similar to: Any experience with Linksys WRT54GP2 as localextensions to Asterisk ?

Hi, I'd just like to confirm compatibility of Linksys router WRT54GP2 as local extensions to Asterisk. Can it register to local Asterisk behing him ? How stable/good is analog interface ? Any experience would be more than welcome. Thanks in advance, regards, Rob.

So I'm using a WRT54GP2-NA when I travel, as I travel alot, to give me a phone at my hotel rooms, etc. During the day or late at night the thing works great - best ATA I've ever used. However, in the mid-evening (when many business travellers are at the hotel room doing work), the outgoing audio channel gets so choppy that the person on the other end can't make me out clearly.

Hi, I'm trying to set up Asterisk behind my WRT54GP2 router that has a intergrated ATA box. My box are not locked in any way so I can access and change all settings. Now to the problem... I have gotten Asterisk to register with my provider and everything works just well.. Now it's time to get the intergrated ATA to connect to asterisk. But the asterisk box in located on the LAN ports of

As far as I can remember I only opened sip and tftp ports for the phone. For some reason (didn't look into it too much) the call stays with sip and doesn't use RTP. The problem you describe (the call doesn't even ring on the other side) is something I had and was solved by upgrading the firmware. Checkpoint's tracker explicitly said what connection attempts were blocked and why.

Check your FW-1 tracker and see if any sip packets are dropped during call initiation. I had this problem and it went away when I upgraded the BT's firmware to the latest (16). Beware, though, that people on the list claim that this firmware breaks functionality of the message button and autoanswer. I haven't checked this yet, cause I can't afford to go back a version. I prefer a

I'm looking for somehting like the standard house hold linksys/dlink router. Basically it needs to have at least 1x100mbit port, wireless G capabilitys and at least 1 x anolog voip/sip connection. I've found linksys's WRT54GP2 which appears to do what i want. Anybody use this? Does it require vontage's service? I'm looking for any recommendations. Thanks -- ~Shaun

I'm looking to upgrade my unit, and would like to not have to wait on our company's suppliers to get back to me on it. Thanks in advance for any help -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050919/347b3af8/attachment.htm

Hallo! I had ordered a Sipura SPA-3000 in the UK, but the supplier turned out to be unreliable and never shipped. Yesterday I went looking for alternative suppliers and found the Linksys SPA3000 device. It's a different box, but the specs look very similar. Is this the same device? Has anyone used this Linksys SPA3000 successfully with Asterisk? Thanks, Frank

>Also, I'm sure you've probably checked on this one, >but are the phones registered with asterisk? >You can make outbound calls on them without them >actually being registered. I'm assuming you can >still get in and see the CLI. What does "sip show peers" >look like? What does "sip show peer xxx" show? >What does the CLI show when you

Hi asterisk users, I am in need of information about how to configure the sip.conf and extension.conf for subscribers to support the dialog event package rfc 4235. I am using asterisk 1.6.0.1 version. The below are the configuration of sip.conf and extension.conf files which I have done. I have three subscribers as one from my application(App) and other are x-lite1 and

Hi asterisk users, I am in need of information about how to configure the sip.conf and extension.conf for subscribers to support the dialog event package rfc 4235. I am using asterisk 1.6.0.1 version. The below are the configuration of sip.conf and extension.conf files which I have done. I have three subscribers as one from my application(App) and other are x-lite1 and

I am using PBXware for configuring users and extensions. Pbxware uses Internal script called init.sh to process the calls based on its own version of extensions.conf defined in the GUI. I have IAX2 Extensions 56 and 101 and SIP extensions 50 and 51. I have used IAX2 extension 101 and dialed SIP Extension 51 But the PBXWare's Init.sh AGI command identifies the DNIS as another IAX

Hi, I'd like to organize my Asterisk to properly handle incoming SIP/IAX/H323 callerids so they can be called back if needed. I have three incoming contexts for sip, iax and h323 calls. To each incoming call I'd like to prepend certain number that will be catched with pattern matching on output calls. For instance for iax I have: [from-iax] exten => s,1,NoOp(IAX call from outside

Sirs, I have recently been interested in your Maxstat. I have computed with my own programme the ranks (by using the Kaplan-Meier method and the log-rank test) with the formula (Observed-Expected)/(SQR Var). The results are similar but not exact to the M value obtained with the Maxstat. I would like to know whether you are using some correction or adjustment in computing the different ranks. Thank

What would be the simplest way to retrieve information form a CNAM database that provides http based query responses? Does an application or script already exist that does this? Basically, I want to do a wget of a URL that contains the callerID number as a variable, and assign the returned text to another variable which can be used to set the caller ID name. Any suggestions?

I have a situation here where a user has an AAstra 480i phone, which function corectly. The phone is behing a nat-router (a linksys wrv200 for it's VPN point to point facility). The phone is plugued in a port wich has qos enabled. And when the user places a call, sometimes (not always), we get this in the console : [May 8 13:41:55] WARNING[5804]: chan_sip.c:1948 retrans_pkt: Maximum

Hello all I'm trying to get my asterisk config to listen to multiple ports. This is since some clients have port 5060 blocked by their ISP. Does anyone know how to do this in sip.conf or if it is even supported? Thanks!

print to standard error output in your perl script: print STDERR "This is how perl-AGI prints to Asterisk CLI output\n"; MATT--- -----Original Message----- From: Robert Rozman [mailto:rozman@fri.uni-lj.si] Sent: Tuesday, August 24, 2004 8:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Perl AGI - no output from agi script to Asterisk Hi,

> -----Original Message----- > From: Robert Rozman [mailto:rozman@fri.uni-lj.si] > Sent: Monday, January 17, 2005 7:45 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Can I start recording channel in > the middle ofconversation ? > > > Hi, > > I'd kindly ask for simple example if this is possible ? > > Is

> Date: Thu, 23 Jun 2005 08:50:50 +0200 > From: "Robert Rozman" <rozman@fri.uni-lj.si> > Subject: [Asterisk-Users] Loosing hair on connecting Panasonic PBX- * > - Euroisdn Italy > > I'm pulling my hair down and getting bold :-) ..... I have Asterisk between > Panasonic KXTD816 and Euroisdn in Italy (beronet octobri and bristuff > Asterisk).... Plenty of