similar to: asterisk and mISDN on Solaris

Hi there, I want to set up an extension set that acts different depending on the count of used lines. I have a EuroISDN E1 board with mISDN and I only want to offer 10 lines. Therefore I set up a global variables LINES in the general section of extensions.conf and instantiate it with 0. I a call is incoming I check the LINES variable wether is 10 or more. If so I make a call transfer. If not

Hi list, I am searching for a possibility to do a certain call transfer method which is called "path replacement" in QSIG. But I want to do that in DSS1 (EuroISDN). If my asterisk does a call transfer I want the machine to signalize on dchan that the call path has to be replaced to a direct connect between the caller and the called, i.e. my machine is to hang up after the transfer and

I have just got a Cisco 7941G and am experiencing the exact same problem (phone is requesting .tlv file from TFTP server and never asks for .cnf.xml file). The phone originally had SCCP on it, but I downloaded and flashed with the latest Cisco SIP image (8.4(3) released 2009-01-13). In reading your message below, it looks like you were going to try an incremental upgrade?did you have any

Hi on the list, does anyone of you have experience with asterisk 1.6 and mISDN, pri primarily? Thanks in advance & Regards, Christophorus

hi list, I set up a new asterisk machine with asterisk 1.2.13 and misdn 0.3.1rc27. I use an e1 card with sip clients. My extensions look like this: [E1] <snip>...<snip> exten => 33006733,1,Set(CALLED=${EXTEN}) exten => 33006733,2,Dial(SIP/1@192.168.0.23) exten => 33006733-ANSWER,3,Answer() [SIP] exten => _X.,1,Noop() exten =>

Hi, How can I integrate Asterisk to Nuance TTS engine instead of Cepstral? Has anybody done this? How is the architecture and can Java AGI be used to communicate between them? regards, Vela Sivasankaran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091021/56254a0e/attachment.htm

Hi, does anyone of you have made it to get the ANI also picked up? I mean: if I fetch a foreign call to me by using the pickup application I want to see the callerID/ANI of the caller to the foreign extension. Is that possible and if yes - how do I achieve that? Regards, Christophorus

Hi list, I have bought some Cisco 7941G-GE IP phones and want to use them with asterisk. Before bying I tested the whole setup with three different models of the old 79X0 series (a 7912, 7940 and a 7960). Flashing the formerly provided SCCP-Image to SIP was no problem, but now it complains about a nonexistent CTLSEP<mac>.tlv file. Most of the howtos say something about an empty file but

Hi, Until now I've been using my Digium B410P cards with misdn 1.0.x. I would like to upgrade my systems and am now wondering which is the "best" route to take: - use the latest release of misdn v1 - upgrade to the latest "stable" kernel and use the built-in misdn v2 - use misdn v2 as a seperate package (disable misdn in the kernel) - use dahdi's support for misdn

Hi i've installed this software: ******************** SOFTWARE mISDN-1_1_7 mISDNuser-1_1_7 Asterisk-1.4.15 ******************** SOFTWARE misdn is correctly loaded by misdn-inist start Here there is the misdn.conf (copied from an existing and working installation with Asterisk 1.2.x and one BN8S0) ******************** MISDN.CONF [general] misdn_init=/etc/misdn-init.conf debug=0 bridging=no

I would like to know if the following misdn warnings are relevant. Currently, I don't need echotraining. However, I took a quick look at the * source code and l1watcher_timeout seems to be defined (echotraining was not found). Currently I'm setting l1watcher_timeout to 0 which is default (so I suppose that this warning won't affect me). Any comments on this? *CLI> misdn reload

Hello, After upgrading my kernel to mISDN-1.1.1 while keeping asterisk-1.2.16 I no longer match any extension. Apparently the "dad" is empty. However I can see the number just before it (146472130): P[ 4] I IND :SETUP oad:!?145201798p ?146472130 dad: ?146472130 pid:2 state:none P[ 4] EXPORT_PID: pid:2 Mar 23 09:35:28 WARNING[6725]: chan_misdn.c:4750 chan_misdn_log: Extension can

Have you got termination set correctly? I have a B410P working with 2 x NT and 2 x TE ports successfully. I had to turn the 100ohm termination on on the NT ports (even though I have them set as PTP in mISDN.conf). HTH -->> -----Original Message----- -->> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- -->> bounces at lists.digium.com] On

Hi list. Asterisk GUI (current version) does include the menu required for setting up the mISDN Trunks ("mISDN Config"). But how about using those. Currently that seems not to be supported by Asterisk GUI or did I miss something? Whilst adding a new user extension I may select SIP, IAX and Analog "Technology". "mISDN" is not an option that may be selected. Is the

> uname -a Linux pbx 2.6.15-1-686 #2 Mon Mar 6 15:27:08 UTC 2006 i686 GNU/Linux I'm using the 2.6.15-1-686 kernel package from debian testing. mISDN was installed using the latest CVS sources. Regards, Chris --- From: "MBIT Technologies" <mail@mbit.com.au> Subject: RE: [Asterisk-Users] Asterisk, mISDN and a Fritz card -- kernel crashes To: "'Asterisk Users

Who mentioned fax? :-) Native CAPI isn't an option for us (unfortunately), as our service is a point-to-point service, which the AVM CAPI drivers don't support. We've now got another problem -- we're now able to make calls -- once. The kernel panics as soon as someone terminates a call (and, so it would seem, at various other times too). This has only been occuring since mISDN

Hi, I have installed Asterisk 1.4 with mISDN with the install-asterisk.tar.gz script from beronet.com. On my system I have two cards, one a AVM Frit!Card Pci 2.0 and one HFC-S chip. I know both to work well with mISDN on my system from a previous installation. Now however, the AVM card works well at first glance, i.e. it "registers" incoming calls and works through the asterisk

recently i've upgraded asterisk from 1.2.4 to 1.4 All works fine but i'me experencing some instability on misdn channels. In the last week i've experienced twice some problems with misdn (I am using mISDN-1_0_4) dmesg output: mISDN_rdata: rport queue overflow 256/256 [addr:52020501 prim:120282 dinfo:ffffffff] mISDN_rdata: rport queue overflow 256/256 [addr:52020501 prim:120282

Hi, I have an Asterisk 1.2.9.1 box on a Debian distro with mISDN drivers. I installed the new driver (0.3.1-rc30) on our pbx but since no voice was passing I decided to go back to old version (0.3.1-rc23). Last friday everything seemed to work fine but now every incoming call drops after 3-4 seconds while Asterisk console is showing these messages: Apr 23 12:42:39 DEBUG[7625]:

Hello everyone! I'm sorry I can't be more specific. So here's the setup: a Samsung router with analog and ISDN ports. the phone company says the outgoing line is analog landline, but I'm sure it's some VOIP. so connected to the ISDN port of the router is a Fritz AVM card, used with mISDN. when I try to make a call with asterisk I get something like this: cli>>