similar to: Display number dialled

Hello When using LookupCIDName, Asterisk 1.4 says that it's deprecated, and we should use "${DB(cidname/${CALLERID(num)})}" instead, but I don't know how to use it: ;DEPRECATED exten => s,1,LookupCIDName ;ERROR exten => s,1,${DB(cidname/${CALLERID(num)})} I guess I should use this as a parameter to a function, but which one? Thank you.

Hello, Sometimes, when a call comes in from the PSTN through our VoIP gateway, the information that is sent to our web page that logs calls includes the original CID name instead of the one that is we expect to be rewritten on the fly using Asterisk's LookupCIDName: ================= ;extensions.conf [internal] exten => group,1,LookupCIDName exten =>

I have found that I can make the phones display any one word on this second line by adding a fromuser=<word> in sip.conf. This really isn't good enough though. When you look at the received calls or missed calls directory, each item has two lines, the first is the CID name, and the 2nd is supposed to be the CID number. However, if it is asterisk, or some other word, when you hit the

Because i want a ringing signal while people are in a waiting queue i've created a wav file containing our local ringing indication If I make an inside call to the queue, the correct sound is played, but when i make an external call, no signal is heard. everything else looks ok, and all other functions are ok.... Can somebody point me out what i could have done wrong? my musiconhold.conf

I'm having a problem with intermittent lookup of Caller ID Name info using LookupCIDName. The same problem occurs when doing: asterisk -rx "database show cidname" No data is returned on every fourth or fifth query. No errors are being logged. I'm currently running CVS-HEAD-07/07/04-17:04:31 and first noticed the problem a few weeks ago. Is anyone seeing a similar problem?

David, I should also note; 246 is my extension, it has IP 172.22.3.238. 172.22.4.8 is the PBX, and 172.22.4.12 is the 2800 gateway. The trunk is called ?testing? at the moment. The route that selects this trunk uses a 9 prefix. This system is in semi-production, so there might be fluff in the log from other active calls. Brendan Ord OntheNet - Network Engineer P 07 5553 9222 F 07 5593 3557

Yes indeed. Do you have the dialplan, eg from /etc/asterisk/extensions.conf? Something is getting this OUT_3_SUFFIX variable and including it in a Dial to 172.22.4.12. On 18 August 2015 at 16:21, Brendan Ord <bord at staff.onthenet.com.au> wrote: > Starting to make sense when I saw this line: > > > > [2015-08-18 15:01:33] DEBUG[19366][C-00001cfc]: pbx.c:3785 >

Folks, When I have a dial string like this: Dial(SIP/3254101&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.

Release 0.68 of IPSwitchBoard is ready for download at: http://www.voip-info.org/tiki-index.php?page=IPSwitchBoard+BETA Release Notes: Import/Export extensions to "cidname" in Asterisk Server, integrate this in your dial plan to see who's calling by using the Asterisk app. "LookUpCidName()." Set "Do Not Disturb" for an Extension, and type a reason for DND

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

Hello, So, I found this line under macro-dialout-trunk, in extensions_additional.conf (FreePBX, so it controls the conf files mostly); exten => s,n,Dial(${OUT_${DIAL_TRUNK}}/${OUTNUM}${OUT_${DIAL_TRUNK}_SUFFIX},${TRUNK_RING_TIMER},${DIAL_TRUNK_OPTIONS}) If I grep for OUT_3_SUFFIX in all files in /etc/asterisk I get nothing.. Here's a paste of a few things out of the two files that I

Hi there, I'm converting a dialplan callback type application to fastagi as I'm hitting the buffers with respects to getting useful results from CDRs. It works by a spool call file triggering a Local extension, that extension then does the first dial to a client. I dial to a local context from the spool file as I need proper return codes as in ${DIALSTATUS} which are not available

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

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!

Hi all. I am seeing incoming calls from digital lines (mobiles e.g.) dialling my main number + 3-digit extension just fine ("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

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

Hello Instead of using PrivacyManager, I'd rather use my own dialplan to prompt the user for a ten-digit number if they called while blocking CID. This code does prompt the user, but 1) hangs up if the user didn't type the ten digits before the timeout 2) if the user did type the right number of digits, it still hangs up instead of Returning and then jumping forth to the "cid"

Hello, I'm trying to fill CID data into the astdb using AsteriskWin32's asterisk.exe, to no avail: The batch file stops after the first line, and just waits: ---------------------------------------- rem c:\cygroot\mystuff>import.bat rem rem c:\cygroot\mystuff>C:\cygroot\bin\asterisk.exe -rx 'database put cidname 123 "My cellphone"' rem rem Asterisk module

Hi All, I'm looking for a way to filter the AstDB cidname family to show only those entries with a specified area code in the Asterisk CLI. If this were a SQL database it would be something like: SELECT number, name FROM cidname WHERE number LIKE '1234%' I've tried "database show cidname 1234*" and substituted "%", "$", "-" for the wildcard

Admittedly this is probably pretty stupid of me, but there are just some things I can't understand by reading documentation. Any suggestions or recommendations about how to handle my problem are greatly appreciated. I'm trying to achieve the same functionality as my Nortel PBX, without rewriting much 'C' code. In my dialplan I'd like to compare two variables as a means of