similar to: voicepulse problems since new configs

Hi, for some weeks now I have been unable to make calls via my voicepulse connect IAX account? When I attempt the console looks like this:- rt*CLI> -- Executing Dial("SIP/2008-cf55", "IAX2/NBhXXXXXX:XXXXXXN82@gwiaxt01.voicepulse.com/12124565900") in new stack -- Called NBhXXXXX:XXXXN82@gwiaxt01.voicepulse.com/12124565900 -- Call accepted by 66.234.228.160

I'm trying to use the SPEEX codec with Voicepulse. Here's what I see in the CLI when I RELOAD: -- Reloading module 'codec_speex.so' (Speex/PCM16 (signed linear) Codec Translator) == Parsing '/etc/asterisk/codecs.conf': Found -- CODEC SPEEX: Setting Quality to 5 -- CODEC SPEEX: Setting Complexity to 5 -- CODEC SPEEX: Perceptual Enhancement Mode.

Is any one else having problems with Voicepulse today? Suddenly, I can't register and calls to my Voicepulse numbers get a fast busy. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815

I have been picking at Asterisk for about a week, and I think I'm close. I was hoping for a little guidance to bring this on home. I want to be able to make outgoing calls from my SJPhone clients using my VoicePulse Connect account. I have the two requisite items from Voice Pulse, but I've had no luck successfully integrating the VoicePulse settings into iax.conf. My current config:

I am wondering if there is a way to create a SIP/IAX group of outgoing lines like Zap groups. I am currently using the following method, but would like to use features such as ?g2? that would list all the accounts for a SIP or IAX connection. exten => _1NXXNXXXXXX,1,Dial(SIP/account_name:Password@gw5.voicepulse.com/${EXTEN }) exten =>

I attempted to install an X100P card but it was not correctly recognized by my Redhat 9 install. I had a test install running without any cards which was working great minus the outward dialing since no cards existed. Now that I have a card, I want to add it to the system. Do I have to scratch the whole current install in order to get the X100P running on my system or is there a way to get it

CentOS-5.6 fuse-2.7.4-8.el5 fuse-curlftpfs-0.9.1-1.el5.rf I am trying to mount an ftp connection as a local file system using fuse and curlftpfs. I can connect to the remote system (an HP3000 running MPEiX 7.5) and the directory seems to mount but I cannot use it. What I get is an Input/output error: $ curlftpfs -v hahp3k01.harte-lyne.ca tmp/hp3000 * About to connect() to

Hi- I built openssh-3.0p1 on a Tru64 4.0G without any problem. The system uses enhanced security, so the sia_* routines are used by sshd. Unfortunately, password authentication fails because sia_ses_authent() returns 0 in auth-sia.c. The thing is, the password is CORRECT; I verified this by inserting debugging statements before the call to sia_ses_authent(). The call to sia_ses_init()

I've been looking at several posts for weeks now and finally concluded through testing how to install Samba 3.X into the Windows Active Directory environment. I was completely under the impression that you needed to load Kerbos/ LDAP and a bunch of other stuff. In our case our ADS is running in native mode and I was able to join the domain quite easily. I've tested authentications and

Greetings, I'm having a miserable time getting Asterisk working with FWD. All the samples show something like... exten => _7., .... How do I get Asterisk to wait until the user is finished dialing instead of trying as soon as it gets the second digit? I can use _7XXX, and dial the FWD 3-digit test numbers fine, but I'd like to be able to dial others... Same problem for outside

I've been using my voicepulse connect number for over a month now, but today it simply won't connect. My partner and I each have a number, both are mapped in my iax.conf and extensions.conf files. This has been working fine. Today, either number gives this message: Feb 16 21:53:14 NOTICE[4330]: chan_iax2.c:5757 socket_read: Rejected connect attempt from 66.234.228.170, request

Help. Why is it that I can call out from my GSBudgetone SIP phone but the audio is "one-way'? Why is it that when I call my asterisk phone number, I get a fast busy?

I posted this question a while back, and I'm posting again in hopes that someone has some ideas. Sorry if you've already seen this. When dialing out using a SIP or IAX provider (Broadvoice, SimpleTelecom, VoicePulse Connect) I often find that after the call is answered the first few seconds of audio are cut off (i.e. I don't hear the called party). This usually results in the called

I have been having problems getting PrivacyManager to work correctly. Right now I am running the 1/21/05 CVS but I have been unable to get this to work on asterisk-stable either. You can see from the debug below that the inbound call is arriving via IAX2 and both the CALLING NUMBER and CALLING NAME are both set as "Unavailable". However, PrivacyManager executes and determines that

I am having some problems getting TelIax service to work with *. Outbound calls work just fine. When I try an inbound call the phone rings and there is no audio. Upon further investigation "iax2 show channels" indicates that the codec is "unknown" The provider confirmed that they are set for ulaw and so am I. Does anyone have an idea what could be causing the codecs to

Is there anyone out there who has VoicePulse Connect working with DTMF? I've been unable to get it to work from the start, and the recent VoicePulse updates did not help. A caller to my DID's hears Asterisk, but pressing DTMF does nothing: On call setup "iax2 debug" shows: ----------------- Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK

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I am trying to do the following: 1. Call comes in to my * box over IAX (VP Connect DID) 2. Check to see if call should be forwarded to my cell 3. Forward the call to my cell phone and take * out of the media path. I am able to do all of the above except * is not able to natively bridge the call. I am using sixtel and for the call forward portion, but the calls don't connect before sixtel

Hallo Did you allow udp outgoing on 4569 as well.. i found udp bit different than tcp when comming to firewalls liaan ----- Original Message ----- From: "Bartosz Wegrzyn - asterisk" <junk@lexon.ws> To: <timebandit001@gmail.com>; "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Monday, February 21, 2005 12:29

I recently switched from BroadVoice to VoicePulse Connect on my Asterisk box. Too many Meetme quality complaints (whether real or perceived). I had to make a choice to use IAX2 or SIP with VoicePulse. I first tried to go with SIP because I already had it working and all of our devices are SIP. Problem is that every time I turn my back, the Asterisk registration with the VoicePulse SIP server