similar to: Example Polycom XML applications, anyone?

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

Interesting notes on the 79xx series. The 7920 is the wireless phone; not mentioned here. For a more complete guide to Cisco's phones, see: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheets_list.html The 7902 is the "very inexpensive" Cisco phone, and it looks like it will be SCCP (Skinny) only. Twiddling my thumbs here waiting for the chan_sccp to

Bug in the register= code; see details below from the developer of "ser" (SIP Express Router) Apparently, ACKs don't need to be sent on OK's to REGISTERs. Plus, malformed data somewhere... no details on that, though. JT >Date: Sun, 23 Feb 2003 23:54:07 +0100 >To: John Todd <jtodd at loligo.com> >From: Jiri Kuthan <jiri at iptel.org> >Subject: Re:

Judging from log analysis, there has been in the past some interest in the Asterisk example files that I have made available from my experimental platform (home office) so I've put my newest versions up on my site. Feel free to dig through these and cut pieces up for your own use, or just marvel at the bad logic flows. :) http://www.loligo.com/asterisk/current/ JT

We are still using 1.4 and were going to start testing with 1.6.0, but then 1.6.1 was released and now 1.6.2 is already in beta 2. That seems like a lot of independent releases to maintain. I read about all the regressions ans hurried dot releases, makes us nervous. How is everyone doing their testing? -Matt -------------- next part -------------- An HTML attachment was scrubbed... URL:

I really should be doing something better on this beautiful weekend, but I'm trying to save myself some time for later projects by documenting some things that have been particularly troublesome in the past. That being said... I've written up a configuration guide for the Cisco ATA-186, which describes some of the features that are possible to set in the ATA and specifically what

I seem to be having a conceptual problem with the "While" and "EndWhile" applications. It seems that on the first cycle, even if the result of the "While" is false that the enclosed applications will get run. Is this expected? It seems to be counter-intuitive, but I don't know what the intent of the While routines is. I could of course put a

Hello, Good news on the Polycom front for those that are interested. It looks like we may get a dedicated Engineer for Polycom/Asterisk!!! Happy Day! Here's the message I got tonight: Matt: I heard back from our VP of Engineering- she is prepared to have an individual dedicated to working on the Digium- Asterisk project. Can we discuss again Friday or mid next week? Scott Willard

Sipura does include STUN as an option. It has for quite some time. We are using it with all of our Sipuras behind NAT'd gateways and it works great! Try upgrading to the latest Sipura firmware rev. Darren Nay > -----Original Message----- > From: John Todd [mailto:jtodd@loligo.com] > Sent: Saturday, May 22, 2004 1:57 PM > To: asterisk-users@lists.digium.com > Subject:

Hello all - Second-to-last notice to collect RSVP's for interested parties for the Asterisk Birds of a Feather dinner/beer meeting. There are five people signed up for this at the moment. Looks like it's going to be the 23rd of September, since that fits well with the VON conference schedule. Please email me directly for details. JT >Date: Thu, 21 Aug 2003 01:52:52 -0700

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?

I've been involved with getting better data for running Asterisk on the Amazon EC2 cloud computing system. Here are some calculations I've made on costs based on current published prices on Amazon's system. Feel free to tell me that I'm wrong with these calculations - but be specific if you find any problems, as I suspect others may glom onto these figures as gospel and

OK, here's a configuration challenge: I want to have certain line appearances able to be "interrupted" by various other line apperances elsewhere in the office. This is harder to describe than it is to demonstrate, so I'll do that: Let's assume I have Cisco 7960's on all desks. 1) Call comes from inbound line X destined on extension 1234 2) Phones A, B, C all

Hi Polycom experts, I'm having a problem getting changes to the Polycom IP 601's (mac)-directory.xml file to update the button list on the phone. If the phone is newly provisioned (i.e. if I "Format File System" on the phone) then the new list will show up on the buttons, but of course this is pretty drastic way to do it. - Environment: Asterisk test setup with 7 phones,

Hello all Sorry for the repost, but I believe this is essential for Aastra phones to use directed pickup with BLF in 1.2 (and it's still experimental in 1.4 I think). pickup-mgernoth-2006-07-28.patch.txt If anyone knows otherwise than that this is still required in latest 1.2 or 1.4 please reply with success/failure stories. Any info would be appreciated. We can get directed BLF pickup

John, I spoke with Level(3) last week regarding SIP termination. They quoted $0.01/minute, with an 11 Million Minute / Month minimum. Ugh! -dg -------------------------------------------------------------- Darnell Gadberry President binaryMedia darnell AT binmedia DOT com ------------ Date: Fri, 19 Dec 2003 21:12:22 -0500 To: asterisk-users@lists.digium.com From: John Todd

> >> Jason Brown wrote: > >> | Anyone have experiece with polycom phones? > >> | > >> | I am experiencing a really weird problem. In an office > where I have > >> | the following extensions: > >> | On the Polycom phones, when I want to dial from extension > >> 100 to any > >> | extension 120 or above, or dial out, it

I am exploring the contacts directory in Polycom and I am wondering if a big 000000000000-directory.xml on the boot server will eat up the memory and crash the Polycom phone once downloaded onto the phone. The asterisk directory extension is good but because users cannot see the names I thought to find another alternative. Does anyone have any experience?

I bought a couple Polycom Soundpoint 300's, and have them working nicely with SIP... but I'd like to be able to do automatic config via FTP, but it requires some XML config files. The docs discuss them in detail, but I can't seem to d/l them from Polycom. [No, it doesn't appear to be on the CD that comes with the phone.] I've created an account at Polycom's tech support

Anyone have experiece with polycom phones? I am experiencing a really weird problem. In an office where I have the following extensions: 100 101 102 103 104 110 111 120 130 140 141 150 200 On the Polycom phones, when I want to dial from extension 100 to any extension 120 or above, or dial out, it dials just fine. If I want to dial from extension 100 to extension 101,or 102 or 103 or 104,