search for: outgoingmsn

...open one is free) the forwarding is processed (snom 200) all seems correctly. Then the message that the b channels all busy, but so is it not. Forwarding to a sip phone works. Can anyone help me with that ? nicolas SNIPS: ------ == DISCONNECT_IND PLCI=0x201 REASON=0x34a2 -- CAPI[contr1/<outgoingmsn>]/78 is busy -- CAPI Hangingup -- removed pipe for PLCI = 0x201 == Everyone is busy at this time -- Executing Congestion("Local/<nummer to forward>@default-f1df,2", "") in new stack -- Local/<nummer to forward>@default-f1df,1 is circuit-busy ==...

...want use both B channels on my isdn card (B1 ISA) but chan_capi open one channel and asterisk say 2. channel is busy. Must i use another isdn card ? I have a old B1 ISA card. Can anyone help me with that ? nicolas SNIPS: ------ ??==?DISCONNECT_IND?PLCI=0x201?REASON=0x34a2 ????--?CAPI[contr1/<outgoingmsn>]/78?is?busy ????--?CAPI?Hangingup ????--?removed?pipe?for?PLCI?=?0x201 ??==?Everyone?is?busy?at?this?time ????--?Executing?Congestion("Local/<nummer?to?forward>@default-f1df,2",?"")?in?new?stack ????--?Local/<nummer?to?forward>@default-f1df,1?is?circuit-busy ??==...

...sterisk 1.0.7.dfsg.1-2 - chan_capi 0.3.5-11 - Debian Sarge on Linux 2.6.15.1 - FritzCard module built from the latest sources available by AVM My capi.conf looks like this: [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] msn=3413 incomingmsn=3413 outgoingmsn=3413 controller=1 softdtmf=1 accountcode= context=isdn_incoming If you need any more information on my * configuration, please feel free to ask. Thanks in advance!! Regards, Christian Schmidt

...30 ERROR[384]: chan_capi.c:2051 capi_handle_msg: did not find > device for msn = 523065 > -- INFO_IND ID=001 #0x0021 LEN=0022 > Controller/PLCI/NCCI = 0x101 > InfoNumber = 0x70 > InfoElement = <c1>523065 I think the outgoingmsn is missing / the one you are looking for > > > Here are my entries of /etc/asterisk/capi.conf: > ================================= > ; CAPI config > ; > [general] > nationalprefix=0 > internationalprefix=00 > rxgain=0.8 > txgain=0.8 > > [interfaces] > &...

...plan to the outgoing MSN will be 260 than it works as expected: 253 will be called, but the called party gets 260 as callerid instead of the 265. My capi.conf looks like this: [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 language=de [interfaces] msn=260 incomingmsn=* outgoingmsn=260,261,263,264,265 controller=1 softdtmf=1 context=fromcapi language=de echocancel=yes devices=2 Okay, I could also set msn=265; but in the future, more msns should be redirected to VoIP channels, so this is not a long term solution. Any help is very welcome! Thanks for asterisk! :wq! PoC

...l.so => (ISDN4Linux Emulated Modem Driver) But if I define a test extension such as: TRUNK=Modem/g1 exten => 2468,1,Dial(${TRUNK}/91234567:0412345678) and try to dial it, the console says: Dec 14 13:29:17 WARNING[15375]: chan_modem_i4l.c:608 i4l_dial: Outgoing MSN andrew not allowed (see outgoingmsn=,, in modem.conf) -- Called g1/91234567:0412345678 -- Modem[i4l]/ttyI0 is busy -- Hungup 'Modem[i4l]/ttyI0' I gather than "busy" is used for pretty much everything except for no connection, but are there any suggestions of where to look? Thanks in advance, Andrew -...

Hello List, because this is my first post to this list, i'd like to introduce myself. My name is Thorsten M?ller, 26 years old and live near Frankfurt/Main in germany. Okay, now to the reason for this posting: I just installed my first asterisk server (Debian 2.2 Kernel 2.4.18-686) with AVM ISDN Fritz PCI card (passive). I followed the configuration on http://voip-info.org and the

...ured at 0x2000 IRQ 201 NETjet-S: ISAC version (0): 2086/2186 V1.1 NETjet-S: IRQ 201 count 0 NETjet-S: IRQ 201 count 6 HiSax: DSS1 Rev. 2.32.2.3 HiSax: 2 channels added HiSax: MAX_WAITING_CALLS added Asterisk's modem.conf driver=i4l group=1 msn=(my first number) incomingmsn=(my other numbers) outgoingmsn=0 device => /dev/ttyI0 device => /dev/ttyI1 ;features=dtmf,noquelch ;features=dtmf Asterisk -cvvvvv shows: Asterisk Ready. *CLI> == Starting Modem[i4l]/ttyI0 at IN-isdn,(my msn),1 failed so falling back to exten 's' == Starting Modem[i4l]/ttyI0 at IN-isdn,s,1 still failed so...

Dear list, today I installed a new asterisk machine, bound to replace my current pbx. I am using a Fritz ISDN card; on the old machine with the drivers coming together with the super-old rpm asterisk installation of SUSE 9.2. The new machine is finally on asterisk 1.0.9, with chan_capi 0.5.4; now I am doing a nightly test. Apparently I can receive calls, but I can't dial out. I seem to

...-- 2-2 test OK SIP Phone (on Asterisk) call 5221 ---------------------------------------------------------- -- Executing Dial("SIP/4122-d20d", "Modem/g1:5221") in new stack Aug 3 02:54:22 WARNING[1141783472]: chan_modem_i4l.c:608 i4l_dial: Outgoing MSN 4122 not allowed (see outgoingmsn=,, in modem.conf) -- Called g1:5221 -- Modem[i4l]/ttyI1 answered SIP/4122-d20d (blah blah.. and hang) -- Hungup 'Modem[i4l]/ttyI1' == Spawn extension (lesmuids, 5221, 1) exited non-zero on 'SIP/4122-d20d' Strange think I can call a SIP extension ----------------------...

Hi I have installed the http://asteriskathome.sourceforge.net/ with a Digium card with no problems, very good ! Now I want to install my Billion PCI ISDN card (HFC based) in TE mode. I get a little confused with Isdn4Linux, ZapHFC HIAX and the need to install Capi ! Please suggest best and easiest approach ? Thank you ! HB Norway

I'm trying to use an hfc based pci card with asterisk but every call fails falling in the congestion extension. exten => _0.,1,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}}||tr) exten => _0.,2,Congestion Looking in the syslog i can see: isdn: HiSax,ch0 cause: E001B it seems that this is a terrible error when arrives... hard to tell what is the cause. Also terrible is finding a lot of material

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Hi, I am having trouble configuring CAPI so that call transfers work. I make a SIP call to asterisk which goes out on ISDn via CAPI. Then I try to do a transfer from the SIP phone which doesn't work and results in the call being disconnected. The error message given by asterisk is that it chan_capi can't find an entry for the outgoing msn for the transfer however the outgoing msn is the