search for: 26xx

Displaying 9 results from an estimated 9 matches for "26xx".

Did you mean: 26x
2005 Feb 05
9
Hot Fallover
...d found that a product ''UCARP'' (http://www.ucarp.org/) might provide a solution. Our current setup is (same as on Shorewall web site) as follows: T1 ---- | Cisco 26xx Router ----------------- | Shorewall Firewall Server (FW) ------------------------------ | ------------------------------- | |...
2004 Jan 26
0
canreinvite and codec negotations... and NAT
...uot; <123-456-7890> disallow=all allow=g729 [321] type=friend secret=abc nat=yes host=dynamic canreinvite=yes qualify=200 context=default dtmfmode=rfc2833 mailbox=321 callerid="Joe Blow" <321-456-7890> disallow=all allow=ulaw Okay, in this configs, gateway would be my cisco 26xx gateway.. ext 123 would be a g729 customer.. and 321 would be a ulaw customer. When someone calls ext 123, the cisco send the call to asterisk, it *WILL* use ulaw... then it will initiate a call to g729, well... Now we have a codec mismatch, and canreinvite won't work... EVEN though gateway c...
2004 Jan 29
0
canreinvite and codec negotations...
...uot; <123-456-7890> disallow=all allow=g729 [321] type=friend secret=abc nat=yes host=dynamic canreinvite=yes qualify=200 context=default dtmfmode=rfc2833 mailbox=321 callerid="Joe Blow" <321-456-7890> disallow=all allow=ulaw Okay, in this configs, gateway would be my cisco 26xx gateway.. ext 123 would be a g729 customer.. and 321 would be a ulaw customer. When someone calls ext 123, the cisco send the call to asterisk, it *WILL* use ulaw... then it will initiate a call to g729, well... Now we have a codec mismatch, and canreinvite won't work... EVEN though gateway c...
2004 Sep 16
0
ISDN BRI termination via Cisco?
...y would prefer not to run a one-off box with a relatively foreign (to us) OS which seems like even those with some familiarity have trouble making work. So I was looking at Cisco stuff, for something unrelated, and noticed that they have the "VG 200", which appears to maybe be a rebadged 26xx class router. Does anyone have any experience terminating NI1 ISDN BRI's on one of these for use with Asterisk? How hard is it to make this work? .. JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net "We call it the 'one bite at the apple' rule. Give...
2005 Jul 08
2
best Fax board?
Prospective user question? What is the simplest/inexpensive board to use in order to be able to receive faxes in Asterisk. I have a couple of cards I bought off E-bay think they were TX-1000 (or supposed to be anyways) but I assume I need some form of fax card though. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:
2005 Feb 04
4
BRI in the US?
OK, I asked this about a week back and met with no repsonse at all. But perhaps its worth trying again. Does anyone on-list have * running BRI to their local telco? I'm considering this as an alternative to my TDM400p card. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc.
2006 Dec 07
5
CISCO 2600 - VWIC 1MFT-E1
Hi to all, I got a Cisco 2651XM wired to an E1 PRI. What I want to do is to pass all incoming calls to my asterisk. This is my actual conf: http://pastebin.ca/270677 with this I'm able to call my number from outside, but the call stop on the 2600, infact I can hear the tone, but I'm not able to forward calls to my asterisk. Anyone got an idea of my errors? Thanks to all. -- .:FaberK:.
2008 Oct 03
3
OT: RIP settings for private netblocks
I am contemplating converting some of our internal networks from routable to private IPv4 address space. I have a question about RIP as implemented under Cisco IOS 12.x. Presently the setting for rip is: router rip version 2 passive-interface [[FastEthernet]]0/0 network aaa.bbb.ccc.0 no auto-summary What I would like to know is how one routes the entire 192.168/16 address space using rip.
2013 Mar 21
9
Asterisk disconnecting SIP Calls after 15 Minutes
Hi @ll, I just moved my Asterisk Box and changed the Provider and Internet Access to a full IP Access by Deutsche Telekom. I set up my sip.conf as I found various examples throughout the Net. Calls and some other stuff is basically working. The problem I ran into is, that the outgoing and incoming calls are dropped after exactly 15 Minutes. Solution for this should be setting the