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...nt. Iax.conf... register => username:secret@iax.host.net [username] type=friend context=iax-in user=username secret=secret auth=plaintext host=iax.hust.net ---------------------------------------------------------------- Message: 1 Date: Thu, 23 Dec 2004 17:55:29 +0000 From: Mike Dent <mcdent@gmail.com> Subject: [Asterisk-Users] IAX2 calls failing one way. To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <9e769e4e041223095564d9e749@mail.gmail.com> Content-Type: text/plain; charset=US-ASCII Hi, I'm having troub...

Hi, Does anybody else have problems like this. I'm in the UK with a 1mb ADSL service from Eclipse. I have a Draytek Vigour 2600 ADSL router. My * box is configured with a public IP address which is presented on one of the switch ports on the rear of the router. When there is some SIP activity, incoming mainly, towards my * box, the router will lockup after a short period?! I've tried

Hi, I'm slowly getting to grips with *. My next quest is to get IAX2/FWD calls working. I've setup a fwd account and added IAX capability to it via the website. I got the email saying it had been done with some welcome text and sample phone numbers to try, such as 10001 for the answer phone. I followed the setup example on the fwd site for configuring * to work with fwd's IAX.

Hi, I have a Cisco 7960G phone for which I know the pinout of the headset socket. I have a couple of standard telephone headsets which I do not know the pinout of. I'd like to connect the two. If I have the pinout of a normal/standard headset I can rewire the ones I have to match the cisco. Thanks Mike

Hi, it seems if a user leaves voicemail and hangs up the call when done, then HasNewVoicemail never gets called on the next line in the context. However if they press # to finish their VM, then it moves to HasNewVoicemail and this works? eg:- ...... exten => 2002,3,VoiceMail(u${OFFICEVM}) exten => 2002,4,HasNewVoiceMail(2002) ...... exten => 2002,105, do something cos vm has been

Hi, I've recently got * working (thanks Clive and list!) at home. We have 2 PSTN lines connected via X100P cards. I've got 3 x SIP phones (2 are Budgetone, the other is a Tecom SIP). One of the lines is our standard home line, the other a business line. Presently I've got * set so you dial 91<tel no> and 92 <tel no> to select which line to dial out on. I should probably

Hi, Is there some script which can be called from a * extension to playback the recent incoming callers on a particular PSTN line? In the UK 1471 is a BT number which plays back the most recent callers number, it also gives you the option to call this number back (now charging you for this service too!). Is there anything similar in asterisk-land? thanks Mike

Hi, I recently upgraded the firmware on my Blackberry 8700 to 4.2, this seems to give it the ability to play wav files. I wondered if anybody out there had managed to get their BB to play the wav files as attached to the Asterisk voicemail emails? Mine seems to ignore the attachment. I am using BES 4.1 for sending these emails out via Exchange 2003 if that makes a difference. thanks Mike

Hi, Anybody here from the UK using Asterisk at home? I'm looking for a SIP phone which will work with Asterisk and not leave me broke! I got one of the Tecom ones from Solwise but it refuses to login to Asterisk server for some reason. May have to send it back. What are the other options please? Thanks Mike

...------------------------------- ------------------------------ Message: 3 Date: Tue, 11 Jan 2005 04:12:53 -0800 (PST) From: Frank Kostin <frankostin@yahoo.com> Subject: Re: [Asterisk-Users] Vmail.cgi - "Hrm, can't seem to open /var/spool/asterisk/voicemail .... To: Mike Dent <mcdent@gmail.com>, Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <20050111121253.28205.qmail@web80903.mail.scd.yahoo.com> Content-Type: text/plain; charset="us-ascii" Hi, Just doing a "chmod" OK Halas, not a spe...

Hi, I've been setting up * recently and slowly getting to grips with it, however I'm getting rather confused with all the different configs for IAX calls, FWD calls iaxtel etc etc. What I think I need it a basic understanding or even a comparison of these different voip systems (if thats what they are?) I'd like to be able to make calls to other voip users, both in the UK and abroad

Hi I have been trying for a while to find a way to get an SMS send when I receive a voicemail into my asterisk system. I don't want to send an SMS if the caller doesn't leave a message. I have voicemail.conf set up to email and delete. 302 => 302,Website Sales,sip@example.com,,attach=yes|delete=yes However I can't seem to find a way to test is a message is left. I have tried

Hi, Whilst trying to get this Tecom phone working with Asterisk, it seems to be unable to login. Using the 'sip debug' command from the CLI does not produce any output even though the debug of the phone shows it trying to login every second or so? The phone seems to be based on a "Centrality PA1688" processor. Is there any other way I can see why this phone fails to login? I

Hi, I've recently set Asterisk up, 1.0.2 version. With 1 x X100P card and 1 SIP phone. I've noticed some horrible buzz/rasping type of sounds! These seem to occur when * is trying to play back some audio or sound to me? E.g. If I have an exten rule which plays one of the music on hold files when I dial 800 lets say, I get a really loud buzz for about 2 seconds and then the music plays.

Hi, I'm running * 1.0.2 . Which patches can I used against this version to get BT callerID to function please? Thanks Mike

Hi, now I've got caller id working on my BT line in the UK, I'd like to play a different message to those pesky sort who with hold their outgoing number. How can I do this in my extensions.conf for my [incoming-analog] context? I realise some people may call who are unable to change the way that their system withholds the outbound number, so I'll give them chance to leave a voice

Hi, I'm having trouble getting IAX calls through in one direction. On the receiving end I see:- Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00014ms SCall: 00001 DCall: 00000 [1.142.164.215:4569] VERSION : 2 CALLED NUMBER : 2003 CALLING NUMBER : dave CALLING NAME : Dave Smith LANGUAGE : en CALLED CONTEXT :

Hi, the topic says it all really. Does the Sipura 3000 detect and report UK clid correctly? thanks Mike

Hi, I'd be interested in comments from any users of the vonage service in the UK? http://www.vonage.co.uk is the website. Where are the servers located, traceroute would be useful. What is the general reliability like? Thanks Mike

Hi, Is it possible to put some kind of bridge which will do traffic shaping/prioritising between my 6 external IP addresses and my PPPoA modem interface? My other option is to put some kind of device at the edge of all my networks to shape the traffic in/out. I'd rather do it in one box if possible? thanks Mike