similar to: meetme problem maybe connected to features.conf

Hi list, I have a server running a simple hfs isdn card running with chan_misdn. the problem is, I can't get asterisk to pick up the phone, outgoing calls work fine. when running asterisk with -vvvvvvvvvvvc I get the following "log" when I try to dial the isdn server. P[ 1] * Starting Ast ctx:kpn-in dad:0594643637 oad:0878722291 with 's' extension P[ 1] MGMT: SSTATUS:

I am trying to use misdn insted of zaphfc to drive two billion isdn cards zaphfc is ok, but the problem with cdr and the fact tha you always have to wait the bristuffed version of asterisk took me to try another way. so I downloaded the misdn installation script from beronet for the last version ( I am using asterisk stable 1.2, so now is 1.2.5) wget

I'm using avm fritz card usb with kernel 2.6 patch with mISDN. The module is load correctly when I type lsmod I've got the following output: Module Size Used by zaptel 178308 0 avmfritz 21388 0 mISDN_isac 14336 1 avmfritz mISDN_dsp 191424 0 l3udss1 34184 0 mISDN_l2 39040 0

Hi, I have switched a while back from chan_capi to chan_misdn. When the number is dialed and the phone is then picked up everything works just fine. Some users however FIRST pick up the phone and then start to dial... I did not get this to work with misdn. When two digits have been dialed, asterisk sees the extension as complete and does not wait for further digits. I am using an midsn NT

Hello ! I would like to get working a Fritz PCI card using chan_misdn operating in ptp mode. Afer compiling mISDN into the kernel and building chan_misdn Asterisk stops loading with : [chan_misdn.so] => (Channel driver for mISDN Support (Bri/Pri)) == Parsing '/etc/asterisk/misdn.conf': Found UnLocking config_mutex == Registered channel type 'mISDN' (This driver enables

Hello. I have problems to integrate an isdn phone. I don't know why but the isdn phone rings only once and than it looses its connection to his base station. I can make a call from the isdn phone to an VoIP Phone inside my network but when i pick up the phone the isdn phone also crashes. misdn.conf: [ntport1] ports=5 context=isdn-telefon msns=* extensions.conf: exten =>

Hi. I'm having problems with chan_mISDN configuration. Line "echotraining=yes" causes warning, when Asterisk is parsing misdn.conf and I'm confused why the PBX doesn't accept the setting. No matter which section I try to offer it, it is always invalid or out of range. The setting itself is supposed to be valid, it is in the sample configuration file of chan_mISDN 0.3.1.

Hello all. Dialing with tT options and function Read (to prompt number) has a trouble for me. Can I temporarily features off during Read? features.conf: [featuremap] blindxfer => ## ; Blind transfer (default is #) atxfer => ** ; Attended transfer I try: exten => s,n,Set(LOCAL(tmp_atxfer)=${FEATUREMAP(atxfer)}) exten =>

Hi Is there a way to disable blind and attended transfer during a call. I am trying this configuration but unfortunately with no luck: - in features.conf [applicationmap] disabletransfer => 9*9,self,GoSub(disabletransfer,s,1) - in extensions.conf [incoming] exten => 99,1,Set(__DYNAMIC_FEATURES=disabletransfer) exten => 99,n,Dial(Sip/alice,120,tT) exten => 99,n,Hangup()

Hi folks, I was using the following featuremap: blindxfer => *1 disconnect => *9 atxfer => *2 parkcall => *7 automixmon => *0 and everything worked. Then the need arouse to use some features like automixmon during a conference, but MeetMet has the * key bound to the (admin) menu. Thus, in order to enable features like automon and transfers even during a conference, I

I have 10 msns on my isdn bri incoming (AVM Fritz, chan_capi 3.5) When i set in capi.conf [interfaces] msn=1,2,3,4,5,6,7,8,9,10 incomingmsn=* controller=1 softdtmf=1 accountcode= context=tcom-in echocancel=yes devices=2 Only msn 1,2,3 are working. What can i do? Thanks Jens

Hello, I am using a system with an AVM ISDN PCI card (fcpci) and asterisk with chan_capi-cm-0.6. The hardware is connected to a Siemens Hipath 3550 PBX. As a BRI connection has 2 channels and allows 2 simultaneous calls, numbers/MSNs 6391 and 6392 were for provisioned for each channel. The system is working (partly, read on), the trick is the correct cable wiring and setup the PBX's port

Hi, how can I set up multiple ISDN controllers with chan_capi, so that every controller has its own configuration (MSNs to listen etc...) ? I know I can configure "controllers=1,2" in chan_capi's capi.conf but then the controllers have the identical configuration and context within asterisk. My setup: I have 2 BRI-controllers (in one AVM C2). 1. BRI: point-to-point, Number:

Hello, i wonder why i didn't find a solution for this problem yet, because it seems very common: I have an asterisk server with an AVM (Fritz) ISDN-Card (BRI), and some SIP-Softphones which i can call from outside by calling the phonenumber of the Asterisk-Server and then dialing the number of the SIP-Phone. If I make a call from a SIP-Phone into PSTN, only the MSN of the asterisk-server is

Hi all, I've been playing about with Asterisk for years now on and off, just SIP to SIP calls, using FWD and suchlike. I'm moving house at the beginning of September and have decided to build an Asterisk based system for my home office. I'm in the UK and wonder if anyone can give me advice on lines and hardware to use. Had planned to go with an ISDN2e line coupled with a BT

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'm wondering whether there's a problem with the blindxfer and atxfer commands. I was using Asterisk STABLE and pressing the # key to transfer calls worked fine, except of course when you called up FedEx and they asked "Enter the number of packages, followed by the Pound key". I found on the wiki (http://www.voip-info.org/tiki-index.php?page=Asterisk+config+features.conf) that

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi there, i'm in debian sid 3.1 with kernel 2.6.7, * last cvs & chan_capi 0.3.4b; nt1+ with 2 bri in ptmp (http://www.voip-info.org/tiki-index.php?page=DDI) i tried to install avm c4 following step by step http://www.voip-info.org/tiki-index.php?page=Asterisk%20How%20to%20connect%20with%20CAPI step 1. i compiled capi 2.0 support in kernel

Hi folks! Is it possible to tell asterisk not to strip the leading 0 of *incoming* MSNs? I use asterisk with i4l and whenever I get a call from an long-distance party, the leading 0, which should be there according the german numbering, is not. So if I get a call from a mobile phone 0177-1234567 should be displayed, but 177-1234567 is displayed. I double checked if I've forgotten to remove an

i have TB instaled and i cant get call pickup when another phone rings i tried ** , *8 , *8# , **+ext but nothing seems to be ok.on extention menu i put call pickup=1 and call group=1 but nothing look at my features.conf; ; Sample Parking configuration ; [general] ; do not manually enter parkinglot config information, use the parkinglot module ; ; the parking_additional.inc file is