similar to: Any updates on the Cisco 7920 and SIP?

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

John Todd wrote.... > Cisco has an 802.11 phone called the 7920, which is apparently > shipping now. It is very expensive (>$550 USD) and only runs SCCP at > the moment, which is Cisco's proprietary VoIP protocol. However, if > it falls in line with some of Cisco's other high-end VoIP equipment, > that means it should have a trailing-edge SIP image running by

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 have several phones running Cisco's 6.0 SIP software release at this time. Two of the phones have not shown any abnormal behaviors, but one of them has an unsettling propensity to lock up after several hours, where the softkey labels disappear and the phone stops registering, requiring the standard *-6-settings reboot sequence. Otherwise, the phone seems to work OK except for a slight

Anybody on the list who implemented Cisco ATA + * + Cisco 2600? I cannot get my calls from ATA to terminate to the Cisco gateway via *. I am not sure if it is my hardware problem. I'm getting the following "codec negotiation problem" from Cisco. 23:39:08: Unexpected VoIPCodec Type :g729br8 23:39:08: Unexpected VoIPCodec Type :gsmefr I appreciate any help I can get. Thanks.

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

This is almost certainly not an Asterisk-specific posting, but due to my inability to find a VoIP-focused Cisco list, I'll post here in the hopes of finding a more diverse user community. I am using a Cisco 7960 (version 6.0 SIP firmware) with Asterisk, and have been experiencing situations where the phone locks up. "Locks up" means that the bottom part of the screen

-----Original Message----- From: Adams, Gavin Sent: Monday, August 04, 2003 6:10 PM To: 'asterisk-users@lists.digium.com' Subject: Cisco 7960, SIP, NAT, Voicemal Hey all, I've got a couple 79xx phones working peer-to-peer and am now trying to work on the voice mail. In extensions.conf: [ATL] exten => 4001,1,Dial(SIP/gadams)|10 exten => 4001,2,Voicemail,u4001 exten =>

hi! i had some good news regarding the cisco 7920 and the internetworking with asterisk (and possibly SIP ?). Status: chan_sccp.so not coredumping anymore :-) Phone contantly in reboot loop [see below] :-( Reboot Loop means: ------------------ Phone auth's with AP Phone gets IP from DHCP & TFTP Server Phone loads OS7920.TXT Phone loads SEP<macaddr>.CNF.XML Phone loads

Has anyone managed to get their hands on a 6.0 image for their 7960's yet? Or is it still in beta? Rumor (official rumor, from Cisco) is that it supports paging and intercom. I'm anxious to start working with those features, if they've been implemented sanely. What would be just as nice would be NOTIFY messages for pushing XML URL's to the phones, but sadly that feature

This is not specifically Asterisk-related, but I think that sometimes a "pre-emptive" clue is a good thing. Viking Electronics (http://www.vikingelectronics.com/) has some neat widgets that attach to phone lines, which I'm sure many of the people on this list would find at least somewhat useful in conjunction with their Asterisk systems. Many of the widgets can be replaced with

http://www.loligo.com/asterisk/misc/Presentations/Asterisk-features-20040106.xls I had been asked a while ago to put together a short Excel spreadsheet listing many of the "common" features of Asterisk as compared to a typical PBX. Many PBX vendors supply an exhaustive list of their features, and I figured I'd take as many of the unique features as others had offered, and put

does anyone have a sip working through a cisco pix firewall? i can get the phone to register and the call to be negotiated, but as soon as the call is answered there is no sound and the call ends immediately. im sure this is due to the RTP negotiation being rejected by the pix. any helpful ways around this? right now my only solution is to put a small box outside the firewall and IAX the

I see that there's now a 2.16.1 upgrade path for Cisco ATA-186 devices, dated (variously) July 11 or July 14 2003. Here are some interesting bugs that claim to be fixed. Most notable is CSCeb17953, at least from my perspective, as I've hit this bug before. CSCea42480 The Cisco ATA ignores the Require:100rel header and processes call. CSCea69889 The Cisco ATA builds a 302 Moved

Hi, There is any simple way to have external access to the Asterisk database? I want to be able to edit a specific family only, which is a phonebook. Thanks, Dan

I read about this intercom stuff on page 62 & 63 of the book "Developing Cisco IP Phone Services" isbn 1-58705-060-9. Primary calls take place on streaming channel 0. When streaming channel 0 is not in use, streaming channel 1 can be used for asynchronously streaming (in and out) stuff like voicemail, email, and, yep the one we want, intercom. Page 87-88 of the book talks about

http://www.boingboing.net/2006/06/05/play_zork_by_phone.html Let me preface this idea with one comment: I don't have the time to do this - I don't even have time to eat these days. But someone out there has the cycles to do this... and it would be very cool. OK, so now Zork is attached to Asterisk, but using the less-than-clear Festival engine. There are beta tests of the LumenVox

I'm sitting in the Emerging Telephony Conference, so this seems a particularly apt place to pre-announce this... I've wanted to be able to gateway calls between Skype and Asterisk for a while, which of course would require some type of protocol converter (IAX or SIP to Skype, probably.) This of course is directly not in Skype's interest, since they would like to keep the network

This is Off-Topic for Asterisk, but I wanted to get some feedback on headsets for Cisco 7940/7960 phones. We have about 10-20 people who wants/needs a headset for their phone & was hoping to collect some real-world input. Thanks!!

Hi I read through the archives but could not find much reference to * using SIP on TCP instead of UDP for signalling. Can * be configured and if so how. My service provider will only accept SIP signalling on TCP. Thanks Master -------------- next part -------------- An HTML attachment was scrubbed... URL: