similar to: Using rxfax over SIP

Now this is really frustrating. Everything was working fine, and now it isn't ... I don't think I've changed anything that would affect this, but I guess you never can be too sure. My setup is as follows: SIP softphone (SJphone) connected to Asterisk running my Linux NAT firewall box. This is all on the internal network. Asterisk then dialing out through various means - SIP to

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 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]

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: --

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 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

Sorry not to have observed etiquet and lurked here for a bit before wading in with a question but I have an issue that may well be because I dont know enough about what asterisk is actually doing under the hood to understand why I cant do what I want with asterisk. Im hoping that someone can point me in the right direction :-) This is what I have: Mandrake 2006 running Asterisk 1.2.3 - no

Hi, Has anyone got Gossiptel working with Asterisk? - I am having real problems getting it to register - i'm just getting timeout errors. Thanks --ian

Has anyone out there used the PipeMedia/PipeCall PSTN gateway? Anything good/bad to say about it? I'm considering using them for a new customer. They seem to have good rates, good provisioning tools and (better still) give commission on usage to dealers. -- David Gurr Congruity Ltd. Fax: 0871 661 1756 Hemel Hempstead UK

I'm building a phone-in demo system to use for introducing Asterisk to prospective clients. One of the things I'm wary of is their likely preconceptions that VoIP systems will have poor audio quality. As a result, I'd like to ensure that the voice prompts I'm using have the best possible audio quality. Is it possible to use sound files at higher than 8kHz sampling? My callers

-----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

I'm looking for recommendations for UK-based VoIP-PSTN gateways. They should ideally offer: - IAX connection - Multiple simultaneous calls on a single account - Lower call rates than BT Business - Auto-top up or invoicing in arrears I can find several that offer one of these facilities, but none that offer all. Thanks! -- David Gurr Congruity Ltd. Hemel Hempstead, UK

What FXO interface methods are folks using successfully in the UK? I'm looking for good, known-to-work solutions for commercial use for two PSTN trunks on an Asterisk box. Here's the options I have, as I see it: i) Two Digium X100Ps. Pro - cheap (c. ?120), CE approved. Con - UK line impedance mismatch, with resulting echo problems, plus needs two PCI slots. ii) Digium TDM400P with two

-----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

Can someone confirm what I should expect the correct behaviour to be on incoming calls if I have multiple SIP phones configured for the same username? I'd expect all the phones registered under the username that that extension is associated with to ring, and the first one that answers gets it. What I get, is just the first phone that registered gets a ring. The second one doesn't ring at

Hi, Below if the error message which I got from asterisk. I was trying to fax to asterisk from my fax machine. I really dunno what is the problem. I use alaw & ulaw codec only through my ATA 186. Can anyone help me what could be the problem. -- Executing Goto("SIP/-080ef9a0", "13732|s|1") in new stack -- Goto (13732,s,1) -- Executing

Hi, I have Asterisk setup and registered with Gossiptel but i'm only getting 1 way audio. If I call 160 (echo test) or 123 (talking clock), it makes the call but I just get silence. If I call my Gossiptel number from a pstn line, I get gossiptel -> pstn audio but not pstn -> gossiptel audio. I've got ports 5060 and the rtp ports forwarded in on the firewall and I have 3 other sip

I'm curious as to folks experiences in selling asterisk-based solutions. In sales-speak, what are the common "compelling reasons to buy"? I can think of the following potential ones, but I'm keen to find out what seems to work in practise: - Customer wants to cut cost of calls, implements * and signs up to a VoIP/PSTN gateway - Customer wants a new PBX but doesn't want to