similar to: Good source for Polycom IP Phones

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, 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

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, 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

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 getting ready to install Asterisk and I was looking into the Polycom IP600 phones. I spoke with Polycom sales to verify the multiple line appearance and they said it would work. More specifically, if lines 1-3 all contain the same SIP registration info, the Polycom will only send out 1 SIP registration to the server and then handle the calls ringing on multiple lines. I was wondering if

I encountered the following problem, can anyone help me? when i use "nmblookup -B 192.168.1.255 redmaple" on my linux box named redmaple, something like "(tdb(null)): could not open file /usr/local/samba/var/locks/unexpected.tdb. No such file or directory" is output, and the name lookup failes. my samba is 2.2.3a, IP address of redmaple is 192.168.1.5 rgds aloha

Hi We've setup an Asterisk PBX recently and I encountered the following problem: When [mac address]-registration.cfg file includes the FQDN of the Asterisk PBX for the Polycom SoundPoint 501 phones it will not (even try to) register with the Asterisk PBX unless the DNS (it asks) successfully resolves the name: _sip._udp.[Asterisk FQDN]. Did this happen to anyone else? PS - The

Dear Group, Does any one have experience using SoundPoint(r) IP 300? I have one call center on Snom 200's I'm adding a second and was looking at the SoundPoint, but needed some input. Thanks Shad Mortazavi --------------------------------------------------- Nexus Technical Manager n|m Nexus Management Inc Sydney -------------- next part -------------- An HTML attachment was

Hi All, Did anyone came across a 411 XML service I can feed to the "service" button with XML? Some other feed I can manipulate to XML query? Assaf Benharoosh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041004/dbf552ac/attachment.htm

Has anyone been able to get the minibrowser on the Polycom SoundPoint IP 500/600 phones working? If so could you share the relevant sections of your config with me?

Problem: Polycom SoundPoint IP phones (running SIP) ceases to send DTMF that Asterisk can detect and use. It worked in 1.0.5, but has not worked since. This has been verified on SoundPoint IP 300's and SoundPoint IP 600's. Workaround: It used to be that for DTMF to work, I had to set the mode in sip.conf to "inband". Without making any configuration changes on the

Hi, I'm using Asterisk@home and am having trouble using the conference bridge that comes built in. We're using Polycom phones. When we transfer the first person into the conference room (e.g. 8101) , they get into the room fine. When we try to transfer a second person into the conference room, they get dropped as soon as we finish the transfer. This is using Polycom SoundPoint 301