similar to: ISDN call-progress IE in SETUP frames

Hello, I have been running some tests to get a better understanding of PRIs and the HANGUPCAUSE variable and I'm not having any luck. I have tried calling disconnected numbers and the call is connected through to my extension and I hear the tri-tones. And it looks like HANGUPCAUSE is always 16 (AST_CAUSE_NORMAL_CLEARING). Am I doing something wrong, or am I just misunderstanding? Also,

Hi, We've exhausted our internal capabilities as well as Sangoma tech support and were hoping someone with some expertise could help us with a pointer. Briefly, our issue is as follows. Periodically (several times an hour), we get either of the following error messages in our asterisk messages log. These correspond with dropped outbound calls on a one-to-one basis when the second error

downloaded and compiled today's CVS (04/08/2004) tried using callprogress on Via mini-itx (running RedHat Linux 9) if callprogress is set to yes on x100p, an i call the line connected to x100p, asterisk would execute the first app and will wait forever. anyone had success using callprogress? thanks. __________________________________ Do you Yahoo!? Yahoo! Small Business $15K Web Design

Dear All, I am trying to setup an ISDN line from local telco on a digium card. The problem I am facing is that I am not getting any caller id from the telco. They say that they have enabled caller id. Please help me out. My zapata.conf -------------------------------------------------------------------------------------------------------------------- [trunkgroups] [channels]

I'm attempting to use callprogress in my system, and I'm having trouble. Callprogress always can tell if the line is busy or ringing, but when the line is answered, the call does not get bridged. Messages showing that "line is ringing" stop in the console and if the called party hangs up, asterisk reports the line is busy. Are there any settings that I could use to help with

Hi, As you may know, these ISDN BRI features are very important here in Europe as ISDN Basic Rate Access is very popular among Small & Medium Entreprises. I don't really know why but it seems that in many countries, default is to install small PBX using Point-to-Multipoint (PtMP) mode as opposed to Point-to-Point (PtP) which is the norm for PRI. So basically, in several countries, SME

Hello, I recently bought a Junghanns Octobri Card. I have some problems with this card to make outbound calls but I can receive calls. I have 3 lines to PSTN and 3 lines to my existing PBX FRANCE TELECOM <-- OctoBRI --> Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1h <-- OctoBRI --> PABX e-Generis <----> ISDN Phones |

Hi to all, I'm using a TDM22B. When i establish an external call to the PSTN through an FXO port, I'm not able to know the status of the call (no answer, busy, ...). If I enable call progress (callprogress=yes) in Zapata.conf, I am able to detect the no answer state but if the callee on the PSTN answers the call asterisk doesn't detect that and it jumps to the NOANSWER state and

Hello, I haven't seen any recent posts on call progress detection, so here's a question: How would one accomplish an automated outbound dialing application using *, whereby a requirement is to wait for the greeting to complete (live person, answering machine, voicemail) before delivering the message? For example, playing a 'reminder' message to a list of recipients. I know its

Hi, I've seen this ISDN subaddress feature added to libpri. Which countries are using it ? How is this billed ? Do you have to pay an extra to your telco to benefit from this subaddresses ? Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090925/da0771bb/attachment.htm

I have some data that looks like this: > dataP input output corpusFreq pvolOT pvolRatioOT 1 give(my sister, the old book) P 47.0 56016 0.1543651 5 donate(her, the book) P 48.7 68928 0.1899471 9 give(my sister, the book) P 73.4 80136 0.2208333 13 donate(my sister, the old book) P

Hi, There seems to be no solution for call progress on analog lines and using outgoing spool call files . My wave file starts playing before the person has answered the phone so the first part of the message is missed. Can the backgrounddetect app be used for this. I have tried but the message still plays before I answer. I generated 60 seconds wave file. [callprogress] exten =>

Hi All, Thanks for every one who helped me on this regard. I think i was able to rictify the problem. what i did is remove callprogress=yes usecallinpres=yes and restart asterisk. Today i didn't report any drop calls. Many thanks for Eric. :) I hope this situation will continue. Regards, Vidura. On 3/8/07, Vidura Senadeera <vidurased@gmail.com> wrote: > > Hi, > >

Hi list, I'm wondering if it's possible to transfer a call that is still ringing??? Actually, the problem is that my telco provider doesn't offer an uniform method for answer/disconnection supervision, and by that I mean, some of it's line (I think) offer a polarity reversal, but other lines (of the same service provider) do not offer anything at all, so the answer of a call

BUG/SYMPTOMS: 1.Under certain circumstances, octoBRI (and most likely quadBRI) ISDN cards (Junghanns/CologneChip) severely distort certain ISDN payload. 2.Although these claims relate to the bristuff patch, the problem might not be limited to bristuff and in fact be rather asterisk/zaptel/libpri related. 3.Signal distortion is limited to the use of the CVS version of the Bristuff patch for

Hello all! I have tried and tried to resolve this one to no avail. Hopefully one of you can help... The system in question is a Compaq Evo D600 (iirc) business desktop, with a 1.4GHz Pentium 4 and 512mb of RAM, running a stock install of PoundKey 1.2. It has two Digium cards installed: a TDM400P with four FXO modules, and a TE110P hooked to a Carrier Access Adit 600 which serves 8

Asterisk gurus, To have analog call progress (which as far as I know asterisk does not have right now) does it not come down to 4 states to detect - and I hope that some asterisk gurus can implement those 4 states easily. I see this as 4 states: Busy - a 50% duty cycle Ringing - a % on duty and MORE of a bigger % off duty cycle - or basically not BUSY and not TALK detect Talk - something

I'm experimenting with asterisk. This is my environment: - Debian sarge (vanilla kernel 2.4.29) - Asterisk 1.0.7-BRIstuffed-0.2.0-RC8g - Two sip phones (One cisco 7905 and one soft-phones X-Lite) - Digi International Datafire Micro V (Europe) (rev 02) (zaphfc) After two days of work now I can call between the two sip phones and I can call from sip phones to outside numbers via ISDN card.

Hi, I would appreciate some advice on the following: how does one use a single BRI port HFC chipset based ISDN cards with the latest DAHDI, libpri and Asterisk 1.8? For Asterisk 1.4 I would first install mISDN & mISDNuser and then build Asterisk 1.4. Now I noticed that Digium's HFC chipset based B410P is supported by the latest DAHDI & libpri but reading the source of the

I have 2 Digium 4 port FXO cards in my system. The system is a P4 2.4Ghz, 512MB RAM, Promise FastTrax 100 TX2 Pro Raid, 80GB RAID1 for storage - whitebox - running RedHat 9. With pretty much any CVS HEAD version we are getting, what I will call, "phantom" calls on some lines. What I mean by a phantom call is that the line will ring, Asterisk will log that the Zap channel has been