similar to: Routing through ENUM

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: --

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 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

-----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

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 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'

-----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

Since e164.org added DNC and ADDRESS records my enum configuration has failed. Using both the old EnumLookup app and the new ENUMLOOKUP function, the lookups have consistantly failed since e164.org added E2U+ADDRESS and E2U+DNC records. Mar 6 17:39:44 WARNING[14222]: enum.c:235 parse_naptr: NAPTR Regex match failed. Mar 6 17:39:44 WARNING[14222]: enum.c:354 enum_callback: Failed to parse naptr

This has been giving me a headache for a little while, so I thought I would throw the question out to the list. Imagine there is a database lookup function that takes place that, given a number, returns call routing information. It might be a postgresql lookup, it might be an enum lookup. If I have something like: exten => _.,1,LookUp(${EXTEN}) exten => _.,2,Dial(${RESULT}) Then the

Hi all, I have a problem with 1.6.1.7-rc1 and ENUM (with an own PowerDNS server and NAPTR record). Maybe somebody has more experience with this or can give me some input. The dialplan: exten => 292,1,Set(DIAL_NUMBER=43660123456) exten => 292,2,Set(sip= ${ENUMLOOKUP(+${DIAL_NUMBER},sip,,1,ns3.e164.xxx.com)}) ;x'ed out the domain name starting from here exten => 292,3,NoOp(${sip})

I'm running into errors when using Enum lately. I can't figure out what the problem might be as I've had Enum up and running in the past. I'm running the latest CVS-Head compiled version. I've also tried using the new Enum function with the same results. When doing a lookup on a number that exists in the enum server I get the following results: -- Executing

I'm setting up a private enum zone to simplify/centralize dialplans for a number of Asterisk servers. In several dialed number situations, there are a handful of possible destinations for the call, and I'd like to have * try ENUMENTRY1, ENUMENTRY2, .., ENUMENTRYN just in case the first result is temporarily unable to handle the call. In at least some cases, I'd also like the order in

Ok, so you've read the Wiki and gotten call routing using ENUM to work (http://www.voip-info.org/tiki-index.php?page=Asterisk%20E164%20Call%20Routing) with your own ENUM-alike domain, e164.example.com. But how do you populate it with data? You can do it manually, but that gets very tedious very quickly. Or you can use the nifty DDNS updating program that comes with bind9. The first thing is

Hello everyone, I playing around with ENUM and have configured * to query a few sources for testing purposes (fierymoon, e164.arpa, e164.org). I'd like to know if there is a way to query these servers manually (ie outside of asterisk via nslookup or equivalent) to find out if particular exchanges are listed with wildcards, so as to terminate calls to those prefixes (I'm not trying to

I've been tinkering with ENUM and found that the lack of a debug message in enum.c that says it has actually succeeded in resolving an address is a bit of a nuisance. It makes it difficult to see if failures with ENUM are due to problems with parsing NAPTR records (in enum.c) or mistakes in extensions.conf An extra line of debug information would be much appreciated! Iain