similar to: Analog Bridged Calls Pulsate

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

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.

test only. good luck! james.zhu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070807/0fd2b827/attachment.htm

hi again How can i extend 5 times for effect.pulsate beats? I need that continue beat until other event client happened about my other mail, is there other solution more elegant that this? : new Effect.Pulsate(leccion[i_leccion],{duration: 5,from: 1}); thanks again rag

Hi Everybody I've got a weird problem. I am running one Asterisk system on a dual processor box. This box mostly do VoIP only but it has a Fritz PCI ISDN card installed with latest drivers. Dialing out through the ISDN cards from an internal Snom phone works fine and so does dialing in. Except - if I load the ztdummy module (for IAX channels) the capi drivers starts acting up. It is hard

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

Somone have tested the last scriptaculous version that ships with last prototype? I you make an Effect.Pulsate, the element stays hidden after the effect finish if the element don''t have opacity stablished. This is for the changes on the setStyle method on prototype. The original code is:

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

Hey all, I have an as5300 I use for dial in customers, we have 4 PRIs on it. We have a few free channels on it. I'm wondering if I setup SIP on the as5300 I can have asterisk use the free channels for dial out. I'd still have to use my TDM04B for incoming calls, but at least I can expand my outgoing. Anyone done anything like this before? I've never messed with VoIP on Cisco

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

Hello, I've got my dialplan configured to do a double ring when a customer service call comes in, and a normal ring when an extension is dialed directly. When a customer service call is transferred, I want to ring to revert back to normal. In the local extension macro, I have the following ; make sure ring is set to default exten => s,n,NoOp(${ALERT_INFO}) exten =>

Maybe someone can help me to make the Effect.Pulsate working on Internet Explorer in this example? <ul id="list"> <li>Element 1</li> <li>Element 2</li> </ul> <script> Sortable.create("list"); new Effect.Pulsate("list"); </script> Thanks. --~--~---------~--~----~------------~-------~--~----~ You received this message

Hi there, can someone confirm a bug with IE and Effect.Pulsate()? When use the Effect wihtout the option pulses the element is hiden after the effect has finished. When i use Effect.Pulsate(element, {pulses:3}) all runs fine. In Firefox both calls runs fine... Bye, René --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google Groups

Hi Folks, Apologies to anyone who'd prefer not to see this query on this list; but I'm asking because it is probably the forum where I'm most likely to get a good answer! I'm interested in the provenance of the name "normal distribution" (for what I'd really prefer to call the "Gaussian" distribution). According to Wikipedia, "The name "normal

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

Anyway to make hitting `0` during a voice mail dial an extension? The bosses used to have that feature and love it. Their VM prompt would say: "Hello, My name is blah blah. I am currently unavailable. If you would like to speak to an operator press 0 now, otherwise leave me a message". Extension 0 exists, but dialing it during a VM prompt does nothing. Thanks, -- Daniel Jimenez

http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultaneous+registry From the WIKI: Contributions Manager: Daniel Jimenez (cuban) Bounty: $50 USD Date opened: July 10, 2004 Contributors: cuban ($50) Detail Yes, Yes I know you could do all sorts of fun with the dialplan to produce a similar effect, but I still would like to be able to do this. Plus it's easy money :). I

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