similar to: Beronet card - issue?

Hello. Just installed the Beronet BN4S0 card. But i can't connect to my ISDN Line. misdnportinfo gives (what does ":Layer 4 protocol 0x04000001 is detected, but not allowed for TE lib" mean?): best regards and thanks t. asterix asterisk # misdnportinfo Port 1: TE-mode BRI S/T interface line (for phone lines) -> Protocol: DSS1 (Euro ISDN) -> Layer 4 protocol 0x04000001

Hi All, We've got msidn configured: Port 1: TE-mode BRI S/T interface line (for phone lines) -> Protocol: DSS1 (Euro ISDN) -> childcnt: 2 -------- mISDN_close: fid(3) isize(131072) inbuf(0x8fd5060) irp(0x8fd5060) iend(0x8fd5060) and running on Asterisk 1.4.21.2: pbx*CLI> misdn show stacks BEGIN STACK_LIST: * Port 1 Type TE Prot. PMP L2Link UP L1Link:UP Blocked:0 Debug:0

Hi, I am struggling to get plain Cologne chip cards to run in NT mode, runs nice in TE mode despite the error message: login as: root root at 192.168.2.22's password: Last login: Tue Sep 1 23:09:24 2009 from 192.168.2.50 Welcome to Elastix ---------------------------------------------------- misdnportinfo Port 1: TE-mode BRI S/T interface line (for phone lines) -> Protocol: DSS1

Hi, I have a Digium B410P 4-port BRI card. I installed misdn 1.1.3 with hfcmulti driver and misdnuser 1.1.3. I configured the card "correctly" as misdnportinfo reports: # misdnportinfo Port 1: TE-mode BRI S/T interface line (for phone lines) -> Interface is Poin-To-Point. -> Protocol: DSS1 (Euro ISDN) -> childcnt: 2 -------- Port 2: TE-mode BRI S/T interface line (for

Hi list, I've installed Asterisk 1.4.0 with newest mISDN 1.0.4 + mISDNuser 1.0.3 on Fedora Core 6. I get many compilation error on mISDN. It wants to include linux/config.h That I fixed by removing the #include line at every occurance. (Don't know if that was a wise move, but it then compiled). mISDNuser and asterisk compiled fine, and asterisk can find and use the ISDN BRI port in

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

Hi. When I call my RNIS numbers (with a mobile phone for example), I can see 2 incoming calls on the IPBX, which should not happend. I'm not sure if it's a problem with the telco France Telecom and their ISDN setup, or if it's a problem with the MISDN driver on the IPBX itself. I'm stuck ... Any advices for troubleshooting that? Someone provide working configuration files

Hi, Since a week ago I am trying to get chan_misdn working with asterisk 1.4.19, using HFC based ISDN card on Linux 2.6.22. My setup is done as detailed on wiki and FAQ. * mISDN and miSDNusers are 1.1.7.2, unpacked, build and installed. After installation and misdn-init, I have this: aragorn:root/pts/1: # misdnportinfo Port 1: TE-mode BRI S/T interface line (for phone lines) ->

Hi all, I'm fighting with a really strange problem that is really busting me. I have an asterisk 1.4.22 ( from a trixbox 2.6.2 ) and mISDN 1.1.7 3 extension on hardphone and 3 extension in softphone ( zoiper ) What happens is that sometimes the people on the other side of communication hear my voice as metallic and chopped. This happen either on incoming call than on outgoing call. If I

Hi. I'm running Asterisk server with 10 sip phones, and 2 grouped T0 lines with 10 DDI numbers. My provider is France Telecom and my setup is : - Debian Lenny - Asterisk 1.4 - Linux kernel 2.6.25.17 - mISDN 1.1.8 driver - Sip phones Thomson ST2030 No problem with the SIP . But when reveiving a call on RNIS line (any of the DDI numbers), the associated SIP phone rings indicating _two_

hi all, how can I set the interval in second from retrasmit the magic packets when qualify is set to on? I want to view whitch voip-phone is connected but I don't want to DOS my adsl connection.... ;) Thanks Enrico P. -- Pasqualotto 'Pasqu' Enrico enrico AT pasqualotto DOT org web: http://www.pasqualotto.org skype: epasqualotto

Hi all, I have to make for a client an asterisk system for process up to 250 calls between conference and normal call. At disposition I have 4 xserver 346 with dual xeon 3.0Ghz and the client require a failover system. Anyone have experience for this type of solution? Is better ultramonkey, dundi or SER proxy in front of * server? Thanks Enrico P.S. Now during all this year I have to work

Hi I use Asterisk with some SIP phone (grandstrea), while with my notebook when I'm out of home/office I use X-lite and all work. Some days ago I try to install wengophone and I decided that I want replace X-lite for use wengophone as client for my Asterisk. One of the reasons is that wengophone support g729 codec. I make some test and I see that is possible to configure other sip server

Hi all, I want to write a simit sip client in PHP with asterisk API, in this moment I'm able to compose a number on my browser and call between 2 hw sip phone. I digit a number, my phone ring and after hanging up the cornet the second phone ring. But I want to add a features.... I want to hang up the cornet of my phone, compose the number in my browser and call a second phone. In witch

hi all, I'm trying to set up some iax2 trunks in Realtime architecture with the same backend. All work better (make call, receive etc etc) but when I do "iax2 show peers" some asterisk don't show anything and other show the iax2 peers but with status "unknow". Name/Username Host Mask Port Status ctm1/trixbox 10.0.0.131 (S)

Hi all, I use the G option in my dials for redirect both parties in the conference. There is a way for auto-include in a conference other parties that first two without using AGI? I try with: [from-internal] exten => 9999,1,Dial(IAX2/DIP02/9999||G(fromiax^9999^1) [fromiax] exten => 9999,1,MeetMe(9999,qdxAa) exten => 9999,2,MeetMe(9999,qdx) exten =>

Hi all, I have 2 asterisk box connected with IAX trunk. One box have connected a SIP phone and the second have a TDM card with one analog phone. When from SIP phone I try to park the call from analog phone with #700 the call is correctly parked but in the second asterisk I see this log: -- Executing Dial("Zap/2-1", "IAX2/CTM1/STI1|30|rjtT") -- Called CTM1/STI1 --

Hi all, I have a network with some asterisk in trunk with IAX2 and some SIP/ZAP phone connect to this *. In every call I need to use only alaw codec so in all conf file I have set disallow=all and allow=alaw. I try also to make some tuning of my environment removing unused codec and application. If I remove the codec_ulaw.so when I try to call I see this: [Oct 5 12:15:33] WARNING[16637]:

Hi all, I use asterisk with realtime features for extension, sip and iax. In extensions.conf I have put these lines: [from-internal] include => parkedcalls switch => Realtime/@ [fromiax] switch => Realtime/@ There is a way for put in my database the context also? Now if I want to add a new context I have to modify the extensions.conf with: [newcontext] switch => Realtime/@ but I

Hi all, I have a question on asterisk conference. Now I use appl Meetme with option A & x because when a marked person hangup I want to close all the conference. But what I have to do if I want two marked person and kill the conference when one of two hangup? Is possible? Thanks. Enrico. -- Pasqualotto 'Pasqu' Enrico enrico AT pasqualotto DOT org web: http://www.pasqualotto.org