similar to: Cisco IP Phone- disjoin conference

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

To whom it may concern, When dialing out an 800 number (888,866,877) through VoicePulse IAX you'll get a choppy sound. This is not due to a problem on your Asterisk or your line- the bad sound effect occurs in VoicePulse. (just spend lots of time finding that out) Assaf Benharoosh MCP, MCSA, MCSE ab@franticllc.com <mailto:ab@franticllc.com> Frantic, LLC. -------------- next part

Hi all, Does anyone know how to get the channel ID on the other side of the call? For example: When SIP/50 calls SIP/21, and the call is answered by SIP/21 I get: SIP/21-6735 answered SIP/50-b456 ${CHANNEL} will show me SIP/50-b456. Is there a parameter or a workaround to get the SIP/21-6735 part? Thanks. Assaf Benharoosh -------------- next part -------------- An HTML attachment was

For some reason I can't get the Dial command 'C' flag to work. The calls are recorded in the CDR with the 'C' on. Does anyone have an idea? extensions.conf: exten => 114,1,Dial(SIP/114,,C) It shows in the lastapp: cdr: | 2004-10-05 13:16:02 | "112" | | 114 | intern | SIP/112-3fb6 | SIP/114-0e7a | Dial | SIP/114||c | 6 |

Hi, Has anyone used polycom phone IP500/600's conference feature? I ran into some issues. Polycom is not behind NAT. Polycom calls pstn number A, (gets two way voice), presses conference, (A is put on hold), calls pstn number B, (gets two way voice), presses conference again, suppose to have all three parties in the call, polycom and B have two way voice, but polycom doesn't have two

Has anyone tried using BroadVoice for VSP? I have Asterisk configured for a home office & I've been trying to decide which VoIP provider to go with for a little while now. I had heard you could get sub $.01 calls but I have not found that to be true yet (not saying it's not possible, I just haven't found it!). Also I'm not sure if BV will support multiple lines. Any

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

Hi Everyone - I have webvmail up and running, I can see the messages, forward them, pretty much everything but listen to them. Here is what I see in my logs: 192.168.0.1 - - [25/Feb/2005:08:15:40 -0800] "GET /vmail/vmail.cgi?action=audio&folder=INBOX&mailbox=2377&context=default &password=000012&msgid=0000&format=gsm&dontcasheme=4624.gsm HTTP/1.1" 200 9438

Shawn, That's a complete load of manure. I have an office full of 7960's, they work great with asterisk with the SIP images loaded. I'm about to pick up a lot of 7912's (simple one line phones, same as the 7905 but it has a built in switch). These phones have also been confirmed to work with Asterisk. I would recommend not going directly to cisco, and just find a reseller who

Hello, Anyone else having problems with the Cisco 7940/7960 (5.3 firmware) and the latest CVS build, placing conference calls from the phone? I've noticed the party on the Cisco phone's side will sound very garbled, and delayed by several seconds. I haven't begun troubleshooting yet, though I'm able to reproduce this easily... Thanks in advance, -a

I am guessing you are running without reinvite's, I'm running with reinvite's with latest CVS release and 79x0 phones without any issues with conferencing... > -----Original Message----- > From: Adam Rothschild [mailto:asr@latency.net] > Sent: 08 October 2003 15:49 > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Cisco 7940/7960 phone and >

Hi All, Does anyone have any experience getting a 7936 to work with Asterisk? Do you need to use SCCP or is there a SIP image for the phone? I have a few 7960G's and they are working with SIP, just curious if the config of the conference phone is the same and if anybody has any good setup links. Thanks! NB -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have one Cisco IP Conference 7935. Somebudy have any idea how I can config this phone to work woth "*". My "*" server is now working with GrandStream Phone and X-Lite SoftPhone, I need to add this Cisco 7935 but I don't know how I can convert to SIP. Thanks, Pedro Mansilla Consulting Integrating Services pmansilla@cis-international.com T:

Hi, I have one Cisco IP Conference 7935. I'm trying to config the SKINNY Protocol. I config the skinny.conf file same like sample for Cisco 7910. When somebody call me my phone ring and answer the call but I can't hear anything, But the other people is hearing me very good. When I try to call somebody the * show me an error : RECEIVED UNKNOWN

Hi All, I need some help w/supervised transfer and conference w/a 7940 phone. When I do a blind transfer the calls go through great, but when I do supervised transfer the 7940 tells me "Transfer Denied". When I do a conference call I hit the "conf" key and then dial the next extension. The new call connects and I hit "conf" again but the calls do not get bridged.

Hi All, My current 7940 phones use P0S3-06-3-00. I'd like to upgrade them so they're not massively out of date. I found a page at http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx that gives some info, and using the cisco links there have tried to upgrade. According to the procedures, I should be able to upgrade, but once the phones loaded and reboots it says it downgrades

Hi, Has anyone here ever used a Polycom IP 4000 Soundstation SIP Conference Phone with asterisk? If so, how well does it work and how does it sound?

Dear all I have configure Asterisk 1.4 with sccp and configure Cisco phone 7975G model with Asterisk and it is working fine but i have one problem when i going to confance like when i call to someone and put it on hold and want to take in another New call to new person and want to take both live call in confrance how to do it in asterisk is there any configuration XML

Hi there. I''m looking for a solution/gem/tool/tutorial to include the functionality to start a call on a Cisco IP-Phone 7945. I''ve basically no knowledge about how to get access to a Cisco IP-Phone through the network. I would be happy about any information you guys have. Maybe there is a CLI tool to make calls with an IP-Phone? Atm, I''ve no idea where to start so any

Hi, >[...] >> > There should also be a digitally signed version of that file >> > (cmterm_7920.*.sbn), which the phone probably requires. >> >> nope. no sbn. according to my cisco source the file is not signed. > >Funny, that would be the first phone with unsigned firmware. >But I'll double-check after the next firmware update. At least for my other