similar to: Dial and congestion

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 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]

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I'm having some problems using the asterisk modem driver with an ISDN4Linux card (AVM Fritz). (I realise that it's better to use CAPI, but unfortunately this card doesn't have any CAPI drivers). Every so often the ISDN just stops working (it neither dials out nor accepts incoming calls). Trying to dial out just logs: --

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? --

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

-----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 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'

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

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

I have made bri-stuff.0.0.2rc19 to work (I think) but I can not achieve any in-dialing nor I can dial out; this is what I have from "pri intense debug span 1" command ---------- *CLI> pri intense debug span 1 Enabled EXTENSIVE debugging on span 1 -- Executing Playback("SIP/201-a862", "tt-weasels") in new stack -- Playing 'tt-weasels' (language