similar to: error while compiling asterisk-oh323

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

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? --

-----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: --

-----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

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 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

Folks! This is to let all of you know that I am making D'Link make an all out effort to make D'Link Phone DPH80 and DPH100 work with Asterisk. I have provided the Asterisk Platform to D'Link's R&D Division located in Goa, India, where their IP phone's SIP Bios is undergoing modifications based on my recommendations/suggestions. I have also provided the test bed &

Hi Folks, I am newbie to asterisk. Recentely I have installed asterisk on Linux Fedora 2 box. After reading some document, I tried to configure the server. When I connect to our server, SIP user-agent shows that I am logged in. But it doesn't show my system(client) IP when I issue command at astrisk CLI. The O/P is as below. *CLI> sip show peers Name/username Host

-----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

Hi there. I thy to compile asterisk-oh323-0.6.3a but it fail in the make command. I have the pwlib-v1_6_6-1 and openh323-v1_13_5-1 as saying in the README file of the packet asterisk-oh323-0.6.3a I do make and this is the error: # make for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done make: *** No rule to make target `ccflags'. Stop. make: *** No rule to make target

-----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

Hello I?ve just install last cvs version (Mon Sep 6) of Asterisk with asterisk-oh323-0.6.3b and pwlib-v1_6_6-src.tar.gz, openh323-v1_13_5-src.tar.gz and . this is the error loading asterisk with chan_oh323 module:: [cdr_csv.so] => (Comma Separated Values CDR Backend) [cdr_manager.so] => (Asterisk Call Manager CDR Backend) == Parsing '/etc/asterisk/cdr_manager.conf': Found

Hi, I have asterisk,openh323-v1_13_5 and pwlib-v1_6_6 installed on my PC. each time i run asterisk -vvvvc, i get the following error: [chan_oh323.so] => (OpenH323 Channel Driver) == Parsing '/etc/asterisk/rtp.conf': Found == Parsing '/etc/asterisk/oh323.conf': Found [1]WrapH323EndPoint::WrapH323EndPoint: Compile-time libraries OpenH323 v1.13.5, PWlib v1.6.6 segmentation

Hello to all, I have a strange behavior of my asterisk box. I'm running asterisk with asterisk-oh323 channel driver and everything works very well. But after few hours, my asterisk stop running and I have to restart it by typing "asterisk -vvvc". Most of the time I connect to my asterisk with a remote host so I don't know exactly which error causes my box to stop, but I found on

-----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'

Thanks Vamsi I have not been able to locate pwlib - 1.6.6 or openh323 1_13_5. I found the latest versions through sourceforge and I found some older versions on another site, but not these versions. This has been quite frustrating. Anyway, I think by using the asterisk-oh323 branch under channels in the asterisk source tree I will have more luck. At present it seems to compile successfully, but

Hi there I have Asterisk running beautifully on our test server. Over the past few days I have been tearing my hair out trying to compile various versions of asterisk-oh323 on various versions of pwlib and openh323. pwlib is now up to 1.8.3 and openh323 is now 1.15.2 stable. asterisk-oh323 is currently 0.7.1 I have tried these three with many errors. I have tried 0.7.1 with pwlib 1.5.2 and