similar to: Which end hungup?

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 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 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 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

-----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? --

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 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

-----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'

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

Hi. [I asked a similar question a while back, but unfortunately wasn't around to reply to the responses, so sorry if you experience any deja vu.] I have a * server acting as an IVR system. The calls come in via IAX. After a period of about 40 seconds of silence (either waiting for the caller to dial an extension, or with the audio paused in controlplayback), the call hangs up. All I see in

I installed this card, everything work, i can make call and receive call with no echo and great sound quality, but after between 5 to 50 secs the call disconnect by itself, in the log i don't see nothing revelant. I don't share any IRQ, zttest show me values between 99.98 and 100. The only thing i see, the pstn line is not realy a true pstn line, it's plugged in an Arris Telephony

Hello! There's the "g"-option for the Dial-cmd that allows to execute the next extensions in the current context when the callee hangs up. I would need the same for a call where the caller hangs up, concretely i have to inform a agi-application of the end of a call. Does someone know a way to do this from the dialplan? thanks Christian

Hi, I'm running SER with Asterisk, and I've configured VoicemailMain like this: exten => 201,1,VoicemailMain(@default) exten => 201,2,Hangup() Although, after any user enter his voicemailmain mailbox, when the phone is hung up, the call still continues running in Asterisk, because I can see it in the debug output of the Asterisk CLI. The call only stops if before hung up, I

Hi, I'm tying to delay the HungUp. I tried this way: exten => h,1,NoOp(Start) exten => h,n,Wait(5) exten => h,n,NoOp(End) exten => h,n,Hangup() but it doesn't work, Any idea? Thanks in advance.

I've got the following setup: IAXy -> Dect Base Station. When you dial from a SIP phone (cisco 7960), the rings with very little delay. However, if you hangup it takes 3-4 rings after hanging up before the dect base station phone stops ringing. The same applies when an incoming call is directed from PSTN FXO -> Dect Base. Is there a fix to this I've looked about on voip-info but

Hi all, We have tested both tdm400 and BRI 1 port HFC based with mISDN driver and with both cards we have the same problem... so probably it's due to our misconfiguration, but we couldn't find anything useful on google. With TDM400 we can place and receive calls, but if we receive a call from TDM400 and we reject it from a sip phone , asterisk send the hangup but the line continue to

Hi All, I am facing the following issue, while using Libvirt version 0.8.4. (1) Libvirt API does not return, since the qemu-kvm process is hung up. Based on the information available on mail thread *https://bugzilla.redhat.com/show_bug.cgi?id=676205* (comment 26 - scenario 1), this has been fixed in later version of Libvirt, specifically in RHEL 6.2(libvirt 0.9.4). I have checked the