similar to: application doesn't dial out...

hi everyone. can anyone provide me concrete examples on how to use the GotoIf application? can't figure out how to use it in my dialplan coz im having errors....thanks! : ) __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -------------- next part -------------- An HTML attachment was

hello! i've tried to use asterisk 1.2 beta version and all installed fine except that when i config asterisk to run at bootup it gives me an error which says 'execvp:Permission Denied!' then the loading fails so i stil has to start asterisk manually when i've logged in as root which is too much work, i want asterisk to initialize immediately on bootup. can someone help me in

In my application, I am using AMI to run an Originate command between a channel and a dialplan application (NOT a context). In my case, the application I want to invoke is FastAGI. The Originate AMI command works correctly, but Asterisk generates a very short (0-1s) duration for the CDR that results from this call, regardless of the time spent running the FastAGI application. I want the CDR

Hi, I'm trying to make an apparently simple thing work, but I don't see how it is possible with Asterisk. This is my extensions.conf: exten => 1234,1,Dial(SIP/123456/555-555-5555|20|D(7777)) ;After call connects, send DTMF 7777 exten => 1234,2,VoiceMail(1234@context); What I obviously want is that if nobody answer the call, go to voicemail. Basic stuff. Problem is Asterisk

Hi All, Asterisk 1.4.12 on CentOS 5 I'm trying to increment an AstDB key with the length of the last outgoing call. Here's what I've got for "01" UK geographical numbers: exten => _01.,1,Dial(${UKGeographical}/${EXTEN},,g) exten => _01.,n,Log(NOTICE,Call to ${EXTEN} lasted ${DIALEDTIME}) exten => _01.,n,Set(CALLTIME=${DIALEDTIME}) exten =>

Hi All, I'm tring to do some DB operations before and after a call. I see the 'g' option in dial to continue in context if the destination hangs up, but what if the originator hangs up? Basically I do a DB get/put before the call is placed. After the call is completed I want to do another get/put; however the dial application dies when the originator hangs up. Any way to get around

How can i configure the following scenario, - User 'A' dials into Asterisk, - Asterisk puts user 'A' on hold - Dials Out to User 'B' - Consults user B' if he wants to take the call (Press 1) or divert to voicemail (press 2) - Depending on the option chosen, either user A' call is bridged with the out call or transfered to voicemail. Thanks, Sharath Chandra

Is it possible to force the Dial function to skip to the next priority if it doesn't find the server of the called contact within a few seconds? I know I can use: Dial(Technology/resource[&Tech2/resource2...][|timeout][|options][|URL]) where I can use some short timeout in the "timeout" option, but if I do so, when some call is well succeeded, it will only ring for that

ENUM lookups at e164.org return a IP route for ALL toll-free numbers. I was surprised to observe that ALL toll-free numbers get a hit at e164.org. It appears that ALL toll-free prefixes have been delegated, thereby publishing an IP route for YOUR TOLL-FREE NUMBERS, my toll-free numbers, and even toll-free numbers that have not been allocated. :-) See below Should I care? Even though this

Hello, What i am trying to do is ..... Dail a number and ask if you wana talk to XXX press 1 and if you dont wana talk press any other key. For this purpose i am using this link<http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial> . *I am using this option :- * *M(**x**)*: Executes the macro (x) upon connect of the call (i.e. when the called party answers). IMPORTANT - The CDR

I checked out llvm from cvs & llvm-gcc from svn last night and again tonight. Each time they compiled and installed fine. After installing them, I recompiled compiler transforms I had written for opt. opt seems to load the my transform libraries fine, but it complains: opt: Unknown command line argument '-mytransform' whenever I try to specify one of my transforms on the opt

Hi All, I setup auto dial on my asterisk server. The problem is asterisk does not wait for called party to answer the call but proceed to process the extension specifed in my .call file My sample call file : hannel: local/0124787924@outbound-reminder MaxRetries: 5 RetryTime: 300 WaitTime: 40 Account: Reminder context: remindem extension: s priority: 1 Set: MSG=0135.20070601.0124787924 Set:

After updating my libtool to 1.3.5, I got ogg installed OK with: ./autogen.sh --prefix=/Local/Public/bungi/Vorbis make install But, now I'm trying to build vorbis (which exactly the same command as above) and I'm getting a failure in configure since it can't find the ogg headers. The config.log shows: configure:1888: cc -o conftest -O20 -D__NO_MATH_INLINES -fsigned-char

Hi, The pc that runs the samba server is my own property, it also dubs as router and dhcp server. I assign WINS server to the clients by dhcp using dnsmasq. I do not need or want to setup a domain. I just want to share files between the two networks using windows neighborhood and a workgroup, just like the clients would have been connected in the same network. I have deleted the "remote

Running build 1.8.5.0 (compiled from source) I seem to be having an issue with codec negotiation. I have a Grandstream HT503 FXO port connected to a pstn line, a Polycom SP501, and a SIP trunk with callwithus. What I'm essentially looking to accomplish is for ulaw or g729 (preferably ulaw) to be used to the Grandstream FXO or any other internal endpoint, and for g729 only to be used outbound

PR1317 says that it is resolved. Looking at the details of the report, it says that a fix had been committed for the "instant case." Am I right that means a fix has been submitted that makes LLVMHello work with opt, but that the general problem has not been resolved and is reported in PR1318? Reid Spencer wrote: > Hi Ryan, > > On Tue, 2007-04-10 at 00:12 -0500, Ryan M.

Il 19/02/20 10:22, Rowland penny via samba ha scritto: > On 19/02/2020 08:05, Roberto Tagliaferri - Tosnet srl via samba wrote: >> Sorry for the english :) >> >> Good morning, i've a little network with samba 4.2 on centos 6, NT style >> domain and win7 client. > > Excuse me, but do you not believe in updating things ? > > Samba 4.2.x went EOL at the end

Hello all, I am testing the Dial application with the fall through priorities for different cases what i want is the flow after failure of the Dial application which simulates response codes like 1)404 -- Not found 2)480 --Temporarily Unavailable 3)486 --User busy i did manipulate the priority flow like the following for the case 2 and 3 ... exten => _XX,1,Dial(SIP/extension) exten =>

Hello, I would like to implement a few decision making process inside the dialplan using information stored in MySQL (like LCR, etc.). I see the MYSQL() application, but as far as I understand I have to connect to the database each time I want to query it; this seems a CPU eater to me. Is this indeed the case, or can I open it once Asterisk starts and leave it open?

Hi, I am trying to get priority + 101 to work with Dial application. My dial plan is like this: [dial-mobile-peak] exten => s,1,AbsoluteTimeout(${ABSOLUTETIMEOUT}) exten => s,2,Dial(${TRUNKONE}${CALLEDNO:1}) exten => s,103,AbsoluteTimeout(${ABSOLUTETIMEOUT}) exten => s,104,Dial(${TRUNKTWO}${CALLEDNO:1}) I have changed password for first trunk to simulate trunk failure. Trunk one