similar to: Need help: X100P connection/configuration in GERMANY

hi anybody running with german SIPGATE? my configuration don't works :-( regards thorsten@gehrig.de

hi, i´am new in tcc (tcng). i try to define my qos for VoIP-Services. For this i wantto define a class for a port range 10000 till 15000. how is the right way? this down works: class (<$voip>) if tcp_sport => 10000 || tcp_sport <= 10000 ; are there any examples of real installations - maybe including VoIP, HTTP and P2P services? regards thorsten gehrig

Deutsche ?bersetzung folgt / German version following ===================================================== Hello, i have Asterisk running with 2 ISDN-Cards. One AVM Fritz for connection to german ISDN and one HFC-compatible-Card (NT mode) for connection to ISDN-Phone (later: ISDN-PBX). Here is my actual installation: ISDN -> Fritz - ASTERISK ? HFC-NT <- ISDN-Telephone If i pick up my

Hi, we have a speech recognizer here that gets some of its input from a microphone array. To the point we used shorten for the archiving of the material, but since we want to change the license of the speech recognizer to gpl we are searching for an gpl alternative (like FLAC). It seems to be pretty good for our needs (24bit rather than 16bit in shorten and so forth). The only problem is that our

Hello. I am a spanish student, so excuse my English. I have this HW: - 2 X100P PCI with two analog lines plugged in. These lines are two extensions of a panasonic PBX. Zap/1 = X100P <-- analog line --> extension #237 PBX Panasonic Zap/2 = X100P <-- analog line --> extension #245 PBX Panasonic - 1 TDM20B with two analog telephones plugged in. Zap/3 = TDM20B

I tried adding wwww to my dial string... It appears to not made any differnece. I see by watching show channels that the w's are there in the dial. I have this box connected to bell south down in georgia. I have an identical box connected in indiana. The box in indiana works fine when dialing out. When in GA dialout is getting "you must first dial a 1 to place this call". show

Hello! We are using the Digium 405PP card, and getting the following messages: Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1 Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 My config file is below. We are trying to set up D-Channel on channel 24, 1-23 in trunk group 1,

Hi All, I have just migrated from Asterisk 1.0.0 to Asterisk 1.0.5 and I have an X100P installed. The old asterisk was working, but now the new version isn't picking up any calls! However, I did notice that after installation, I performed modprobe zaptel and modprobe wcfxo and they worked fine, but when I executed ztcfg, I get the following errors: ioctl(ZT_LOADZONE) failed: Invalid

FishSound 0.9.0 Release ----------------------- libfishsound provides a simple programming interface for decoding and encoding audio data using Xiph.Org codecs (FLAC, Speex and Vorbis). This release is available as a source tarball at: http://www.annodex.net/software/libfishsound/download/libfishsound-0.9.0.tar.gz New in this release ------------------- This release introduces support for

FishSound 0.9.0 Release ----------------------- libfishsound provides a simple programming interface for decoding and encoding audio data using Xiph.Org codecs (FLAC, Speex and Vorbis). This release is available as a source tarball at: http://www.annodex.net/software/libfishsound/download/libfishsound-0.9.0.tar.gz New in this release ------------------- This release introduces support for

Oggz 0.8.6 Release ------------------ liboggz is a C library providing a simple programming interface for reading and writing Ogg files and streams. Ogg is an interleaving data container developed by Monty at Xiph.Org, originally to support the Ogg Vorbis audio format. This release is available as a source tarball at: http://www.annodex.net/software/liboggz/download/liboggz-0.8.6.tar.gz New in

Oggz 0.8.6 Release ------------------ liboggz is a C library providing a simple programming interface for reading and writing Ogg files and streams. Ogg is an interleaving data container developed by Monty at Xiph.Org, originally to support the Ogg Vorbis audio format. This release is available as a source tarball at: http://www.annodex.net/software/liboggz/download/liboggz-0.8.6.tar.gz New in

I have a bit of a weird problem that I'm having great trouble debugging. I have a TDM400P PCI card with two FXO and two FXS modules. Both FXO modules are connected to BT lines here in the UK. Both BT lines have V23 Caller-ID, which works fine with Asterisk. Both asterisk and zaptel are fresh from CVS. Both FXO modules (channels 3 and 4) are in "group 1" for outgoing calls. My

Hi list First of all could you please explain loop-start and ground-start for me? What are they used for? Next, I have the following configurations: dahdi-channels.conf : context=pstn-channels signalling=fxs_ks channel=>130 context=phone-channels signalling=fxo_ks channel=>127 chan_dahdi.conf : [channels] cidsignalling=dtmf cidstart=dtmf signalling=fxo_ls pulsedial=no

Steve, Here is the config, I pulled from my server, that works with D'Link Phones: Main Menu -------------------------------------------------------------------------------- SIP.CONF [general] port = 5060 ; Port to bind to (SIP is 5060) ;bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) bindaddr = 67.109.153.236 disallow=all ;allow=ilbc allow=gsm allow=ulaw

I have turned off call progress in zapata.conf and things seem to be working better. Jerry ---------- >/ I tried adding wwww to my dial string... It appears to not made any />/ differnece. />/ I see by watching show channels that the w's are there in the dial. />/ />/ I have this box connected to bell south down in georgia. />/ I have an identical box connected in

Hi, I've seen this USB product from Sangoma : http://www.sangoma.com/products_and_solutions/hardware/analog_telephony/usb_fxo.html Is it working ok ? Is it easy to integrate it with Asterisk ? How would you rate your experience with it ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Help. Why is it that I can call out from my GSBudgetone SIP phone but the audio is "one-way'? Why is it that when I call my asterisk phone number, I get a fast busy?

Sorry, that name is a misnomer. However, I was excited to find that Ruby has a built in Observable module and I''m pretty bored, so I apologize in advance.... require ''observer'' # one who is observed class Celebrity include Observable attr_accessor :name attr_reader :is def is=(val) @is = val changed notify_observers(self) end end # one who

Hi all, The CentOS team is preparing for LinuxTag 2007 on: Wed 30 May 2007 until Sat 2 June 2007 in Berlin, Germany We have a wiki page dedicated to LinuxTag 2007 at: http://wiki.centos.org/Events/LinuxTag2007 If you're interested to help out organising this event, subscribe to our new promo mailinglist at: http://lists.centos.org/mailman/listinfo/centos-promo And if you were