similar to: Fw: X100P questions

Aloha, Has anyone written an XML application for a Ciso 7960 phone running SIP? I can't find any examples anywhere! Anyone know of any resources for this? I have read it can render XML & can get input from the keypad & softkeys. Aloha, Matt

Aloha, Does anyone have a good source for Polycom SoundPoint? IP 600/500/300 phones? Everyone sells Cisco 79XX. Aloha, Matt

Hello! I've been following this list for several weeks now and would like to purchase some hardware for a VOIP/ voice-mail solution. This card appears strikingly similar to a modem. Is it? Is there a product to bring in more than one POTS line short of a full T1? It just seems silly that the technology hasn't advanced any further than to have a single line per card. Any thought

Skipped content of type multipart/alternative-------------- next part -------------- [root@LINUXVM root]# asterisk -vvvc == Parsing '/etc/asterisk/asterisk.conf': Found Asterisk 0.4.0, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer <markster@linux-support.net> ========================================================================= == Parsing

Aloha, Are there any hard phones that support ILBC? I see lots of soft phones but not Cisco, Polycom etc. Aloha, Matt

Aloha, I have an application to build an eight conference bridge for a Norstar. I would have a converter to turn the digital ports into analog with disconnect supervision. On the digium site I only saw a single port FXO card. I would like to use a higher density FXO card like a D/41 or newer from Dialogic. I noticed in the handbook 2 (draft 3/30/03) that there is a price to use dialogic cards

Aloha, Did anyone ever get the formating manual for the XML brwoser on the Polycom IP600? Does anyone have a sample? Aloha, Matt

Aloha, Anyone know how to disable call forward on a Polycom Phone. Calls being accidentilly being forwarded somewhere is the #1 trouble that we have to respond to. The real issue is the 'end call' button becomes 'forward' when the call ends....therefore the user thinks they are pressing 'end call' but the call ends just before they press the button so they end up

Aloha, We are looking to roll a solution that will have the following network layout: ISDN-PRI <--> Asterisk <--> T.38 <--> ATA <--> Fax Does version 1.8 with the Digium fax driver have this capability? I like 1.8 because it is a long term support version. What ATA's are people using? Any working solutions would be great! Aloha, Matt

Hiya, here is a patch to fix that: [root@mailmx2 channels]# cvs diff chan_vpb.c Index: chan_vpb.c =================================================================== RCS file: /usr/cvsroot/asterisk/channels/chan_vpb.c,v retrieving revision 1.9 diff -r1.9 chan_vpb.c 100,102c100,102 < static VPB_TONE Dialtone = {440, 440, 440, 0, 0, 0, 5000, 0 }; < static VPB_TONE Busytone = {440,

Hi we've got Asterisk CVS-HEAD 18-Aug-04 (modified by Voicetronix as available on their site for use with the vpb driver) and an OpenLine4 (4xFXO). The same server also has two X100P. Calls on the Voicetronix card drop instantly when the called party picks up. The vpb driver reports that it detected a hangup (loop drop) yet there is no hangup when connecting the X100Ps or analog phones to

Why hasn't someone found 50 people who sound alike, put them in sound studios and record the 10,000 most commonly used words. You would all differnent forms of the 1,000 most words, i.e. leading, trailing, question etc. You can synthesize the other 0.05% when you run into them. With hard drives so big, processors so fast and EXT3 that can handle 30,000+ files in a single directory that

I am running CVS-09/11/03-14:03 on Redhat 9.0 Trying to get call waiting / call waiting callerid working. The setup is: X100P asterisk -> ATA 186 MGCP --> analog phone. What changes to I need to make to my mgcp.conf and extensions.conf file to allow answering of Call waiting calls? And how do you answer call waiting calls with the system. I have usecallerid=yes, hidecallerid=no,

>-----Original Message----- >From: Ben Greear [mailto:greearb@candelatech.com] >Sent: Monday, October 06, 2003 10:24 PM >To: Bart De Schuymer >Cc: netdev@oss.sgi.com; vlan@wanfear.com; bridge >Subject: [VLAN] Re: [PATCH/RFC] Let {ip,arp}tables "see" bridged VLAN >tagged{I,AR}P packets > > >Bart De Schuymer wrote: >> - add some code in

Aloha, Does anyone have any screen shots of the Asterisk admin GUI? I couldn't find any links in the archives or the asterisk web site. -Matt

Aloha, Does anyone know how to have a hot keypad on a Cisco 7960? It allows you to dial on-hook without press the SPEAKER button. Very handy once you get used to it! -Matt

Has anyone been succesful pushing a VLAN setting to a Polycom phone via DHCP? I can push the boot server via option 66 but that is about it. I have set it for 'fixed' and tried many different option numbers with a couple differnet DHCP servers. SIP firmware 1.3.4 or 1.4.1 doesn't make a difference. Aloha, Matt

>From the Wiki: 'There are aready a few bugs in 1.5.2 but more fixes some good new features' Anyone know where the bugs are being listed? I am working through a few issues: 1. When rebooting, the phone will pause for exactly 180 seconds with the screen reading 'updating initial configuration'. I know it is 180 seconds because when I tail the FTP server log the entries are

Aloha, I have a Cisco Gateway the is functioning as my SIP Gateway to the PSTN. I have an Asterisk box, RC2 with a for port FXS card providing dialtone for a Norstar Key System. I have it working so when you press a line key on the Norstar you get dial tone from the Asterisk box. The user has to dial '9' then they can dial there number which is sent to the Cisco GW via SIP and the call

Ron, VoiceTronix worked with me last week and backported the new chan_vpb that's in the current HEAD CVS tree to the 1.0 stable branch. So if you are using 0.7.2 or 0.9.0 you can go to the VoiceTronix website now (note that this is very new) and download an Asterisk package. When you download it there is an new chan_vpb.c that needs to be put in your asterisk source code tree. Also there