similar to: Asterisk and Voice Pulse Open Access

I can't seem to dial out with Voicepulse Open Access service using *. Incoming works fine. Another user posted a few weeks back that they were having problems and there are some threads at dslreports.com about this as well. Maybe someone here can figure out what the issue is from the sip debug info below. I am at a loss. The audible error message from Allison is 0984 (from VP server) Here is

Hello: I am new to the list. I am trying to set up asterisk with voicepulse. I have a voicepulse username + password, and SIP DID. When I login to voicepulse, I have this under my devices tab: Devices *Login:* Sysxxxxxxx *Password:* xxxxxxxxxx *Context:* VPWS *Connects to:* gw5.voicepulse.com My question is: Do I need a 2.4.x kernel? Currently I am running Debian/stable stock 2.2.x ? Has

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:

Hi, I have the setup of my xp100 plugging into my dlink gateway that i use with a voip provider. I notice that when someone calls my pstn # that goes to the asterisk box it works but when they hang up asterisk does not recognize the hangup. What needs to be done to make it work with a dlink gateway. when i plug the xp100 into my land line jack everything works fine and the xp100 reconizes

I am looking for some great instructions on using asterisk with pulse. I'm using centos 7 and pulse as a user and not having much luck. I have changed all permissions for the asterisk directories. set asterisk.conf user and group to be my user that is running. No go. Anyone done this? Thanks, Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am thinking of signing up with voice pulse connect to connect to my asterisk server and using it as a regular line. Is it good? Or should I go with vonage or others ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050124/16792f10/attachment.htm

Wanting to connect my asterisk box off of 2 unused analog extensions on the non* PBX system. Can I bring those lines in to * on XP100 FXO cards? Any special wiring/open loop issues to watch for? The desired config is inbound PRI -> non* PBX, attendant picks up and if caller wants services on * box then xfers call to one of the analog extensions where * picks up and handles call from that

Has any one found any issues (bug?) with ver 1.2.5 as CallerID generated on an outgoing FXS port (to the handset) fails when UK tones are used, with a message 'Didn't finish Caller-ID spill. Cancelling.' Any tips on getting this running ? Looked at Mantis, but only known bugs seem to relate to XP100 cards - not the TDM card. Thanks

Hi John, I've been using Packet8 via a physical ATA and XP100 card for some time. As far as I know it is not possible to connect to the Packet8 service without the ATA. If this is not the case I would be very interested to hear this. In addition since moving to the USA I now only have a single packet8 line into my asterisk box (I used to have this and a 2nd regular pstn line) I used to be

Hi I just signed up with voicepulse's voice connect service. then emailed me over configs for my extentions and iax i enter in all the info and when i start up * and do show registry it seems to be rejecting my login. Has anyone seen this before.. Any further insite will be greatly appreciated. thanks frankie (aim)cronparser (irc)crontibs 17006240093 -------------- next part

In the absence of "The Definitive Guide to Asterisk Dial Plans" book, I'd like to do something possibly unique with the formatting of extensions in my dial plan, and am having trouble. We use VoicePulse connect, which gives us local DID for inbound and outbound calls (even though DTMF tones are not working in Voice Pulse Connect at the moment). To dial local numbers, you have to

I have dealt with machines that have multiple network cards in them before, but never when they were on the same subnet so this issue has never come up before. My problem is that I can only access one IP address at a time. I started out using dhcp and found that if I went through the dhcp song-and-dance then that address became active and the other one was disabled, and vice versa. On our local

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Hi all, I'm running Asterisk 1.0.0. I am a customer ( and supporter ) of voicepulse. For me, it works perfectly, but one of my customers noticed a small problem: During a conversation, when the otherside isn't talking, it's almost like the mic turns off. Not that big of a deal I know, and the more I think about it, the more this seems a voicepulse issue. But in the off

I have a Centos 5 machine here that, up until about a year ago, was happily running Icecast and serving streaming audio through through three network connections, consisting of one "local" connection (local address 192.168.1.5) and two cable modems to talk to the outside world. We shut this down about a year ago, but now I am attempting to get it going again on one outside connection

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 am trying to setup a simple auto attendant with Asterisk using SIP extensions. I have an IP trunk to voicepulse and my outgoing calls go over that. I can also receive calls on that voicepulse trunk and want it to an auto attendant. Everything works except on the following: - one of the options is to allow the caller to press the extension that they would like to be connected to. I have

Hello all Here is the scenario: We have a mail server VM which currently has two virtual NICs attached to it. One NIC is has an IP on a subnet with a default gateway defined and the other NIC has an IP on a different subnet with a different gateway on a different VLAN defined. Now when I activate both NICs, and run an ifconfig -a, I see that both IP addresses are showing. Now here is the

Just a quick note that the priority in the Pulse AO plugin (ao/src/plugins/pulse/ao_pulse.c) should be changed from 41 to 50 (in the ao_pulse_info struct). Since the plugin now ships as part of libao, its priority should be a factor of 5, as that's how I designed the plugin priority system. Plugins that ship with libao should be a factor of 5, so that 3rd party plugins can fit in between

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