similar to: Réf.: Re: Asterisk with UUI support ?

right, so far, here is what I've done: I succeed in take in a new variable the UUS1 field sent with the connection request for incoming calls. It was quite simple afterall... (I just had to find where the data CMSG->Useruserdata is coming in chan_capi.c) Now I would like to know where this field is instanciated for outgoing calls in order to control this step? I am looking for that but I

Hello List, I'm trying to get a PrimuX Card (www.primuxisdn.de) working. The Manufacturer says that chan_capi (the older one) used to work. Now I'm trying with chan_capi-cm and have got the following problems: Outgoing calls "ausgehend_ueber_ntba_primux.txt" the other phone rings but when I answer the call asterisk crashes. Incoming: first tried with immediate=no

OK, so I'll do that... Is there any infos I need to know about chan_sip.c (because I suppose it's it that I need to play with)? Does anyone know an interesting website where I can find infos about UUI in ISDN (with CAPI maybe?) ? Thanks for your help.

Hello everybody! happy new year ;-) I have a problem with inbound calls on my Diva Server PRI This is my problem: 2005-01-05 15:38:04 ERROR[31716]: chan_capi.c:1696 pipe_msg: did not find device for msn = 4132 capi.conf: [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 language=de [interfaces] msn=4132 incomingmsn=* controller=1 context=demo mode=immediate

Hello, i have asterisk with chan_capi ( AVM Fritz card ) working, isdn line BRI, setup to point-to-multipoint. working fine, but when both channel on interfaces are used and somebody wants to call in - the response is "ha-la-li" - sounds like msn or called number does not exists. i think that right response should be - line busy tone... right ? can anybody tell me what is wrong ?

Good day, The unusual thing about this problem is that it doesn't occur just during a CAPI call, or just during an IAX/SIP call. Only during IAX/CAPI I'm having some trouble with the CAPI interface and it only occurs when a call comes in on an IAX channel and goes out the CAPI interface. The capi debug in the asterisk console is below as well as the relevent parts of .conf files from

Hi all, i have asterisk 1.0.9 with an Eicon Diva 4bri and chan_capi-cm-0.6. I have 2 NTBAs (one with did and one without). When using the one without did, i am able to place outgoing and incoming calls. When i use the NTBAs with did i have a layer 2 error. Anyone an idea? -- Executing Dial("SIP/2004-9634", "CAPI/g1/43XXXXXX") in new stack > data = g1/43XXXXXX

Hello everyone, I have been using an AVM Fritz! card with chan_capi and mISDN for quite a while in PTM mode and it was working finely. Now, I needed more DID/MSN, so I switched to PTP. But now nothing works anymore :( I am using Asterisk on Debian Sarge stable and installed Asterisk along with chan_capi from apt-get. I installed mISDN from the CVS of isdn4linux.de. It is : - Asterisk

Hi there, Is it possible to manage UUI with asterisk and ISDN (T0 Fritz card). Basically, is it possible to send User to User Information using the D-channel, while making a call?

Dear all, How do I set the UUI informations for outgoing calls and read the UUI information for incoming call in asterisk? Thanks in advance.. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100531/ffcceeee/attachment.htm

Hello List, I need to connect with an Avaya PBX (this part is done), and i would like to get and send back User-to-User Information (UUI) with the call. The UUI need because I need to identify the call based on something witch is available on Asterisk and Avaya too. It is possible, or have a better solution? Anybody did it before? Thanks for the help! Zsotya

Hi, I came thru ISDN UUI (User-User Information) protocol which is defined in this RFC - http://www.ietf.org/id/draft-ietf-cuss-sip-uui-17.txt But I don't understand how to use this with Asterisk. Any idea would be much appreciated. Thanks. Gopal. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello configuration as follows, dial-out works: capi.conf: [hfcpci] ;;PointToPoint (55512-0) isdnmode=MSN incomingmsn=* ;msn=61 controller=1 devices=2 context=incoming extensions.conf: [incoming] exten => _XX,1,Playback(demo-abouttotry) exten => _XX,n,Dial,SIP/xlite1 exten => _XX,n,HangUp When call is placed, the following debug info is shown, after the last line, it stalls until

Hello Armin, hello List I'm trying to get chan_capi working with asterisk from debian stable (asterisk 1.0.7, the debian version number is 1:1.0.7.dfsg.1-2). I managed to get it compiled by providing my own version of ast_copy_string. This is an Austrian PTP line. I can do outgoing calls fine (no comprehensive tests yet). For incoming calls, I'm getting "No answer" on the

I have a Diva Server BRI installed in a Debian system running Asterisk 1.2.22. When a call comes in the Background application plays a couple of times. Halfway through the second time, the call is disconnected (total time connected about 25 seconds). I've posted various configs below. Any advice on how to debug would be appreciated. We are located in New Zealand. CLI showing capi debug (just

Hello, I need to detect both fax and answer machine, and it should be valuable that the detection will be run by the Diva card itself. So : - I read Diva Documentation, and I found that the Diva could send some specific DTMF, if I had "[..] enabled [this functionnality] by the application for a designated controller through a manufacturer request command 9 [...]", but I didn't

Hi everybody, I have a problem which I cannot eliminate on my own. Has anybody any idea for the following: I am using the asterisk-version from Debian-Testing (1.2.13) with the latest chan_capi (also tried an older version). When using the Capi-Channel, everything works fine except from the sound it sounds extremely choppy and is unusable :-( When e.g. capisuite is used for fax, everything

Hi, I am having a problem with (Asterisk is crashing) with a Fritz card PCI / chan_capi. Receiving Calls from PSTN works, but outbound calls make asterisk crash (Speicherzugriffsfehler/Segmentation fault). The crash occurs upon dialing with the other phone not even ringing. I hereby ask if somebody reading this list can confirm or disprove my issue. Does anbody run a recent asterisk 2.6 with

I have a problem receving calls via the ISDN line, using the followin components Asterisk 1.0.9 with asterisk@home chan_capi-cm-0.6 AVM Fritz card datalink protocol = point to multimode I can make calls out with no problems so the issue is only incoming calls. When I make the call from an external line to the ISDN line connected to asterisk, I get a busy signal after about 5 seconds. I have

I tried that but it still doesn't work... I think I don't have the correct approach. Have I to install any ISDN drivers (=modules ?) BEFORE dealing with CAPI ? If yes, why shouldn't I use the hisax drivers (which are kernel ones) instead of fcpci drivers (which doesn't seems to work, by the way...) And finally, how is it possible to link the two modules together? As you can see,