similar to: Voicepulse down

Hi all, I have the below extension defined under sip.conf: [2203] type=friend username=2203 secret=123456 host=192.168.0.164 mailbox=2203 context=intern canreinvite=yes dtmfmode=rfc2833 When trying to register from a softphone installed on a PC behind a nat with IP=192.168.0.164, I got 503 FOrbidden...Does anyone have any idea about what could be the issue? Regards -------------- next part

Anyone know where I can get an incoming DID for Bogota, Colombia? Fred Posner fred at teamforrest.com Tel: +1 (212) 937-7844 x501 Fax: +1 (954) 252-4187 www.teamforrest.com -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 2162 bytes Desc: not available Url :

Here's the story... Nortel system set to use g711 @ 30ms payload ... Asterisk box would need to communicate to that box @ 30 ms and another end point at 20 ms. I've seen discussions of setting this to a different size, but seems to be limited to the entire codec and not on a per peer basis. Anyone have luck with this? The Asterisk can be 1.4 or 1.6.x... I've a preference for

I missed the talk that was given on wideband codecs @ astricon last week. I tried to lookup the speaker on astricon.net, but that website seems horribly broken at the moment, showing only a tmcnet video, whatever page i click on. Would somebody have the contact details for that speaker ? Greetings, Zoa

List, We're working on an upcoming job that may require us to access a web service (WS). I'm curious to hear peoples thoughts on the best way to do this with asterisk. We'll be submitting a single number to the WS and it will return a success or error. One solution would be to write a simple perl script to interface into to the WS, and use SYSTEM() from asterisk to call it.

Just a "heads-up" ... my home asterisk server is being flooded by someone from IP 184.73.17.150 which is an Amazon EC2 instance by the looks of it - they're trying to send SIP subscribes to one account - and they're flooding the requests in - it's averaging some 600Kbits/sec of incoming UDP data or about 200 a second )-: This is much worse than anything else I've

On Fri, Jul 30, 2010 at 1:15 PM, Fred Posner <fred at teamforrest.com> wrote: > On Jul 30, 2010, at 5:04 AM, Andra=C5=BE wrote: > > > Ok, problem is another, when I run configure, it write this: > > checking for tds_version in -ltds... no > > configure: *** > > configure: *** The FreeTDS installation on this system appears to be > broken. > >

Hi All; How to increase the waiting time between entering the digits for the analoge phone device that is connected to fxs? Is it by DigitTimeout? But how it will be apply for analoge station if the user just pickup the handset and dialed the number? Any help? Regards Bilal

Hi, Inspired by a recent rant about one particular provider, I am getting very curious about something I've never mastered. I'd like someone to explain this here or at least post a link or two that can educate me and probably countless others who have no knowledge in this area. I'm sure there are several of you reading this that know all about the subject. What are the various

I have two questions regarding number portability... 1) If I port a DID over to Voicepulse, can I then move that DID elsewhere somewhere down the road. Or does voicepulse now OWN that DID? 2) Can I take a DID assigned by Voicepulse and transfer it to someone else? If not, why? -jwb

I'm looking to set up multiple inbound Voicepulse Connect lines and have Asterisk route them direct to different IVR or Voicemail based on the inbound number that is called. Unfortunately, I just can't see how one would go about identifying the number that is being called. Has anyone been able to do something like this with Voicepulse? I appreciate any assistance. Phil

Has anyone been able to get * working with VoicePulse OpenAccess (SIP not IAX). I have found a ton of information about VoicePulse Connect but very little on the proper * settings for OpenAccess. Tried contacting VP with no response. If anyone has this working, can they share their extensions.conf and sip.conf files? Better yet, if it could be posted on the Wiki. Keith

I have a DID with VoicePulse Connect, but the sound quality is horrible, it is often choppy and the caller's voice cuts out for 2-3 seconds at least once a minute, I have contacted VoicePulse many times, and they do not do anything about it! Does anyone have any similar problems? It isnt my Asterisk config because I have 0 problems using NuFone.

I can ping it just fine. I am on gw5.voicepulse.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040615/054c83f4/attachment.htm

Asterisk Users, Has anybody use Voicepulse Connect for Asterisk? I am trying to cover all my bases because in the past, I got burned with poor quality of service, along with failed DTMF tones with 3 different SIP Providers (Vitelity, Broadvoice, and Teliax). I am running Asterisk 1.2.13 on the Debian Etch system, using the SIP protocol. Any insights would be great. Thanks. -John

Dear Sirs, Anybody ever tried running SIP up against Voicepulse? On their http://connect.voicepulse.com they claim they support both SIP and IAX, but I can't seem to get SIP running. I have as mentioned before on this list - huge problems getting any timing devices running on some of my machines, so IAX is not really an option right now. If I try I get a "Service Unavailable"

I'm having trouble configuring these services the way I want. Basically I prefer using iLBC before GSM, however Iaxtel only want to talk GSM. It _seems_ that Voicepulse prefers GSM also, because even if I put ILBC before GSM in the "allow=" part of iax.conf, if GSM is mentioned, Voicepulse will use it. If I don't allow GSM Voicepulse will use ILBC. Does anyone know how to

Is anybody else out there using VoicePulse Connect and having problems this morning? I just noticed that they have absolutely no contact information in their website.. just want to make sure I didn't break something in my asterisk configs. -fedl

I don't know if this has been mentioned yet: Voicepulse is now offering wholesale pricing and IAX2 connectivity for Asterisk users. No fees, pay as you go. They also offer incoming calls for $7.99 per month. See wholesale.voicepulse.com.

Until recently, I was using Broadvoice for my in/out calling thru Asterisk. I was extremely pleased to see that Broadvoice was actually passing the callerid info (number and text) that I had set up on each device in my SIP.CONF file. I had PSTN users tell me that they were actually seeing name and extension info when I called them from the Asterisk box. Last week, due to numerous user quality