similar to: Polycom Buddy Feature

I'm still trying to work this out. I've got this in my sip.conf [1003polycom] type=peer secret=abc123 host=dynamic defaultip=192.168.1.215 context=default mailbox=1003 subscribecontext=phonestatus [1004polycom] type=peer secret=abc123 host=dynamic defaultip=192.168.1.214 context=default mailbox=1004 subscribecontext=phonestatus And this in my extensions.conf [phonestatus] exten =>

Hi, I have was testing some of the different ring types with my polycom 500, and the ALERT_INFO settings. Now when my phone receives a call it won't ring. All the other phones ring fine, and my phone wasn't the only one I rebooted with the changed sip.conf and impd.conf. I have reverted back to a standard sip.conf and impd.conf and I still can not get my phone to ring for any incoming

Does anyone know if SIP will/is support on handheld PCs such as the iPaq or Axiom? With their integrated 802.11b and Bluetooth it seems like a solution to provide a wireless based sip phone for any user would be possible. Handoff between access points might be problematic but most users I know would be using their PDA phone in an airport with free wireless or at the local cafe, etc, etc... Can

I could have sworn I saw a post about this recently but I can't find it so apologies if this is a dupe, but is there anyway to control the order in the Polycom Buddies list? Bill -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070206/b2d3c1fc/attachment.htm

Hi, I am using Polycom 501 and I came across a problem. As soon as I have incominglimit=1 in sip.conf, which is necessary for buddy watching, I cannot transfer calls. On the console it tells me: Call from user '3052' rejected due to usage limit of 1. Can someone please tell me how to get around this problem? (I don't know if this is relevant, but in the phone.cfg file, I have

Greetings, I just received some netweb-301 phones frm Seshu down in NJ. I cannot for the life of me get it to register with the asterisk server, nor upgrade the firmware to the latest (1.41) i'm still using 1.37. The packets are traversing the router, going into the other subnet, hitting the asterisk box, but not actually making it to asterisk. Nothing in the asterisk logs, but tcpdump

Thanks John, You are my savior. This is such a great relief. Apparently realtime will not use either '127.0.0.1' or 'localhost' to connect to the database. I had to use the actual IP address attached to the NIC before it worked. My OS is Debian just a note and Asterisk HEAD from August 20, 2005 Details below for those who might be swimming in the same pool with me.

I know this is not asterisk specific but we all know this group is used for Polycom issues as well... I have hints working ok on Asterisk. However the Polycom phone will only show the buddies key if there is not a call. This defeats the purpose of using the buddies to see if you can transfer a call to another extension (using the Buddy key to see if they are on the phone). Polycom sip

A follow up (late better than never) Finally had time to sit down and look at this sip.cfg <keys key.scrolling.timeout="1" key.IP_500.31.function.prim="BuddyStatus"/> This turns the Services key which I never use on a 501 into the Buddy Status. It even works while on a call. One touch! Bill ________________________________ From: Bill Gibbs

Hi, I configured Buddy Watch function on my Polycom IP 601. It works well, until I make a reload of Asterisk. After reload, It can't monitor any lines and I have to restart the phone to reactivate this function. Is this a specific problem of asterisk-1.2.3? How can I solve it? Thank in advance, regards, Marco.

All, Slightly off topic. Polycom released their SIP software version 1.6.6 for their phones recently. I was under the impression that this release fixed a previous limitation where the phones would only watch 7 buddies, ie send 7 sip subscriptions to Asterisk. I have configured a phone directory to watch 30 or so appearances, and it still seems to only be sending 7 subscriptions to Asterisk.

I've got two Polycom 500's that I'm playing with, and I want view the status of either phone, (busy/on the phone/etc.) from the other. I've got this cute little 'Buddies' button, and I can add contacts to that. But the status doesnt actually update. Do I need to setup realtime for asterisk? Can anyone point me to some documentation or give me some hints? Thanks, Nihal

Hi all I have a model Member that has a habtm relationship to buddies (also members). Now I wanted to sort them using sort, but this ends in an error "undefined method <=>"! Where''s the problem here? Sorry, I''m still beginning to learn Ruby. ;-) Thanks Josh -- Posted via http://www.ruby-forum.com/.

I'm running into an issue w/ Buddy status on Polycom IP650 phones using buddy status (with SIP Hints) on Asterisk 1.4. Sometimes the status on the phones will "stick" in the busy status. I have noticed that I can call that extension & the status will reset (sometimes). Anyone else encountered this or anything similar. -Chris

Hello! I have 2 table: users and buddies User: id, name, ... Buddy: id, user_id, user_buddy_id, ... So if you have 2 users 1,jack and 2,fred and fred is a buddy of jack, there is a Buddy object: id=1, user_id=1, user_buddy_id=2 I can declare only one belongs_to in Buddy and one has_many in User. And there is conflict if I had the second one (the first one is discarded) class User has_many

There seems to be some hostility to the idea of this being a GNOME or Evolution problem: Bug 542280 ? Refuses to report bugs because gnome 2.16.x is too old View Bug Activity Product: bug-buddy Component: general Version: 2.16.x Status: RESOLVED Resolution: WONTFIX Opened by Mark Hull-Richter (reporter, points: 3) 2008-07-09 23:40 UTC [reply] On CentOS 5.2 (and 5.1 and 5.0, and Red Hat

Received two new Polycom 500 phones. Dhcp and ftp configured properly to load the various files including v2.5.0 bootrom.ld, etc. One of the phones loaded all firmware and config files properly, registers with *, and is usable. The second phone loads bootrom.ld (from the same ftp server on the same wire as the phone), but towards the end of the bootrom.ld load process (about 430 pkts as seen by

It seems that all my CDR is dumping into the Master.csv file. There is a way to create per user/extension CDR but I have looked endlessly in the Wiki, docs, README.CDR, mailing list archives etc.. I can't seem to find a way to do this.. Any help would be appreciated. Thanks! -- Start Your Own ISP! http://www.YourOwnISP.com

I reloaded my asterisk and found some red lines flushing by. When I stopped it I see: WARNING[21481]: cahn_zap.c:9773 setup_zap: Ignoring signalling WARNING[21481]: cahn_zap.c:9773 setup_zap: Ignoring echocancelwhenbridge WARNING[21481]: cahn_zap.c:9773 setup_zap: Ignoring echotraining Reconfigure channel 1, FXO Kewlstart signalling Reconfigure channel 2, FXO Kewlstart signalling

I want to block following types of numbers in my extensions.conf like the premium number in Taiwan: exten => _90204X.,1,Congestion Since I have a DID in USA, I need to block these numbers in USA, as well all emergency numbers, but still let open free (???) service numbers. Can you help me to compile such a list? bye Ronald