similar to: Cisco 7960 x Asterisk CVS-HEAD-03/13/05 - registration issues

yes, I narrowed eyes and see the world again. I's eyes narrowed, at present of scenery was draped one hazy cloak, a fairly beautiful come out in the wash. As long as the phenomenon don't essence, some things, some things, some people don't need to see as well, as long as I see things scenery beautiful that side. www.ebuybus.com

I've being using the same non-nat config for a 7960 for about a year with no issues. I have just upgraded cvs head from the 16th of July to todays. About a minute after I make a call I get the message chan_sip.c:1132 retrans_pkt: Maximum retries exceeded on transmission 000b46a0-8661007f-5c457e72-2b0dba2c@192.168.123.20 for seqno 102 (Critical Response) I don't get this for other

I'm trying to place a call from my Cisco 7960 and I'm receiving this error: Mar 1 06:19:44 NOTICE[3060]: chan_sip.c:7399 handle_request: Unable to create/find channel Mar 1 06:19:58 NOTICE[3060]: chan_sip.c:7399 handle_request: Unable to create/find channel I can't place calls, but I can receive them: mail*CLI> sip show channels Peer User/ANR Call ID Seq

At the moment, Compiz draws each head at a time, shifting the glViewport() for every head in a xinerama-like setup. However, the projection matrix is NOT adjusted. This means any translation along the z-axis will cause problems and any rotation along the x/y axis, because the projection matrix is centered around the viewport, not the actual X screen as it has to be to keep the scenery together. I

I'm trying to get my Cisco 7960 configured to work with Asterisk, with no luck. I'm sure I'm missing something very easy... since I know others have this working. I've stepped through Andy Powell's excellent "Getting Started with Asterisk", and it works for my X-Lite softphone. My sip.conf entry for the cisco looks like this: [cisco] type=friend username=cisco

Hi all I have a problem to register a cisco 7960 to an asterisk 1.2.2 I defined in sip.conf the next : ["phonenumber"] type=friend username="username" secret="password" host=dynamic context=work I am trying to catch the register requests with sip debug with no success (empty screen). I can only catch the register messages with ngrep on host it's comming

Can someone point me in a direction where I can acquire the Cisco 7960 SIP firmware? I just purchased two phones of ebay, but they are both in Call Manager mode. Thanks, Marcus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030515/6265b777/attachment.htm

https://bugs.freedesktop.org/show_bug.cgi?id=38673 Summary: all object is black Product: Mesa Version: unspecified Platform: x86 (IA32) OS/Version: Linux (All) Status: NEW Severity: major Priority: medium Component: Drivers/DRI/nouveau AssignedTo: nouveau at lists.freedesktop.org

I'm having a wierd problem with a Cisco 7960 (sccp2) and asterisk (1.4.2) If the call that I'm trying to make goes through, everything works fine. But if there's any sort of error (like me messing around in my extensions.conf, etc). I can't get the connection to drop. ie: If I get the conjestion tone and hang up the phone, I can do a sccp show channels I can see that the

I'm trying to setup a couple of Cisco 7960's in asterisk. I have asterisk working fine for sip clients, and can call the 7960's just fine, but I can't seem to dial out on them. As soon as I enter the first digit, the phone attempts to dial it without waiting for the rest. I've changed timeout settings, etc but can't seem to get it to work. Any ideas? Asterisk

Hi, I'm installing asterisk 1.0.5 for a partner in China. Since the ping round trip takes typically 600 msec, I doubt, whether voice quality will we satisfying, but that is currently not my concern. The problem is, that most SIP phones or software (e.g. SJPhone) do resend the invite request, after approx 500 msec (measured by ethereal). chan_sip from asterisk seems to have a special

I'm having an interesting problem with a Cisco 7960 phone, and two Asterisk servers. I'm not sure if this problem is specific to the 7960, or even to Asterisk for that matter. Here's the scenario. I have an * server at one location with a public IP address (i.e. not behing NAT). I have a second * server and 7960 phone at another location. This one is on a private LAN, and uses NAT to

Anyone help me here? I am a newbie so be gentle ;-)...... It worked once and then I played with the configs. I have a static IP address which is on my private network.. Phone is 192.192.192.220 and asterisk server is 192.192.192.22 I have the 7690 with a SIP iamge (Whatever latest is ) I have 3 lines setup with Free World Dial up and have the 4th setup to connect to my asterisk server. Here

Sorry if this comes in twice. Wasn't subscribed first time :-( Anyone help me here...... It worked once :-( I have a static IP address which is on my private network.. Phone is 192.192.192.220 and asterisk server is 192.192.192.22 I have the 7690 with a SIP iamge (Whatever latest is ) I have 3 lines setup with Free World Dial up and have the 4th setup to connect to my asterisk server.

I previously reported a problem with receiving inbound calls with Galaxy Voice and the Current CVS Head version. I am currently running version 5/17/04 of the CVS HEAD and my incoming calls work correctly with Galaxy Voice. If I upgrade to the Stable release or the current CVS head I am no longer able to receive incoming calls. I would really like to use the current version of the CVS as it has

I tried using CVS HEAD 08-05-2004 for zaptel/asterisk last night but I'm having problems with zaptel. The last CVS HEAD I was using was CVS HEAD 07-31-2004. (BTW... I'm a CVS moron... what command would I use to get back to that version of HEAD because like an idiot I forgot to make a backup copy of the CVS HEAD on my system before I install from the new CVS HEAD. I tried cvs co -D

Hello, I'm considering using Asterisk with some type of Cisco phone, and currently considering either the 7940 or 7960 because of its more-complete functionality (compared to the 7905). I'm currently wondering: Do all the expected functions (transfer, conference, voice mail, message waiting indicator, etc.) work normally with Asterisk over SIP? What caveats are known about using

With todays CVS head I am getting the following being sent after a call has been terminated on my Cisco 7960. It eventually gives up with a critical error. chan_sip.c:1132 retrans_pkt: Maximum retries exceeded on transmission 000b46a0-8661000a-4405e325-7e25031f@192.168.123.20 for seqno 102 (Critical Response) Any ideas I am sure it was working ok with cvs head a month ago. Chris ---- sip

I have a asterisk server running on site listening on a public ip. Tonight I attempted to connect a Cisco 7960 phone from my home location via sip but failed. My home network is simple, Cox cable connection hooked to a linksys wrt router. The firewall on the linksys router is disabled and I even setup dmz to the phones ip as a last resort. I removed the linksys router and plugged the phone

Hi I just joined I have a serer who's purpose life is to be a fileserver/print server right now it is running windows 98 because it is an older computer look at an alternate os such as this will this allow windows based machines to talk to it save files retrieve files and also how difficult is it to install. -------------- next part -------------- An HTML attachment was scrubbed... URL: