similar to: Sip registration change!

Did anybody ever come across an affordable, residential cable/dsl router with support for QoS? The ones I've seen so far (Netgear, D-Link and W-Linx) do not seem to support it. I noticed that even email can damage a G.711 stream on an 128kbit uplink, leave alone file-sharing applications. I understand this is strictly related to *, but nevertheless of interest to many of us. Thilo

Please forgive me if this is a silly question. I've been following this thread in the hope that I could put my * server and snom 200 into full-time service very soon. I need to find out how to have the lines configured so that it does not return a busy reply when only one call instances is engaged. Am I supposed to create multiple extensions on my asterisk dialplan to reflect the 5 call

Ok here is another problem I have run into. I have a Sipura 2000 and I have been able to configure line 1 with only one small problem. But I can't get the line 2 working with asterisk. Here are samples of my sip.conf and extensions.conf. If I disable line 1 I can then get line 2 working. Is there a sample configuration for the Sipura to get both ports working with Asterisk. Sip.conf

Hello all, Last couple weeks we had a lot of business discussions on mailing list, however some people don't like it, some people don't needed it, etc. I had couple discussions with Asterisk community members, who is interested to have business discussions about Asterisk, including but not limited to : business implementations, reselling , Asterisk commercial packages, IP phones,

hello, Is there a way to transfer the call via gnophone, without calling other user and pressing conf on both calls, it seems that all traffic is still going through the gnophone, not that optimal i guess. thanks -- Anton Yurchenko<phila@dg.net.ua> Digital Generation

Has anyone had any success using more than one or two X100P cards? I have 4 in a system, and channels 2 3 and 4 all seem to work just fine. Channel 1 however is acting up. I get random red alarms, disconnects, etc. I have checked the /proc/interrupts and everything is sitting on it's own IRQ. Also checked memory addresses and everything looks good there. Not looking for anything specific,

I would like to know if anyone has come up with a script for 911 dialing rules that put correct information on our locations. We have our office in 3 different building one being our production & shipping dock. It is almost 2 blocks away. We are connected with Ethernet Wireless between the buildings and have Sip phones setup in the other 2 locations. All the phones are working just fine.

I just got 2 IpDialog phones for use with my Asterisk system. I have been able to get the phones to just dial local extensions but it is not able to register with my system correctly. I would like to know if someone has set these phones up before and how they did it! Is there any examples for use with Asterisk? They seem simple enough to config with there web interface. Thanks

I am new to this forum. As well as a new user of Asterisk. My vendor installed the system and we are still trying to get all the bugs out of it! I have a few questions about configuration and a program to view who is on what extensions. I am looking for a program that will work on my Receptionist work station. She is running Windows 2000 pro. We have not plans on upgrading to XP pro so

I have about 5 snom 200 phones working fine with everything. Voicemail, Transfers and all. Except I can't seem to use them to pickup parked calls nor place a call on park. I also have sipura-2000 with analog phones that are able to pickup parked calls and to park them. Most of them are on firmware 2.04g I have upgraded one to 2.05c for testing but this did not fix the problem. I get no error

I am trying to get a building that has 3 company's on one asterisk server. I need to make the IVR via DID take them to there right menu. So far I have everything working except when they goto via standard_marco to an extension and are sent to voicemail they are dropped off in the first menu and not the one they came from. In other word sent to another company's menu. If it happens to be

I have a problem in that when you use IAX2 for trunking and have 2 nics one is used to connect directly to 2nd Asterisk server how do we get the outside Nic card to take IAX connections? Is there any way to get this working via two paths? There is only one bindipaddr=10.1.1.1 for internal trunk but outside address section? -----------------------------------------------------------------

Anyone have one of these yet? http://www.voxilla.com/shop/index.php?action=item&id=38 -Dan

Anyone here have experience with these devices? They would ppear to be an affordable alternative to multiple X100Ps. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves@mstvp.com "Kick at the darkness 'till it bleeds daylight" - Bruce

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

All of a sudden my list traffic appears to have dropped to a few messages/day the past few days. I anyone else seeing this as well? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves@mstvp.com "...I believe in love, its all we've

OK. So it would appear that my quest for FXO adapters unconvers more, and certainly more mature, H.323 based devices...not so many SIP devices. What would be the benefits of SIP over H.323 for a small office * server? All I need to do is bring 4 POTS lines into * with Caller ID, make outgoing local calls reliably without undo echo. FWIW, my * server is Fedora Core 1 on AMD XP2500 with 512 MB

Hi All, Is this working for any of you? We're supposed to be able to dial FWD accounts through IAXTel by dialing 170099+FWD number. I don't ever seem to be able to do this even though I can dial 800 numbers through IAXTel. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc.

Hi All, I'm using DISA on my * server to avoid overseas toll charges when making calls to Western Europe from my cell phone. I have DISA working with a DID from a VoicePulse Connect account. The outgoing call to Europe is also made via Voicepulse Connect. I see that the IAX media path is bridging the inbound call to the outbound call so that the media stream entirely bypasses my server once

Hello All, I have a small * server in my home office with several IP phones. The system is not fully in service yet as I'm still hunting for a cost effective FXO adapter that I can rely upon for my two primary PSTNs. That said, I'd like to move it into service for another application...which brings up a question. I'd really like to stop making international calls from my cell phone