similar to: asterisk solution.

Anyone know what happened to netxusa? Seemed like they dropped off the web overnight. -Matt

Hello, I wonder if anybody is using PJSIP realtime in production environment? I've started to play with it and encountered many problems. Here's my config: sorcery.conf: [res_pjsip] endpoint=realtime,ps_endpoints extconfig.conf: [settings] ps_endpoints => pgsql,users,pjsip_endpoints_v pjsip_endpoints_v is postgresql view. 1. The biggest problem: if I have small number of endpoints

Hi, I would like * to treat a SIP extension as a normal extension, when it comes to the busy functionality. In other words, if someone tries to call the SIP phone and there is already an ongoing conversation, the new caller should get a busy message/tone Is there any parameter that I can set? Is this something that should be configured at my softphone? Best, PHM

I can give you a suggestion that worked for me for the after reboot no printer access. Make sure your printing daemon starts before your smb starts. It seems that samba won't see an lpq/lpstat if there isn't one. Funny how that works!! Hope this helps, and if I'm way off base, someone let me know... Wade -----Original Message----- From: epn.neustadt@t-online.de

A customer contacted us today concerning getting a VoIP to PSTN system with a few IP Phones setup. Asterisk should fit his needs. It is not a big job, but I think that this customer is going to need onsite work. Please contact me off list if you are an interested reseller in the Washington, DC area. Sean _______________________________________________ Sean Robertson NETXUSA p. 800-289-6389

Dear All, I'm integrating avaya commuication manager difinity ver 1.0 with asterisk using B2B E1. following are the details of my H/W, zaptel configs and software installed. Digium TE110p asterisk 1.2.19 cent OS 4.4 zaptel 1.2.18 libpri 1.2.4 etc/zaptel.conf span=1,0,0,cas,hdb3 bchan=1-15,17-31 dchan=16 when i ztcfg -vvv im having this error message and the E1 is not getting up. "cas

Hi list, how does one get a SNOM 200 MWI to work with * ?? When I press the MWI button it doesn't connect with voice mail on my * box. thanks

Hi! Has anyone on the board successfully installed a Mediatrix 1102 or 1104 as a SIP peer on Asterisk? I'm trying to configure different user accounts on each FXS port, but I'm having authentication problems; Asterisk is saying the client is not authorized. Interestingly enough, I can dial a "9" and make a local call through the Mediatrix. Thanks! chris --------------

Hello all, Does anyone has experience of deploying Asterisk based VoIP solution in a universitywide campus. We are at present investigating various Soft PBX for this purpose from different vendors Digium,Snom, Pingtel... We are looking at serving more than 5000 clients and we want to be very sure before taking any final decision. I would be glad to hear from members who are aware of

I would like to prevent * from dialing 900 and 976 numbers. I setup the following settings in extensions.conf. But this does not seem to work! I don't know what I am doing wrong please help! exten => 1900XXXXXXX,1,Congestion exten => XXX976XXXX,1,Congestion exten => XXX976XXXX,1,Congestion exten => 1XXX976XXXX,1,Congestion exten => 91900XXXXXXX,1,Congestion exten =>

Not sure if this is the best place but does any one have any used IAXy's they are interested in selling? I am looking to pick one up cheap for a proof of concept before going all out on them. Also does any one have any real life practical experience with how well (or not so well) that these devices have worked for them? you can reply to me off list at asterisk@matraex.com Thanks Michael

We just did an upgrade on our Asterisk to the CVS version and our Mediatrix 1102s stopped working correctly. Our Asterisk is connected to the PSTN with a PRI. Calls from the PSTN to the Mediatrix 1102 work fine. The issue is calling out to the PSTN from the 1102. Asterisk looks like it process the call just fine except there is no talk path. Get this, though: If you flash hook and then

Hi all, I've got an old avaya partner acs <7.0 system here. I'd like to add a simple voip bridge so I can hook up our remote offices. From my research, it would seem the pre-7.0 series doesn't have a t1 port, so if I wanted to do this, I would have to feed the avaya system fxs ports from the asterisk box. Does that sound about right? Has anybody ever done this? Does

I have a Mediatrix 1204 FXO gateway setup for SIP. I would like to know if anyone has gotten this item to work with Asterisk. I need to get a 2 or 4 port FX0 gateway working with asterisk. The Idea is the following. PBX at lets say any Hotel(Analog lines FXS) - FX0 Gateway(1204) -- {Internet} -- Asterisk - local IVR system. (IVR is not at present running Asterisk old dialogic system has FX0

Has anyone successfully run asterisk with a VIA processor ? I have tried unsucessfully, do I have to run make with any specific switches ?

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

Here is a SIP phone I haven't seen before. Does anyone have any experience with this one? -----Original Message----- From: George Richardson [mailto:georger@netxusa.com] Sent: Wednesday, April 02, 2003 4:56 PM To: clay@ctitec.com Subject: ipDialog Ethernet SIP Phone $199 pad <http://us.st1.yimg.com/store1.yimg.com/Img/trans_1x1.gif>

Hi I want to set up * as a conference bridge. I would like to be able to conference is SIP calls (up to 12) I am looking through all available documentation for * to get info on how it is done. No luck so far. Can somebody direct me to the info in this subject? Thank you Serge _________________________________________________________________ Protect your PC - get McAfee.com VirusScan Online

Anyone have an H.323 trunk tied between their Avaya and Asterisk box that works? I am having some issues trying to get the two systems to connect. I am using the ooh323 channel to try to make the connection between the two system. I have all my configs if anyone would like to look over them. If I do a trace on Avaya I get a denial event 1191: Network Failure. Thanks! -------------- next part

Anyone have a good source for BT-101 phones? I had a lead on some, but they've not materialized. I'm also interested in the ATA-286 (HandyTone) units as well. This is for my personal Asterisk/INOC-DBA setup, that has yet to materialize heh. --- Tom Sparks