similar to: Voicepulse connect - unable to dial out, asterisk says "9696"

Send CentOS-announce mailing list submissions to centos-announce at centos.org To subscribe or unsubscribe via the World Wide Web, visit https://lists.centos.org/mailman/listinfo/centos-announce or, via email, send a message with subject or body 'help' to centos-announce-request at centos.org You can reach the person managing the list at centos-announce-owner at centos.org When

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

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

Anyone know how many simultaneous calls you can receive on a NuFone DID? -Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050310/e20d2eb8/attachment.htm

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.

Hi all, I get "Unknown RTP codec 72 received" message in console when call in progress, but all seems OK. Calls is from X-Lite SIP softphone to PSTN over voicepulse connect (IAX) and to FWD echo test (SIP). But this message only with one SIP client, others (X-Lite too) not giving this message. All X-Lite settings are identical. Asterisk is last cvs version This what I see in console

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