similar to: Problems with Call Progress and fax detection on PRI

> Hi Gareth Blades & Doug, > > Thanks so much for for the feedback. I have searched on lot of documents > but couldn't able to find clear answer regarding it. > > I hope you guys replies are very much help all in aterisk community. > > > Thanks & Regards, > > Vidura Senadeera, > > Network Engineer, > > Debug Solutions > > Sri Lanka .

I want to say thanks to everyone for the help so far. I figured out a way to modify some AAH code that worked for me (well sort of). The line I modified is s,14 in macro-dialout-trunk. Then I just added a variable and passed it from 9_outside. I just have one last problem. This waits for an answer not ringing. So if the called party has a long ring to voice mail the call is dropped and goes

Anyone know how to let asterisk deal with the progress codes coming from the carrier? The problem I am having is when a customer calls an invalid number the carrier tells me the call is invalid via a progress code but doesn't route me to a recording (this number is invalid). Instead they hang up on me causing a fast busy or sometimes hold up the call with dead air for 15 to 30 seconds then a

Thanks Dovid! Indeed looks a bug but regardless of this, this problem made me think that the HANGUPCAUSE could be used for this purpose with benefits. I couldn't find an explanation about when DIALSTATUS would actually be better. The HANGUPCAUSE was reworked in version 11 ( https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause) but I didn't find someone actually stating it is a better

I have a PRI switch type national Asterisk 1.2.16 Zaptel 1.2.15 If I call an invalid number I get * PROGRESS with cause code 28 received Asterisk continues to attempt to connect the call until the timeout is reached and I hear ringing. I want to capture the progress code, which I thought was in HANGUPCAUSE but when I NoOp that variable it's always 16 when I dial an

Hi all, I'm using Asterisk 1.0.9-BRIstuffed-0.2.0-RC8o on box A with a TE410P (EuroISDN cpe) connected to another similar asterisk box B acting as EuroISDN master. I'm performing some load tests by contiously feeding up to concurrent 30 call files to /var/spool/asterisk/outgoing/ on box A which inititate via a dialplan context/extension a outbound call (redirected via chan_local) to

Hello list, Hope all doing well! I've been checking some cases when a Dial fails and dialplan execution continues to handle this. I am finding it a little confusing how we should handle the DIALSTATUS and the HANGUPCAUSE in this situation.... More specifically, I am facing a case in version 13.6.0 where I am getting a DIALSTATUS=BUSY and HANGUPCAUSE=19 after receiving a 480 SIP error. Seems

Just throwing this out here, hopefully someone can tell me why. *CLI> show version Asterisk CVS-HEAD-11/17/04-10:16:38 built by root@wanderer on a i686 running Linux Zap/g1 is pri_cpe to Bell Canada 5551234 is a normal POTS line I have busied out (handset offhook) exten => 1234,1,Dial(Zap/g1/5551234,,g) exten => 1234,n,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is

Hi all, Hopefully, I just need a second set of eyes on this one, but I just can't figure out what I'm doing wrong. I'm using an agi script to dial a number, check the dial result, and act accordingly. The problem is that I'm not getting anything back from DIALSTATUS, or HANGUPCAUSE. Here is the relevant perl code: ===============================================================

I'm struggling to get my dialplan to work with a simple analog fax machine. I have TDM400B zaptel card with an FXO and FXS port. I have the FXO port connected to the POTS machine and the FAX machine connected to the FXS port. The FAX machine itself works fine, I can FAX outgoing messages fine. I can also dial the FAX extension from the internal context, the FAX machine answers and I

I'm trying to write a dialplan that will allow me to "stress" test it. I want to be able to dial an extension, or pretend that the extension is busy or out of order (so that I can see what to do) given the dialplan snippet: [outbound] exten => _X.,1,NoOp(${TEST}) exten => _X.,n,Dial(SIP/${EXTEN}) exten => Busy,1,Busy(2) exten => Busy,n,Hangup() exten =>

Today I get the remark that a call got disconnected after 10 minutes. This what my VERBOSE-logfile tells me : [May 18 15:36:30] VERBOSE[3940] logger.c: -- Executing [00493516426 at intern:1] NoOp("SIP/51-b76023b8", "Gesprek naar GSM-nummer via Telenet") in new stack [May 18 15:36:30] VERBOSE[3940] logger.c: -- Executing [00493516426 at intern:2]

Hi, I am using Asterisk: 1.6.1.20 LibPRI: 1.4.11.4 DAHDI: 2.3.0.1 Echo Canceller: MG2 Wanpipe-Driver: 3.5.15 Sangoma-Firmware: 43 (A104d) I handle some calls with my own PHP-AGI-Script. After a dial-command I use "GET FULL VARIABLE ${answeredtime}" or "GET FULL VARIABLE ${dialstatus}" and get valid information. Sometimes "dialstatus" has the value

I am seeing some curious behaviour with a 1.2.32 system, which I do not understand and so can't work out how to fix it. I have a PRI routed to context default. Here is the complete default context: [default] exten => _9X.,1,Dial(IAX2/m1peer/${EXTEN:1}) exten => _20XX,1,Dial(IAX2/sipeer/${EXTEN}) exten => _X.,1,Dial(IAX2/m1peer/${EXTEN}) exten =>

Managed to implement this on asterisk v1.4.24.1, Also, Hangupcause updating to user field. However, this only works on the edge of my voice network (demarcation point) It does not work on my internal routing boxes as I use IAX to route between remote sites. I was thinking of using some sort of SIP variables to transport these results over the IAX trunk.. Any bright ideas folks???

Hello list. We just upgraded to 1.6.1.11. We are using real time information stored on mysql databases. That is all running fine. Now, since we upgraded, some member don't get calls from queues. In CLI: "queue show" shows something like: 611 (Local/611 at agents) with penalty 20 (realtime) (*In use*) has taken no calls yet We use the extension 611 in different computers, in the

Same thing here. I had this problem awhile ago and made this workaround. Going to another trunk does not work because they are answering and not sending a error code. If you are using AAH code then this waits 10 seconds on your Voip then times out and goes to PSTN. You can modify for your needs The pertinent line is 14 in macro-dialout-trunk I am going to clean it up and repost under my

Hi folks, I was hoping that someone might be able to help clarify some confusion I have on DAHDI Fax detection after spending some time searching. My understanding is this: 1.) Echo cancellation is automatically disabled upon recognition of a CNG tone, regardless of the faxdetect setting. This can only be disabled at compile time. 2.) faxdetect=incoming will, upon detection of a CNG tone,

Hi All! Let me explain the problem. When using the Originate? command from the manager api, the dialstatus variable returns results? for whichever phone picks up first, and in this case it is the IAX/2? connection. It doesn't matter if Zap/G2/XXXXXXX is set as the channel,? or an extension either. What I am ultimately trying to do is get the? dialstatus of the Zap/X/XXXXXXX channel, i.e.,

Hi, I'd like to know what's the most "popular" method for automatic fax/voice detection for incoming calls on mISDN cards such as the B410P (hfcmulti). I'm running: kernel 2.6.17 misdn 1.1.3 asterisk 1.4.21.2 B410P card I'm using iaxmodem and hylafax with asterisk (the setup works for zap channels). I've used the following options in /etc/asterisk/misdn.conf: