similar to: Browsing CVS Online

incominglimit is already implemented for SIP. Just specify under the endpoint how many incoming connections are allowed. For example, [cisco] type=friend username=cisco secret=blah nat=yes ; This phone may be natted host=dynamic canreinvite=no ; Cisco poops on reinvite sometimes qualify=200 ; Qualify peer is no more than 200ms away

Hey all, I was playing around with IAXTEL last nite and have outgoing calls working a treat. I'm sure I woke a few people up in the US with my annoying test calls. :) Anywayz, incoming calls are a different matter. I have a NAT firewall my * box is sitting behind and the server 'appears' to have registered correctly with IAXTEL. Thing is, when I try and call my 1700 number

Hi all, ? Just a quick one.? Should I be able to call myself through IAXTEL using my 1700 number?? I'm behind a NAT firewall and can call other numbers, I just want to test incoming calls somehow to make sure I can accept them from IAXTEL. ? Regards, Jamie Carl Email: me@jazz-inc.net Phone: +61 414 365 466 Jabber: jazz@netmindz.net ? -------------- next part -------------- An

Hey all, ? I'm trying to get an E400P and 2 X100Ps working together in the one box and don't seem to be having much luck. ? I can get the two different types of boards working separately, but not together.? I've made calls on both the X100P and have seen sync and correct signalling on the E400P.? But when I try to enable to configs together I get the following: ? modprobe wcfxo

*This message was transferred with a trial version of CommuniGate(tm) Pro* Hey all, Just noticed something that might be an issue. I have just made asterisk crash consistently by doing the following. I have a D-Link DG1102s running MGCP into asterisk and an extension *9 setup which dumps me into my inbound context to simulate calls coming in from my X100P. This usually works with no hassles

*This message was transferred with a trial version of CommuniGate(tm) Pro* Hey all, I've just done a quick (but functional) web front end for searching the CDRs in a MySQL database. Anyone interested in trying it out? I'm wondering what to add to it next. So far you can seach using source, destination, CLI, channel and date ranges. It also displays ALL fields in the database table.

*This message was transferred with a trial version of CommuniGate(tm) Pro* Will asterisk actually convert between two different codecs????? ie, a SIP endpoint running GSM and another running G.711? Wouldn't that add quite some latency? I was always under the impression Asterisk did not recompress and was smart enough to negotiate the right codec at each end and just pass through the RTP

Does anyone know if this unit works with Asterisk? Thx. B.

Look like SNOM have rolled back the firmware version of the 200's from 1.16w to 1.16q.. Anyone know why? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Hi, I have just been playing with the latest X-Lite.. It works fine with Asterisk.. As for codecs I tested G.711a/u, GSM and iLBC... iLBC is the only one that didn't work.. not sure why.. But the bigger problem is that when I call another extension that is using a Snom200 the call connects but there is no audio in either direction.. I have tried G.711a/u and GSM and while X-Lite shows that

Hi Anybody have a example or use this modem as FXO, acording to intel this modem is a real full duplex modem, so i have one question why i can we use this modem as fxo My idea is simple when you recive a call in the modem you can transfer this call to any other place using for example a IAX2 or Openh323 protocol This is posible? Thanks

Hello, I'm new user of asterisk. Can anybody pls tell me how to use asterisk or any detail how to link???? i installed Asterisk-0.4.0 on i810 onboard sound card with Redhat 7.1. when i type "asterisk -vvvc" i get *CLI> prompt Prakash Get Your Private, Free E-mail from Indiatimes at http://email.indiatimes.com Buy The Best In BOOKS at

I manage internet access for our company. We have our main link which is a 2 meg leased line, we then have a 2 meg adsl line for backup from a different ISP What i want is live failover for outgoing traffic, not only for complete link failures but also when then main isp has routing issues to a given ip range (e.g during the recent disruption to 2 of the transatlantic links) The solution

A while ago SIP transfer via the # key on a call to a cell phone via iconnect was working. I updated to the current CVS tonight and now that functionality is gone. Any ideas as to how to enable it again? Thanks in advance -russ

hi, i have a external usrobotics modem, i want to use it with asterisk to interact with the pstn, what i have to do? thanks, -- santiago jos? ruano rinc?n administraci?n servidores y servicios de internet red de datos universidad del cauca -----BEGIN PGP MESSAGE----- Version: GnuPG v1.0.6 (GNU/Linux) Comment: For info see http://www.gnupg.org

On Sun, Aug 17, 2003 at 03:44:21AM -0500, Gnophone Support wrote: > Hello, and thank you for registering at gnophone.com. Your login > information is listed below: > > Username: miernik > Password: ******* > IAX Phone Number: 17002916107 > > Please login as soon as possible to > http://x.linux-support.net/directory/ to complete the

Hi all, I'm setting up my first * install and have it peering with another * machine using IAX across the internet which provides our pstn gateway. So far I have the IAX "friend" set up correctly but when I make a test call from an external phone, I get: WARNING[5126]: File chan_iax.c, Line 648 (get_timelen): Don't know how to calculate timelen on 8 packets I have set up a

My system seems to be generating a call on its own... Unfortuately I can't give much more information.. I have an X100P and an S100U.. My Modem and the X100P share a common line.. When I am on the internet (which is most of the day) * just sits there and does nothing (apart from when I am testing ideas for the dial plan), but at night when I am sleeping and the modem is not connected then

Hi Is posible to make a call from site A to Site C, and my question is, the rtp data is from A to C or is from A to B to C Site A Site B Site C ata186<-------->FW<--------->Asterisk<--------->FW<----------->ata186 Thanks -------------- next part -------------- An HTML attachment was scrubbed...

Hi everybody, I'm trying to SIP register between two asterisk, each one have a Public IP. Asterisk told me that Unathorizae In * one sip.conf register =>usuario1:pass1@<public_ip_2> In * two sip.conf [usuario1] type=friend username=usuario1 secret=pass1 host=<public_ip_1> dtmfmode=inband Logs in * are the followings In * one logs: Sip