similar to: ftmp header

Anyone have experience with Sipura's? Anyone know if they offer a warranty? Would like opinions on these, good or flame. We bought *one* to test with and it died, can't even get a response from Sipura "support". Could anyone recommend another device to replace these? Prefer 1 or 2 port design. Ty :-)

Any ideas? I edited the Makefile as instructed, ty. Now compiling .... sccp_channel.c 279 lines sccp_channel.c: In function `sccp_channel_send_callinfo': sccp_channel.c:48: structure has no member named `callerid' sccp_channel.c:49: structure has no member named `callerid' sccp_channel.c:49: structure has no member named `callerid' sccp_channel.c:49: structure has no

Ok, we're trying to use Asterisk as a PBX in our office. Our original plan was to use a Cisco 7960 with a 7914 attached. Short story is, no one updated chan_sccp in a long time and the 7914 is questionable at best anyway from what I've heard. We couldn't ever get chan_sccp to compile, I went to an older version of Asterisk and that broke some of our SIP devices. We tried using a couple

Having problems getting realtime working, I'm trying to use odbc for all of this. I've got Fedora 3 and have been fighting with odbc for a day now. I think I got it working correctly, however I can't seem to get the realtime portion working. In asterisk 'odbc show' shows it connected, I see it on my (odbc) mysql server connected and all, it connects and just idles. So, without

We're looking at implementing Asterisk in our department in the near future, we're looking at anywhere from 15-25 extensions. The machine we were looking at running this on was a Quad Xeon 450mhz (2MB L2 Cache) w/ 1GB of ram. I've heard bad things about running Asterisk on SMP machines? Would we be running into any performance issues with this machine? Tim Jackson Network Engineer

All, I get this whenever trying to dial to a peer when the peer registered to another server. I'm basically trying to use realtime to check for the peer and dial it. Mar 24 09:16:47 VERBOSE[4527]: -- Executing Dial("SIP/brak-f69f", "IAX2/brak-test/107") in new stack Mar 24 09:16:47 DEBUG[4527]: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name =

Hi all, Please find below an updated Speex Internet Draft document. It would be good if we could book some time for discussion on Speex at the IETF meeting in Vienna (scheduled for 14th July). The cutoff for submission is 9:00am EDT, (GMT -04:00), 30th June. Comments and feedback welcomed! Regards Phil

Looks good to me. Jean-Marc Alfred E. Heggestad a ?crit : > Hi > > Please find an updated version of the Speex I-D attached. The only > change is addition of the copyright conditions in Appendix A, > as requested by Ivo. > > Many thanks for your input. > > I will give you a few more days before submitting to AVT working group > > > /alfred > > Ivo

Hi all We are about to send an updated version of the internet draft "RTP Payload Format for the Speex Codec" to the IETF AVT working group. Before submitting we would like your input, if you have any comments or input please send them to the mailing list. If we don't get any comments in 1 week (by 22. May 2007) we will go ahead and submit it to the IETF. Of course you can comment

Hello, Is there someody who can explain me the meaning of these sentence. "Sip is philosophically horizontal and H.323/MGCP are vertical" Thank U (if you have some links to share about this protocols, share it :) ) Ignace

Do not forget to add the "Copying conditions" to the RFC. Check http://wiki.debian.org/NonFreeIETFDocuments That page contains a section titled "Template for RFC authors to release additional rights". To follow that guideline a section like the following should be added: x. Copying conditions The author(s) agree to grant third parties the irrevocable right to

The latest revision of the draft RTP Profile is attached for review. This will be submitted to the IETF Audio-Video Transport Working Group for consideration immediately, so if you have any more comments, let us know. In addition, we will be applying for an official MIME type. Note that the AVP code and the MIME type in this latest revision have been changed from "SPX" to

Does anyone have some decent Nagios scripts out there that do more than monitor the proc itself? Rather than reinvite the wheel, figured I'd ask. I already saw the one on the wiki. Matt

We asked Henning Schulzrinne (the author of RTP and currently the RTP/AVG maintainer) about getting a static RTP payload type code assigned for Speex, since we meet the criteria in rfc1890. He said no. Here is his answer: On Fri, 22 Nov 2002, Henning Schulzrinne wrote: > Sorry, this is not going to happen, regardless of the codec or its > merit. See the new draft which will replace RFC

All, I'm not sure how to word this question but we're noticing a lot of our asterisk boxes no longer have multiple asterisk child processes. i.e. doing a 'ps ax' reveals only 1 asterisk PID when normally I'm used to seeing 8+ .. There is no rhyme or reason to it, and we're using the safe_asterisk script which has always worked in the past. Ast 1.2.4, zap 1.2.4, naturally..

Ok, here's an odd one. I would have opened a bug on this but last time I tried that I got flamed.. :) Problem: When proxy requests digest challenge (SIP) Asterisk responds normally with the exception that for some reason it changes the FROM: (Also changes Contact: )to what's in the original TO: line. Why on earth is it doing this?! It must be a bug, I've gone over my extensions.conf

Has anyone come up with a way to get power to a TDM400P card installed in a Dell PowerEdge 1750? Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this

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On Tue, Dec 17, 2002 at 11:55:21PM -0800, Greg Herlein wrote: > If such a thing happens, discussion of the RTP profile draft > would be most welcome - please get responses back to the > list! Now, if this were finalised before the conference then we could do a demo and use it for broadcasting the lectures streams around the world... What is currently the best way of doing this? I'm

Hello, I've recently received a Cisco 7960G phone with the factory default SCCP firmware on it. As we're using SIP on our network, the first things i've done was to upgrade but unfortunately the phone just restarted. By looking on the TFTP logs and tcpump output, i've seen that the phone crashed and restarted just after downloading the OS79XX.TXT file, without requesting the image