search for: tryc

>I can not even get IP anymore from my DHCP Hate to ask the obvious, but is the DHCP server on the same VLAN?

...y between PIKA high-density analog boards (Daytona MM) and Asterisk PBX software. ?The Asterisk development community can now benefit from advanced features for fax and echo cancellation in high density analog applications, made possible by PIKA?s DSP processing power on the board,? stated Wojciech Tryc, Enterprise VoIP architect at PIKA Technologies. ?Because of the native bridging for TDM calls, latency is drastically reduced?nearly eliminated? in this implementation. The solution is very reliable, as we have witnessed not only in the lab, but in live customer environments.? Stroudwater Contac...

I'm trying to switch from one call to another incoming call from PSTN. When I'm getting a "beep" I press flash but instead of swithing to the second call, I'm getting a dial tone. even if I press *0, I cannot connect to the second call. Anybody had this problem? Tx, Bogdan

I am forklifting a Merridian option 51c with 112 Nortel Digital Handsets and 400 analog units. For the analog units I have quotes for 9 ADIT 600 48 port fxs units and 17 Rhino 24 port FXS channel banks. I have used neither. Which is the best choice? The price difference is not that great. I am looking at Citelinks 24 port Handset Gateway for the Nortel Digital units. (Any other suggestions

Has anyone been able to get the IP501 to discover the FTP server IP address (via dhcp or dns) and download 100% of the config from a provisioning server? We are still having to touch each unit to enter the ftp server address and password, as well as set many of the options that will not take from the config file. Have a sample config file you are willing to share? What is required in

...l an extension for wakeups The caller is asked to set a time and the number of days for which they want it set. The system then calls at those times, and every ten minutes until it is answered. ------------------------------ Message: 3 Date: Thu, 2 Mar 2006 13:10:53 -0500 From: "Wojciech Tryc" <wojtek@VoIPMan.ORG> Subject: Re: [Asterisk-Users] OT - Cisco IP Phone and PC in diferent VLANs(with 802.1x) To: <joao.pereira@fccn.pt>, "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <002501c63e24$a5f94...

...s the vlan aware driver ("client") it will be able to read tagged packet for the vlan which PC has been configured to use. Nothing to be done on the switch. W -----Original Message----- From: Joao Pereira [mailto:joao.pereira@fccn.pt] Sent: Thursday, March 02, 2006 1:15 PM To: Wojciech Tryc; asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] OT - Cisco IP Phone and PC in diferent VLANs(with802.1x) Ok, but the PC has an 802.1x client that configures the VLAN when he authenticates. Is this going to pass through the phone? And will the switch accept it? Thanks Joao Pereira...

...1204 led turn on and they started to interchange packets, im newbie with asterisk i have been trying another sip server with mediatrix that work so well, but i dont know how to set it up? could u send me all the configuration i need step by step? ----- Original Message ----- From: "Wojciech Tryc" <wojtek@tryc.ca> Date: Mon, 7 Jun 2004 21:59:43 -0400 To: <asterisk-users@lists.digium.com> Subject: Re: [Asterisk-Users] Mediatrix 1204 > The Mediatrix box will not registered with * as the user name and password > for sip are not yet implemented in their firmware. > Al...

..., could you elaborate? Thanks, -- -- -- Christopher T. Aloi USA Datanet - Technical Support Engineer 318 South Clinton Street Syracuse, NY 13202 C: (315) 569 4033 O: (315) 579 7074 E: caloi@usadatanet.com <mailto:caloi@usadatanet.com> -- -- -- _____ From: Wojciech Tryc [mailto:Wojciech.Tryc@pikatech.com] Sent: Friday, February 17, 2006 10:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] A unique 'click to call' project - Could usesomeadvice Why don't you use Local and router functionality to find a ro...

Hi, At first I would like to express how much I like Asterisk. Amazing product. I compiled Asterisk with mySQL support for CDR and Voicemail. Everything seems to be fine, I can see that Asterisk connects to mysql and logs CDRs. I can also see that the VOicemail app is also logged in, however I can not access any mailboxes. Similar messages to others, app_voicemail.c:3011 vm_execmain: Couldn't

I have successfully implemented 1204 in semi production environment. Just want to share that it works very well, through the firewall (NATed). Unfortunately, it can not register with the server (and authenticate) but otherwise everything is fine. The audio quality is very good. Regards, Wojtek -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am trying to get asterisk to email me my voicemail as attachments. What am I missing? Where do I tell it to go for SMTP services? Voicemail.conf: ; ; Voicemail Configuration ; [general] format=wav49|gsm|wav serveremail=pbx.agtcorp.local attach=yes maxmessage=180 skipms=3000 maxsilence=10 silencethreshold=128 maxlogins=3 append=yes [default] 100 => 1234,Sean Garland,sean@siskiyoutech.com

I downloaded Cepstral to my Asterisk Box. I did the install and let it install to /opt/swift. I brought down a new CVS-HEAD as of today 10/1. I added APPS+=app_cepstral.so into the Makefile in /usr/src/asterisk/apps/Makefile Like: # Obsolete things... # #APPS+=app_sql_postgres.so #APPS+=app_sql_odbc.so APPS+=app_cepstral.so # I did this piece but wasn't sure exactly what part of the

All, I get this whenever trying to dial to a peer when the peer registered to another server. I'm basically trying to use realtime to check for the peer and dial it. Mar 24 09:16:47 VERBOSE[4527]: -- Executing Dial("SIP/brak-f69f", "IAX2/brak-test/107") in new stack Mar 24 09:16:47 DEBUG[4527]: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name =

> Hi, > I am not sure if this is the right forum to ask, but I am having small > problem with CDR (Call Detail Records) by Areski.Net. > Everything is fine with the exception of the Stat Graph. It doesn't display > complaining about missing GD2 libs. I have GD2 installed and PHP is also > compiled with it... > Lots of things to do so I can not dedicate enough time to

Hi All, My friend is looking to for a fulltimer to work on various VoIP gateways (mostly Cisco) and few Asterisk servers. Some development skills and knowledge of Asterisk's API would be an asset. At least familiarity with Asterisk's AGI and Perl/C/PHP would help. He is located in Kiev, so obviously Kiev residents are preffered. Remote work to be discussed. Again, this is not a contract,

Why don't you use Local and router functionality to find a route to PSTN based agents? W _____ From: Aloi, Christopher [mailto:caloi@usadatanet.com] Sent: Friday, February 17, 2006 10:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] A unique 'click to call' project - Could use someadvice Hello List, I work for an IP

The only advantage is when you travel. Last year I took my wifi sip phone to Astricon in Madrid and everything worked as expected. I am just packing it and heading for Paris... Wojtek -----Original Message----- From: Mojo with Horan & Company, LLC [mailto:mojo@horanappraisals.com] Sent: Tuesday, June 20, 2006 1:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: