similar to: asterisk 1.6 and mISDN

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, 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 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, I read that installing asterisk on Solaris is supported. Does anyone of you actually have experiences with that? And especially, does anyone of you have experiences in runnning asterisk with misdn unter Solaris? Thanks and regards, Christophorus

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, 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 list, is it possible to get in the running dialplan the status of (SIP) lines without using AGI or anything like that? What I want is a stepwise calling: I have several SIP lines (let's say they are three) which I want to dial to alternatingly. But I do not want to dial to a already busy line and catch the busy. Instead I do not want to dial to that peer but to the next one. I want to have

Hello I fixed my problem. I changed user and group in /etc/mISDN.conf: <devnode user="asterisk" group="asterisk" mode="644">mISDN</devnode> now it works again! Thanx 4 help! ;) Ingrid -- "Bonnie & Clyde der Postmaster-Szene!" approved by Postfix-God http://wetterstation-pliening.info http://dokuwiki.nausch.org --------------

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

Hello, I'm evaluating to possibility to use chan_misdn as a short term workaround, in case latest Dahdi is not stable enough for what we are planning to do (we wish to use Junghanns and Digium BRI hardware with Asterisk 1.6) . I've read www.mISDN.org but still have a couple of questions : 1. Is correct that in a 2.6.27 (and up) enabled kernel, the "embedded" mISDN version is

Hi, I try to compile mISDN 1_1_9_2 (kernel: 2.6.32.13). I had the following compiling error: make[1]: Entering directory `/usr/src/linux-2.6.32.13' CC [M] /usr/src/mISDN-1_1_9_2/drivers/isdn/hardware/mISDN/sysfs_obj.o /usr/src/mISDN-1_1_9_2/drivers/isdn/hardware/mISDN/sysfs_obj.c: In function ?mISDN_register_sysfs_obj?:

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

Hi All I have noticed that when calling over mISDN channels the call itself is made over the channels listed in misdn.conf eg: 1,2,3,4 as misdn/1, misdn/2, misdn/3, misdn/4 etc. However, when looking through the CDR's the call gets logged as misdn/0, misdn/1, misdn/2, misdn/3, misdn/4, misdn/5 etc. I also noted that busy, failed, congested calls link mainly to misdn/0. The only logic I

Hi, I'm installing Asterisk on Ubuntu 6.10 When I first compiled the zaptel package I used: make clean make make install So far so good, but the following command failed: make b410p I did some digging on google and found a guide on how to install it manually, but the result was the same. I got these files ftp.digium.com/pub/telephony/zaptel/b410p/misdn-b410p.tar.gz

Hi, Before diving into this, I would very pleased to know if someone could yes or no, successfully compile mISDN 1.1.8 on Lenny (latest RC1 or beta2 version) ? Regards After a fresh install on Lenny, I can reproduce at will : apt-get install build-essential linux-headers-2.6.26-1-686 cd /usr/src wget http://www.misdn.org/downloads/mISDN.tar.gz tar xvf mISDN.tar.gz cd mISDN-1_1_8 make ....

Hello, I have a strange problem with misdn.conf When I run asterisk, I have this message : Apr 3 23:00:25 WARNING[6824]: misdn_config.c:579 _build_general_config: misdn.conf: "use_callingpres=yes" (section: general) invalid or out of range. Please edit your misdn.conf and then do a "misdn reload". Apr 3 23:00:25 WARNING[6824]: misdn_config.c:579 _build_general_config:

Hi, I'm bumping on this : cd /usr/src wget http://www.misdn.org/downloads/mISDN.tar.gz tar xvf mISDN.tar.gz cd mISDN-1_1_18 make <snip> In file included from /usr/src/mISDN-1_1_8/drivers/isdn/hardware/mISDN/core.h:9, from /usr/src/mISDN-1_1_8/drivers/isdn/hardware/mISDN/avm_fritz.c:20: /usr/src/mISDN-1_1_8/include/linux/mISDNif.h:791: error: field

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