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Hello, I'm wondering if anyone else has been observing problems lately with 1.4.18 and higher releases of asterisk not properly recording calls. When using MixMonitor, the resulting file is only a few bytes long. I think this is because asterisk is doing Native bridging even though MixMonitor should block that. Did something change around the release of 1.4.18 that would have changed

Hello, In my quest to create several proof of concepts for what can be done with Asterisk, I've run into a bit of a problem. I have a pair of SPA-2000's acting as off premise extensions for an analog line. When a call waiting call comes in, the caller id information makes it though the ULAW codec and displays on the caller id box, however asterisk doesn't seem to want to pick

I don't suppose anyone might know why I hear ringing transposed over itself when I place a call out via PRI? SIP to SIP is fine SIP to IAX is fine SIP to PRI is always transposed I mean sometimes you don't notice it much because it's lined up right, but other times you'll hear a really long ring (starts sounding normal, then sounds "weird" -- like two rings played at

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

I've seen lots about presence and Polycom phones recently. I've got one here for evaluation but noticed other phones only seem to appear busy when they initiate a call. If they receive a call, they still show as available. Is this a config problem on my part, or is that as far as presence is working right now? Thanks! Trev

It seems to never fail - after 3 to 5 minutes SIP -> IAX calls drop audio on one side. I place a call out through voipjet, and call quality is flawless. However a few minutes later the person who I'm talking to can no longer hear me. I can still hear them. What should I look for to resolve this? Has anyone else had this problem? Using last night's CVS this problem still exists.

We have a PRI from Telepacific. Asterisk 1.2 and a Sangoma A101 T1 card. Outgoing calls to certain toll-fee (8XX) numbers fail -- we hear ringing but the calls are never answered. All other calls, and most toll-free numbers are not affected. The numbers that are affected are all travel related companies (United Airlines, American Airlines, US Air, Starwood Hotels, etc.) we cannot connect to

Hello, We have an asterisk 1.2.13 box that use a Digium TE205P T1 PRI connect to the telco, users mainly use snom 320/300 SIP phones. When dialing to an external phone number with custom ringback music, users reported that they could not hear the music but can only hear the standard ring tone generated by the system. Is there any kind of settings need to allow the ringback music pass to the

...; http://lists.digium.com/mailman/listinfo/asterisk-users _________________________________________________________________ Stay in touch with absent friends - get MSN Messenger http://www.msn.co.uk/messenger --__--__-- Message: 2 Date: Mon, 24 May 2004 23:44:30 -0700 From: Trevor Peirce <tpeirce@digitalcon.ca> To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Document - contains malware Reply-To: asterisk-users@lists.digium.com hank wrote: >is this a virus? > Yes. --__--__-- Message: 3 From: "Jay Milk" <jay@skimmilk.net> To: <asterisk-users@l...

...unknown to alaw Mar 18 23:41:06 DEBUG[10406]: Didn't get a frame from channel: SIP/201-3e46 Mar 18 23:41:06 DEBUG[10406]: Bridge stops bridging channels SIP/201-3e46 and Modem[i4l]/ttyI1 ------------------------------ Message: 27 Date: Fri, 18 Mar 2005 08:28:27 -0800 From: Trevor Peirce <tpeirce@digitalcon.ca> Subject: Re: [Asterisk-Users] PRI Cause Code Help To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <423B01AB.40901@digitalcon.ca> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Peter Svensson wrote...

Hello, I'm writing an AGI script that receives an incoming call, records the caller's name, places an outgoing call and plays the name back then asks if the call should be accepted or sent to voicemail. I know I can use Dial to call another number and bridge the call, but I need a much more advanced solution. It seems I'm missing a Bridge command, or something of that nature,

Hello, I know not too long ago I saw /something/ _somewhere_ about an adjustment to call parking that allowed blind transfers from SIP phones to park a call and still be able to hear the parking lot stall number. Unfortunately, I have no idea where I saw that (google turned up little, couldn't find it on the list either). I'm using Sipura SPA-2000 adapters and it doesn't seem to

Has anyone tried using a debit/credit card terminal as such: Terminal <-> SPA-2000 <-> Public Internet <-> * <-> PRI I'm hoping someone will tell me they have done this successfully and rarely experience dropped calls. Though I'd like to hear from anyone who has tried and failed as well. Thanks, Trevor Peirce

Hello, I've got some SIP clients, and an IAX2 long distance provider. Ideally, when a the dialed number is busy I will hear a busy signal. Instead, I get Congestion even though * knows it's busy. Is this a bug or am I missing something? The dial plan, in basically this Dial(IAX2/user@provider/19995551234,,) Goto(failedcall-${DIALSTATUS}) failedcall-CONGESTION plays congestion

Hello, To assist Matt's efforts in bug 3554 (2BCT & CNAM), I'm hoping someone can provide a dump of the setup and related messages from a PBX that supports outgoing Station Name to the CO. As suggested in the bug, I tried to ask my telco for a dump of the setup messages for a client that supports this but was told to contact my vendor as they cannot provide that information.

Hello, I just got off the phone with my PRI provider, and was told that I am not sending an expected message when I reject a call with a Cause Code for Unassigned(1) and Congestion (42). Busy works fine though... Anyhow, they are seeing the RELEASE COMPLETE message with cause code 1, however the tech told me they expect a PROGRESS indicator with a value between 1 and 10. Any ideas on how

Hello, Is it possible to set up a timeout for IAX when something like the following happens? -- Executing Dial("SIP/someone","IAX2/somewhere|45") in new stack -- Called somewhere -- Call accepted by 1.2.3.4 (format ulaw) -- Format for call is ulaw <nothing happens here for 15 - 30 seconds - caller gets tired of listening to silence and hangs up> -- Hungup

Hello, I'm at a loss for a way to find the serial number of a Digium analog card without physically removing it from the server. The only time I have physical access to this particular installation is during business hours and that's obviously a bad time to be taking a server down. It seems that I need the serial number to get a free copy of HPEC... but unless someone can convince

Hello, I have some Aastra 9112i's in production that almost function flawlessly. The problem I'm having is when a caller is put on hold they do not hear hold music. If they are on hold for too long (~ a minute?) they are hung up on. All other phones including Aastra 480i and Sipura/Linksys ATAs all seem to be working fine. Is this a quirk anyone else has experienced? Any