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Got my first round of IP500s in today. Anybody have any example sip.cfg files they'd like to share? Tim Jackson Network Engineer Angelina County, Texas (936)639-4827 office (936)414-6723 mobile -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040928/a923e094/attachment.htm

Hello, Please be conscious of Digium's bandwidth and use a Mirror when downloading 1.0. I have mirrored the tarballs at: ftp://ftp.nacs.net/asterisk/ Direct links: ftp://ftp.nacs.net/asterisk/asterisk-1.0.0.tar.gz ftp://ftp.nacs.net/asterisk/asterisk-sounds-1.0.0.tar.gz ftp://ftp.nacs.net/asterisk/libpri-1.0.0.tar.gz -- Vice President of N2Net, a New Age Consulting Service, Inc.

Greetings all, My next step is to purchase a nice VoIP phone for my desk. I have a grandstream, and the sound is great, but I'm looking for more of an office style phone, preferably that can handle multiple lines, has a more flexible display (i.e. name as well as number). SIP would be preferable. Any suggestions? Thanks, Eric

-- Attempting native bridge of SIP/1009-3df0 and SIP/101-f8f1 -- Attempting native bridge of SIP/1009-3df0 and SIP/101-f8f1 RFC3389: 5 bytes, level 0... Aug 27 08:32:16 NOTICE[23572]: rtp.c:289 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible Killed Whenever I make a call between extension 101 and 1009 which are both Xten Xlite SIP clients, I get that error and

How hard is it to integrate * into an existing 3com NBX setup? -Tim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041004/bf01e3d9/attachment.htm

Hi, I picked up a router with 8 voice ports that supports MGCP, but it has several options that I am not familiar with or do not seem apparent in the mgcp.conf. Enter the default IP address for the Notified Entity: [0.0.0.0] Enter the listening port of the Notified Entity: [2427] Enter the IP address for MGCP signalling (Data): [192.168.0.210] Enter the local port for MGCP signaling (Data):

After our department went to using *, I've had several inquiries about doing VoIP for my entire organization (Small county). We have ~10 locations with various links in between (Mostly p2p T1s, some Frame (1.544mbps commit), some ISDN, some VPN over 768kbit internet) Right now we're using several NEC Electra Elite systems, and 2 Nortel Meridian systems. In one of the main locations we have

Hi I want to play a bit with Asterisk. I currentlly install a new system for that and I would like to get your recommendations regarding the linux distro to use there. This is NOT intended to become a general distro flame war. My favorite distro is ******** and no argument that you flame will convince me here (probably because I've heard it before). However I would like to minimize the OS

We're looking at implementing Asterisk in our department in the near future, we're looking at anywhere from 15-25 extensions. The machine we were looking at running this on was a Quad Xeon 450mhz (2MB L2 Cache) w/ 1GB of ram. I've heard bad things about running Asterisk on SMP machines? Would we be running into any performance issues with this machine? Tim Jackson Network Engineer

Hello, I am just curious how many people are hooked up with BroadVoice and have recently been experiencing a lot of dificulty. Joel

I just signed up for the BroadVoice service a few hours ago, but for the life of me I can't get any incoming voice. The incoming connection is fine as it rings my extension from outside, but I can't hear anyone talking. Outgoing voice is working fine though. I've been looking through the archives, but I haven't found a solution to the problem yet. I even tried another router

I recently dug into this, from what I've seen, the best bang for the buck out there is going to be Polycom's. A local vendor has Polycom IP500 phones for $174 shipped to me. IP500 would be comparable to a 7940G I'm assuming. I ran into the same problem with pricing, don't want grandstreams, but can't afford the nice Ciscos. Check out the Polycom's. -Tim -----Original

Same here, I just changed the b to u. Unavailable message is more generic, but it beats it saying busy when its not. -Tim -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Henry Devito Sent: Tuesday, October 05, 2004 8:31 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE:

I'm getting this error on incoming calls on my X100P cards: -- Starting simple switch on 'Zap/1-1' Oct 7 15:49:19 ERROR[74769]: callerid.c:260 callerid_feed: fsk_serie made mylen < 0 (-2) Oct 7 15:49:19 WARNING[74769]: chan_zap.c:5369 ss_thread: CallerID feed failed: Success Oct 7 15:49:19 WARNING[74769]: chan_zap.c:5411 ss_thread: CallerID returned with error on channel

We have Polycom IP500's, and just starting recently (after the broadvoice patch I might add) after about 1-2 days these phones ring, and answer, but we get no audio on the phones. The caller can hear us, but we cannot hear the caller. Its happened 4-5 times and is only intermittent. No errors on the console, using g.711u. Any ideas? Tim Jackson Network Engineer Angelina County, Texas