ** My Dial command looks like this:
2011 Apr 13
4
AGI and forking
Hi. I just want to make sure I understand this before doing something that might break things spectacularly for our users and customers :) We are using Asterisk 1.6.2.9 and my programming language of choice is Perl. I want, when a call comes in on someone's DDI number (which the person who dialled it can only possibly have obtained by dialling 1471 after we called them), to be able to
2003 Aug 21
3
Sending dtmf over an ougoing call from asterisk
Hi list, I would like to know of a possible way to dial a pstn number with an extension . Let the number is 56626965-234 so now i wanna dial 56636965 then wait for some time and dial the extension 234 to reach a particular person.I am afraid that i could not figure it out. I am trying in this way.. [outgoing] exten=>_566X.,1,wait,2 exten=>_566X.,2,Dial(${EXTEN})
2010 Nov 29
3
How to initiate a two-party call from within Asterisk
The desired result is that user A's phone rings; when he picks it up, user B is dialled, and user A's channel is connected to that. (This is to be a back-end for a web-based address book.)
2013 Feb 19
2
Call Pickup how to display CND of incoming number
Is it possible to display the incoming calling number on a handset when trying to pick up a call from another handset? I currently have Call Pickup working using *8, I have also used the PickUp application successfully but I'm not sure how to use these features so the handsets show the incoming calling number and not the number that you have dialled to pick up the call. Regards David
2004 Jun 11
1
"Caller ID" question
Since caller ID does not work with my FXO card, I am wondering if Asterisk supports the following extensions functionality. When a call comes in, I'd like to give the caller an opportunity to enter an extension if he/she knows it; if not, Asterisk will dial one or more default handsets. I know Asterisk can do this, but is it possible to change the default "caller id" when the call
2010 Mar 29
3
Slightly more advanced dialling..
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: