similar to: SPEEX CODEC and Voicepulse

Voicepulse has ignored four emails over the course of two weeks. Anyone have any ideas of whats wrong? - Executing Dial("IAX2/voicepulse-in-01@66.234.228.170:4569/7", "IAX2/acctname:acctpass@gwiaxt01.voicepulse.com/14109649073") in new stack -- Called acctname:acctpass@gwiaxt01.voicepulse.com/14109649073 Sep 13 22:48:25 WARNING[131080]: chan_iax2.c:5375

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

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

hi, I'm currently trying to use Speex with Asterisk from my OpenH.323 client. It seems to mismatch the codecs, below is my log from Asterisk. My Openh323 client crashes in responding to a Speex request for bits per frame. I'm guessing it either isn't running the codec correctly or doesn't support the same subset of speex codecs as openh323. (I'm using speex-1.0.1 with

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

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

Hi. I'm having a terrible time with call quality coming into my * box. I'm using VoicePulse over a 1.5/1.5 mbit line. Outbound calls are crystal clear on both the RX/TX sides of the conversation. Inbound calls, though, are HORRIBLY garbled on the RX side. I can barely hear the caller, but they report my quality is fine. Getting loads of garbled sounds and weird echoes. (Could just be

Samba 3.0.28-0.1.95-1624-SUSE-SL10.3 A strange problem (best read in a proportional font). It only happens on an x64 XP client when accessing a Samba share. The exact same program runs fine on the same x64 XP client when the share accessed is on a Windows server or when it is run on a 32-bit XP client, regardless of whether the share belongs to a Samba server or to a Windows server. I have

Just working on Configing up Voicemail and now that I have got it working and configed and answering the way it should be I have another challange. on the * CLI> I get this -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/6001/INBOX/msg0000 format: wav49, 0x8133390 -- x=1, open writing:

Greetings, I have been experimenting with Asterisk for a few weeks and finally decided to take the plunge and purchase a few DIDs for inbound calling. Our attempts at IAX/IAX2 connectivity with VoicePulse have been less than successful. We get "Registration Refused" errors from Asterisk whenever we launch the server. The front-line support folks at VoicePulse suggested that we are

All, I've almost got my Asterisk PBX setup, but I've having some problems with the VoicePulse IAX trunk. On outbound calls, when dialing a PSTN number through the IAX2 trunk, music on hold (moh, using the m option in the dial command) does not work. The console states that "stop sound" on IAX2 channel. Ring works, but only without the r option. MOH works when trying to dial a

For this week's conference, the two founders of Voicepulse, Ravi Sakaria and Ketan Patel, will be joining us. For those of you who are not aware, Voicepulse is an asterisk friendly VOIP provider that has won awards for service and innovation. We will also have Trixbox news, updates, as well as discount codes. Lastly, we are working feverishly to bring you more information regarding legal

Okay, I can receive calls through Voicepulse fine. All the various attempts (too many to list) to create a workable configuration to Dial to Voicepulse has failed, from 403s to "No authority found" to nothing. The Voicepulse folks told me that the open access was SIP and I shouldn't have a reference in the iax.conf file, but then said that they were refering my question to the

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

Greetings, I am new to all this VoIP stuff and have been having a bit of a hard time getting my soft phone working as a SIP client thru Asterisk. I apologize to start off with such a simple question and hope it's ok to post this and see what others have done. THE GOOD NEWS: I have successfully setup Asterisk 1.07 on an OSX machine. The build is running and working successfully. I am able

Greetings, I am new to all this VoIP stuff and have been having a bit of a hard time getting my soft phone working as a SIP client thru Asterisk. I apologize to start off with such a simple question and hope it's ok to post this and see what others have done. THE GOOD NEWS: I have successfully setup Asterisk 1.07 on an OSX machine. The build is running and working successfully. I am able

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

hello, after playing with an asterisk configuration for voip for a few weeks i'm trying to get outbound dialing with voicepulse going - i've cut down the asterisk to a very minimal install (1 SIP client) to try to localize the problem. The SIP client works fine (SIP and * on the same NAT) and could access the demo from samples before i removed it, and can call itself - so i am

My goal is to have only 1 primary phone number that can seamlessly "pool" multiple VoicePulse accounts. Let's say I have 3 accounts with VoicePulse Connect 212-555-1000 (primary) 212-555-1001 212-555-1002 When I receive inbound calls on 212-555-1000, I want to "forward" or "roll over" the connection to 212-555-1001 and 212-555-1002 so that the 212-555-1000

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