similar to: Cisco 7971 behind NAT

Hi everyone, I'm wondering I would accomplish the following: I want to dial several SIP extensions simultaneously, HOWEVER, for different times (say ext 10 for 15 sec and ext 11 for 30 sec), and potentially with different headers (such as ALERT_INFO) and codecs for each extension. Obviously whoever picks up first gets the call. After the longest timeout expires (30 sec in this example) I want

Hi all, I've just bought some SPA922. First time with this hardware for me. I see no LAN tab in its web GUI where I can setup NAT for PC conected to its LAN ethernet port. However, when I connect a PC to that port, SPA922 works as bridge. Anybody can confirm SPA922 can NAT a PC connected to its LAN port? Does exist such LAN tab for setting up parameters as port forwarding? (by the way,

Hello List, Does anyone have access to the soft key configuration files for the Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and didn't find much up there. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 roy at manistee.org -------------- next part

I have many remote locations that dial into a central server to retrieve voicemail via IAX. Outbound calls are handled as SIP calls from a Snom to a local (to them) Asterisk server that dials the main server thru IAX. I have trained them to check their voicemail via the emailed WAV file, however some of them are, how shall I put it, idiots*, and insist that they *have* to have the MWI indicator

Yeah, I'd like to hear you guys' opinion instead of CleverNickName's! -----Original Message----- From: Luki [mailto:lugosoft@gmail.com] Sent: Tuesday, February 15, 2005 9:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OT: Comments on Vonage SIP port blocking complai nts?? You can always visit Slashdot for countless (useless, well,

-----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Mohammed Firdosh Nasim Sent: Tuesday, March 15, 2005 11:08 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] How to register two SIP phones ( e.g. WindowsMessenger) from different subnet to * On Sat, 2005-03-12 at 07:42, Luki wrote: >

RE Message: 5 Date: Sat, 7 May 2005 23:18:46 -0400 From: Andrew Kohlsmith akohlsmith-asterisk@benshaw.com Subject: Re: [Asterisk-Users] At home Asterisk via Broadvoice? To: asterisk-users@lists.digium.com On May 7, 2005 11:04 pm, John Stegenga wrote: > Broadvoice will give me 2 lines, with 2 phone numbers each - distinctive > ring - for a reasonable fee... Please do a google search for

Guys. Anybody know why sometimes on some servers Asterisk shows more than once while doing a ps? [root@server2 akrall]# ps -ax|grep asterisk 20555 ? S 0:00 /bin/sh /usr/sbin/safe_asterisk 20557 ? S 0:00 asterisk -vvvg -c 20558 ? S 0:00 asterisk -vvvg -c 20560 ? S 0:00 asterisk -vvvg -c 20561 ? S 0:00 asterisk -vvvg -c 20562 ? S

Hi All, As the subject describes, has anyone gotten this to work? I am running an asterisk 1.2.16 server, and am trying to register my cisco 7970 remotely to it, but it just won't go. I am running 1.4.2 internally and the phone registers fine to it. I'm using the latest firmware (i think) - 8.2.1S On the server in question I have tried the following for the sip declaration:

We're investigating audio quality issues in our system; maybe someone can help. We're using Asterisk as a basic PBX, with a single PRI on one side and SIP phones on the other: Sipura SPA-841's. We're experiencing several audio effects which seem to commonly correspond to network failures (packet loss, high jitter, etc manifested as "robot voice", dropouts, periodic

i have been trying to get a newly liberated (from vonage) cisco ata-186 (sip ios v3.1) working properly with asterisk. my client is behind a linksys wrt-54g, which up to this point hasn't proven to be a problem (i have several sipura spa-2000's and polycom phones working just fine behind them). (i'm running cvs-head from yesterday). after looking at the various suggestions,

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello everyone Due to security reasons I want to run asterisk as a non root. I normaly installed asterisk, created an * user, moved the binaries to /usr/bin and chowned all the files and directories mentiont in the * manual (handbook-draft.pdf) Now I can start * but I get the following warning (which I don't get if I run it as a root): Feb 14

Hi, Some help with dimensioning the server will be gladly accepted. -50 sip phones (g729) or g711(to avoid transcoding) in LAN -an asterisk server (1.4) doing normal pbx functions + voicemail in the same LAN -Some sporadic conferencing with no more than 2 sip phones and maybe 2 or 3 calls coming from the E1 for a total of 5 people in a conference. The asterisk server will get an E1(pri) via one

On Feb 2, 2008 10:16 AM, Luki <looky at poczta.fm> wrote: > Hello Carlos > > could you please help me to solve issue with Mustek 1000VA UPS under > Freebsd ? > I am not able to connect with it : > > >Network UPS Tools - UPS driver controller > >Network UPS Tools - Megatec protocol driver 1.5.4 [megatec] > >Carlos Rodrigues (c) 2003-2007 > >

I have used Grandstream phones for years, and have just started testing a Cisco 7940 (with SIP firmware 7.4). I have found something puzzling and don't know whether it's just a limitation or something I haven't done correctly. The Asterisk server is directly on the Internet with a public IP. The phones are on a private LAN with a NAT router to the Internet. The sip.conf entries for

I need to set up an office full of Cisco 7960 phones behind NAT with the server out in Colo. The first test phone registers fine, but the second one does not register. The first phone's registration looks like so: /SIP/Registry/3115552368 :64.169.xx.yyy:38836:3600:3115552368:sip:3115552368@64.169.xx.yyy:5060 When the second phone tries to register, it gets back a 404 not found. Not a

Hi All; I succeeded to have a success call from Polycom behind NAT while Asterisk has public IP address, but I was not able to have a succeed call (it was established, but no voice running, and then the call disconnected) if Asterisk behind NAT and Polycom behind NAT. When Asterisk behind NAT and Polycom behind NAT, I forwarded the 5060 UDP to asterisk (at asterisk router) and to Polycom IP

Hi, I have the following scenario. I have an Asterisk server running on an internal IP address behind a firewall, and I have a remote user trying to connect to my Asterisk box behind his firewall, but he can't seem to get a connection. I have opened up the port (5060) so that he can connect through my firewall, but it still doesn't appear to want to connect. I am pretty sure that the

Hi All, I've an Asterisk Server behind a NAT. Using DNAT, I've opened port 5060 and all 10000:20000 udp. Sip configured with externalip and subnet. I've another site, also with NAT, where I map the rtp port (as defined in the client) to map to the local client (DNAT). Using Xlite, this configuration works, it requires using the quality=yes and NAT=yes/always in the sip ext

i've got * and phones in small home network all behind NAT. Outbound to iconnect proxy works great. Now to get in/out working with another carrier. Carrier2, Commpartners, i have working with one of the phones and a soft phone without * just fine. Next I register the phone with * fine. Create a trunk, but it the trunk fails to register... help I'm getting the following msg during