similar to: mISDN problem

Hello, I have some problems with mISDN. I can't send or receive call from the Billion ISDN card Mi configuration files are thoose: extensions.conf: [general] static=yes writeprotect=yes [SOME] exten => 101,1,Dial(SIP/101,30,Ttm) exten => 101,2,Hangup exten => 102,1,Dial(SIP/102,30,Ttm) exten => 102,2,Hangup include => outgoing [outgoing] exten

Hello everybody, I have installed the Billion ISDN on a Debian machine. I proved to call with a ISDN telephone conected to ISDN Box and it is OK. So I connect the Billion ISDN to the ISDN Box and I call from a extension to outside. But it doesn't work, that is what I have in the CLI: *CLI> -- Executing Dial("SIP/101-f9eb", "ZAP/g1/943833473|45|tTwW") in new

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Hi all, we're buildin an Asterisk box based on an Intel IXP425 board. The board uses a Beronet BN2S0 ISDN card, mISDN 1.1.4 and asterisk 1.4.2. hfc_multi has been patched to compile under big endian cpu, and so also capi kernel files. All the modules seem to load correctly (configuration was made with misdn-init config), but when starting cha_misdn, asterisk outputs the following lines: P[ 1]

hello, I have instaled a analog line, and when I call on the console apears that: I want to redirect the call to 101 extension. *CLI> -- Starting simple switch on 'Zap/1-1' == Starting Zap/1-1 at default,s,1 failed so falling back to exten 's' == Starting Zap/1-1 at default,s,1 still failed so falling back to context 'default' Apr 27 08:15:53 WARNING[3494]:

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

Is there any less cumbersome way of doing conditionalized/branching in extensions.conf other than something like: exten => s,n,GotoIf($["${SIP}" = "PJSIP" ]?pjsip) exten => s,n,Dial(${ARG2},20,TtWw) exten => s,n,Goto(afterdial) exten => s,n(pjsip),Dial(${PJSIP_DIAL_CONTACTS(${STRREPLACE(ARG2,"PJSIP/","")})},20,TtWw) exten =>

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>>

On Wed, 2019-04-17 at 10:04 -0400, Joshua C. Colp wrote: > > You specify the transport in the SIP URI. For example: > > sip:test at example.com;transport=tcp Hrm. This is probably going to be pretty basic, but some googling didn't seem to come up with anything. How do you do this when you are assigning to a variable in extensions.conf such as: FRED=PJSIP/outgoing/sip:fred at

hi i've configured a Beronet BN8S0 Card with misdn beronet utility. the card is configured with all lines in TE and P2P mode, and it is connected with the special cable with an ISDN connection. i've turned on debugging to level 4, this is the output from the asterisk cli when i receive a call: P[ 5] MGMT: Short status dinfo 1000001 P[ 5] MGMT: SSTATUS: L1_ACTIVATED P[ 5] handle_frm:

On Wed, 2019-04-17 at 11:56 -0400, Joshua C. Colp wrote: > On Wed, Apr 17, 2019, at 12:51 PM, Brian J. Murrell wrote: > > > > I can add it onto the end of the variable in the Dial() command: > > > > Dial(${FRED};transport=tcp,${timeout},TtWw); [ the part you trimmed ]: and it has the desired effect, but of course specifying at the point where the Dial command is sets

Hi, at my Asterisk box, I have a few of IAX2 phones (configured with alaw/ulaw/gsm codecs, in this order) and a PRI E1 line. In iax.conf I hav: disallow=all allow=alaw allow=ulaw allow=gsm During some incoming call, I read at console: -- Executing Dial("Zap/2-1", "IAX2/215|20|TtwW") in new stack -- Called 215 -- Call accepted by 10.97.1.7 (format ulaw) --

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'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

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 ....

Anyone know why chanspy was not included in asterisk distribution as of October. ? I tried patching my current 1.0 but seems the patches are for an older version. I posted a bounty of $250 to get this to work with the newest stable. Needs be able to monitor bridged sip calls with or without a monitoring beep. Thanks John Bittner Simlab.net

Hi, I have FRITZ!Card PCI card. I have installed misdn-1.1.0 on stable debian 3.1 with 2.6.8. kernel. Then I reboot system, and it doesn't boot, it stops near "Apache2 starting...". I started my system with "recovery" kernel, and tun off misd, then my system works fine. I think it's problem with memory. Has anybody debian and misdn working fine? Maybe you can

I had Asterisk 1.6.2.2 running fine with a mISDN using a HFC-S based card. I installed my TDM400P into the PC, it's really slow to boot now, when it finally does I gets stuck in a loop of reporting "isac xdu no tx_busy". Anyone able to assist? Thanks in advance! -------------- next part -------------- An HTML attachment was scrubbed... URL:

i am using asterisk 1.6.2 with mISDN and a passive AVM Fritz!card on CentOS 5.5. The only thing, i want to do is a call-redirection from an isdn-call to my mobile via sip-account. My extension conf is: general] static=yes writeprotect=no [globals] OUT_PORT=1 [ISDN] exten => 12345,1,Dial(SIP/012346737222 at sipprovider.local) If i call to the msn 12345, the SIP-call is going out, but after

Hi. Using Asterisk 1.4.13 running on Ubuntu 7.04 with Intel CPU: 1) Not being able to build mISDN on Ubuntu using "make b410p" I have used mISDN-1_1_7 which seems to work ok. QUESTION: Should I expect this version of mISDN to work ok with these cards? Or is there a way to build using "make b410P" on Ubuntu? (make force does not help at all) 2) In some of our installations