similar to: Intermittent breakage with the ISDN4Linux modem driver

I can't seem to see any documentation on how to access Asterisk functions (rather than applications) using AGI? -- - Steve xmpp:steve@nexusuk.org sip:steve@nexusuk.org http://www.nexusuk.org/ Servatis a periculum, servatis a maleficum - Whisper, Evanescence

I understand that the X100P card is matched to a 600 ohm impedance but the UK BT phone system is not (I haven't been able to find much information on the impedance of the UK system). Has anyone come up with an easy way to match the impedance between the two so the X100P can work in the UK? Presumably a simple transformer won't do the job since it won't pass the DC components? --

I am wondering if anyone has a working install of oh323 on fedora Core2. An replies would be appreciated as we need this urgently. Seshu Kanuri -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of steve@nexusuk.org Sent: Thursday, July 22, 2004 6:12 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users]

My Asterisk server is connecting to "sip.plus.net", which resolves to multiple IP addresses: sip.plus.net. 300 IN A 84.92.0.75 sip.plus.net. 300 IN A 84.92.0.76 sip.plus.net. 300 IN A 84.92.5.189 sip.plus.net. 300 IN A 84.92.5.190 If one of these machines is down

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I'm experiencing frequent kernel panics when using the X100P card under the 2.6.6 Fedora kernel. I've attached the kernel output to this message - - it looks like the IRQ stack is overflowing and trashing some memory, causing a series of oopses followed by a complete crash. I have just hacked the kernel to reenable 8k stacks and will see

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I've got a system set up like: POTS <-----> Asterisk 1 <-----> Asterisk 2 <-----> IAXComm The POTS line is connected to "asterisk 1" via an X100P card and "asterisk 1" is connected to "asterisk 2" via ethernet. With incoming calls from the POTS line, everything works, but between 5 and 10

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I was wondering if anyone has found a IAX or SIP client for the Sony Ericsson P900 phone - there seems to be some stuff available for Symbian 60 phones but I haven't been able to find anything for Symbian UIQ phones like the P900 and P800. (The idea is to have the ability to use the phone as a cordless phone doing VoIP over the bluetooth PPP

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Can the Dial() command tell the difference between busy and congestion? At the moment it seems to be treating them both the same on my server. I want to route the calls out via a SIP gateway unless that is congested, in which case dial out through my POTS line (using an X100P). It seems a bit pointless to try dialling the POTS line when the SIP

While re-setting up my Asterisk system I decided that implementing call routing through ENUM lookups would be a good idea. However, I've hit a bit of a problem. Given DNS records like: *.4.4 IN NAPTR blah 9.8.7.6.5.4.3.2.1.4.4 IN NAPTR blah The first record is there to route all numbers starting +44 via a particular gateway. The second record overrides that routing for a specific

Hi Folks, I am breaking my head for compiling asterisk-oh323 properly on my asterisk box from past 1 week. But still after my all efforts, I unable to make it compile properly, My box is Fedora core 2 with asterisk-0.9.0. I was trying for following configuration with openh323 and pwlib. Openh323 and pwlib are installed properly. But problem is asterisk-oh323. asterisk-oh323-0.6.2a.tar.gz

I have no analog line interfaces on my asterisk system, but I do have two UK 0870 numbers routed to two separate VoIP accounts (one with FWD, one with gossiptel). Asterisk is configured to register with these accounts. I get voice calls through just fine this way. I thought I could get one of these 0870 numbers to route through to rxfax, thus allowing folks to fax me directly. I've set up

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Not strictly Asterisk related, but I'm trying to get an AVM Fritz! card to work with Asterisk on a Fedora Core 3 box. I've grabbed the official AVM Fritz CAPI driver from the AVM website and compiled against the current kernel. However, as soon as I modprobe it I get: CAPI Subsystem Rev 1.1.2.8 fcpci: module license 'Proprietary'

Greetings, today I'm recompiling asterisk on a new machine with the same isdn card (not the same model, the *SAME* card) using CVS downloaded yesterday night. When I try to start *, it gives me the error: [chan_modem.so] => (Generic Voice Modem Driver) == Parsing '/etc/asterisk/modem.conf': Found == Loading modem driver chan_modem_i4l.so => (ISDN4Linux Emulated Modem

Hello! Im pretty new of asterisk world: my goal would be using iaxcomm to call ppl over POTN. Yesterday i configured asterisk to be able to hear a song (with Playback() command) with iaxcomm, and it was wonderful. Now, today i tried to configure asterisk to use my ISDN4Linux supported card (a plain HFC card) with asterisk. i configured modem.conf this way, after reading

Hi *, This is my first message to this list, so I hope I am not breaking any rule with my post. My question is the following: can I establish a voice connection over an ISDN interface by using a bluetooth dongle to connect to the ISDN access point? Here follow the details: I have compiled and configured asterisk. Everything seems fine except that when I configure asterisk to use the ISDN

Hi, Having seen the messages on the dev list a couple of weeks ago about chan_sebi I thought I would try to get it going on my system. I am using 1.6.1.4 so I first upgraded the driver to work with 1.6. I think it is mostly correct and the bit I have wrong shouldn't be causing me the problems I am seeing. If your are interested my chan_sebi 1.4 to 1.6 patch is at

hi i would like samples examples to configure with isdn4linux i have hisax card : "gazel" and an ISDN(BRI) line (2 channels B and 1D) In fist time i'll use "sjphone" only Perhaps there is french people on this list who can help me to do first steps with "Asterisk" thanks

Hello All, I was just after a few pointers. I have just setup my Redhat 9 linux box with asterisk. Internal SIP call working fine. My Eicon ISDN card turned up today so - plugged it in and went through the modem.conf. It reports unable to open /dev/ttyI0 The problem is I have never used ISDN with Linux - let alone a telephony app - and I have no idea even where to start. Some pointers

Hello all, does anybody have experiences with DDI on ISDN BRI? I would like to order DDI with 10 or 100 numbers from telco company, and use DDI numbers for direct calling SIP/H323 clients. But I don't know if Asterisk supports it, in archive I found only one answer about it, and I'm not sure. I am going to use it with isdn4linux card. Maybe DDI is called differently in other countries,

hi I have a passive isdn port configured in modem.conf in extention.conf i use this two channel (ttyI0 and ttyI1) with the string: exten => _NXXXXXXXXX,1,Dial,Modem/g1:${EXTEN}|60|r how can i hide my msn? is it possible to activate the clir with the @ before the ${EXTEN}? thanks Cristian