similar to: AGI onAnswer function: does it exist?

Hello, Does anyone know any solution to this? Or is Asterisk-Dev a more suitable list to ask this question? Best regards, Vlasis Hatzistavrou. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Vlasis Hatzistavrou Sent: Thursday, February 16, 2006 3:43 PM To: asterisk-users@lists.digium.com Cc: 'Vlasis

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Hello, On an Asterisk 1.4.33.1 in a simple scenario: [test] exten => _X.,1,Dial(SIP/12345 at peer01,,,) exten => i,1,Hangup(${HANGUPCAUSE}) exten => t,1,Hangup(${HANGUPCAUSE}) exten => h,1,Hangup(${HANGUPCAUSE}) I have noticed that no matter what value we set in the Hangup(<cause code>) commands, if the call is not answered by peer01 for any reason, the actual cause code

Hello all, We have been testing Asterisk RC2 with the native H323 channel driver. We followed the instructions with the needed OpenH323 and PWLib versions and everything compiled ok. Operation of the driver seems ok, except from 2 main points: 1) Audio is passed between the two ends of the call only after the call is answered. This was not the case with previous versions of Asterisk (0.9.2

I'm using get_data in an AGI script and am having a problem when, after a long time in my IVR, when I ask for a 10-digit phone number, the first few tries are always invalid -- the number it reads back is very strange, almost like the DTMF tones from other answers were being cached and then dumped on the call to get_data. Anyone ever experienced this before? I have to do some major

Hello, I'd like to ask if there are any news about CE certification of the E1 boards. I know that the T1 boards are FCC certified but I'd also like to know what is the status for CE certification. Thanks for any input, Vlasis Hatzistavrou.

Hello, I just tried to compile OH323 0.6.3b on a RH9 machine with Asterisk 1.0 RC2 installed but failed. I applied the patch to the required OpenH323 library according to the instructions, and set the proper directories in the Makefile. Here is what I receive after I issue make: ******************************* g++ -DP_USE_PRAGMA -fno-rtti -ffunction-sections -fdata-sections -D_REENTRANT -

Hello, I have used the Dial option 'r' before in older Asterisk versions and it behaved as expected, that is, Asterisk would always give ringback audio before the call was answered no matter what. It has been a while that I have used version 1.4.33.1 without any the 'r' option. Now that I had to use it for a while, I noticed that 'r' would not give ANY audio until the

Hello all, Sorry for the slightly off-topic issue, I need to have a capture from a network sniffer (like Ethereal for example) from a call setup with the MGCP protocol. I thought that since Asterisk now supports MGCP some of the people who develop the MGCP channel driver may have such a capture available. I need this for my MSc thesis and unfortunately, I don't have any MGCP compliant

Hello, I would like to ask if anyone has solved the problem with Asterisk+ISDN4Linux cards, where there are no call progress tones or announcements from the PSTN when we dial ouot through the i4l card. For the moment, if we don't inject the r option in the Dial command, there is only silence during the call negotiation... Using Asterisk RC2 with Eicon passive PCI 2.01 card... Thanks for

Hi, A glance at sourceforge.net/projects/openh323 Help Forum made me wonder if this location is the one to use (I got trouble in the past when google pointed to an obsolete site) : some quite old messages remain unanswered. Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL:

Running Asterisk Head 1.0.9. Below is a trace of a call delivered to my system which had no caller ID. For some reason, dialparties.agi shoves the incoming provider's IP address into the caller ID so you never have a call that is screened for PrivacyDirector. Is anyone else seeing this issue as well? Have I missed a patch? This call shows on the display with a name of "Unknown"

Hello List, This is my first message herein. I was playing around with System() and AGI() and found out something I cound not determine my configuration error. I added before.agi and after.agi to the agi-bin dir. Tried to make before.agi get run before the dial call and after.agi be run after. Only the first priority (step 1) gets executed. Here follows some relevant part of the tests: On

I edited the calleridname.agi patch to only overwrite the name if it is missing. The asteridex option still overwrites the name since it is our master list for known numbers. -- Steven calleridname.agi.patch: --- C:\Documents and Settings\steveb\Desktop\calleridname.agi-orig Tue Jun 13 14:37:09 2006 +++ C:\Documents and Settings\steveb\Desktop\calleridname.agi Tue Jun 13 14:37:09 2006 @@ -16,6

Hi, I have a question about this here: Asterisk-Version: 11.10.2 File: res/res_agi.c Line: 2377 [...] static int handle_recordfile(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[]) 2304 { 2305 struct ast_filestream *fs; 2306 struct ast_frame *f; 2307 struct timeval start; 2308 long sample_offset = 0; 2309 int res = 0; 2310

Hi Everyone, Trying to run a php script after DeadAGI for A2Billing does it's magic. This is the dialplan: [a2billing] exten => _X.,1,System(php pre-call.php ${CALLERID(num)} ${EXTEN} ${UNIQUEID}) exten => _X.,n,AGI(a2billing.php,1) exten => _X.,n,Hangup() *exten => h,1,Wait(5)* *exten => h,n,System(php post-call.php ${CALLERID(num)} ${UNIQUEID})* As you can see above, I even

I am working on a call-back solution where the initiating call should never be answered. I was doing this simply through the dial plan, sending a progress tone, and then dumping the channel, and firing off a DeadAGI which created a call file to make the callback. Now I've tried extending this so that an AGI is fired first to check for things - like no inbound ANI - and play a

Hi all, Tiny, but very important question for me: what it can be when standard agi-test.agi script, like: print STDERR "Testing 'waitdtmf'..."; print "WAIT FOR DIGIT 10000\n"; my $result = <STDIN>; &checkresult($result); on the call from KPhone application does NOT return any DTMF code back (I use dtmfmode=inband in sip.conf): Testing

Hi, "Asterisk 1.8.10.0-1digium1~squeeze built by pbuilder @ nighthawk on a x86_64 running Linux on 2012-03-08 23:05:09 UTC" We have some problem when running a AGI script (build with PHP), existing channels (all of them) gets a "hickup" and then continues. We are using AGI to lookup incoming calls in directory. It is kinda annoying, and I don't understand how it can be

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