similar to: Strange problem with call hangup on Budgetone 102 Phones

Dear all, debian buster, two DCs (DC1 & DC2) samba packages updated from 4.11.9+dfsg-0.1buster1 to 4.12.3+dfsg-0.1buster1 this past Sunday night (Louis' packages). Since then, samba-tool drs showrepl shows a replication failure. DNS updates with samba-tool fail, example: root at dc1:~# samba-tool dns add dc1 samdom.example.com testrec CNAME afs3.samdom.example.com -k yes Failed to bind

[This announcement covers version 1.0.58 too since it didn't get a separate message.] We are pleased to announce the release of libguestfs 1.0.59. Home page: http://libguestfs.org/ Downloads: http://libguestfs.org/download Git repo: http://git.et.redhat.com/?p=libguestfs.git;a=summary New in this release: - Support for Linux extended attributes. - Allow guestfish to be controlled

All: Attached please find yet another RTP profile revision (v5). You can also find the document at: http://www.herlein.com/downloads/speex/docs/ Changes: - added vbr, cng, ebw, sr optional parameters to MIME - added vbr, cng, ebw a=fmtp options for SDP use - added required document attributes for submission to IETF and IANA (format and author contact information). Note that we

I'm looking at RFC 3261, I think the latest SIP standard and have a few questions about the * sip implementation: 1. 8.2.6.1 Sending a Provisional Response says that UASs SHOULD NOT issue a provisional response to non-INVITE requests.

The latest revision of the draft profile for use of Speex in RTP is attached. We plan on submitting this - or a modified version of this, based on immediate feedback - to the IETF on Monday for consideration at the next meeting. Major differences in this revision are: - removed the discussion in the MIME section. It's a duplicate of the SDP discussion anyway, and may or may not match the

> Otherwise Greg, can you send the latest version of the RTP draft so I > can put it on the site (the current one is getting old)? Attached for all to see. Greg -------------- next part -------------- Internet Engineering Task Force Greg Herlein Internet Draft Jean-Marc Valin draft-herlein-speex-rtp-profile-06

Alfred E. Heggestad wrote: > <...> > > 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 on it also after it > has been submitted, but we would like to get the input from the Speex > community first.. > thanks for all the input. please find attached an updated version of the draft. I

Hi all, Please find below the -01 update to draft-herlein-speex-rtp-profile, as submitted to the IETF. Regards Phil <p>-------------------8<-----------------------------------8<--------------------- <p><p>Internet Engineering Task Force Greg Herlein Internet Draft Jean-Marc Valin

Hello, What's the purpose of the 'sr' sdp parameter ? The sample rate is already given in the a=rtpmap line ? Simon Le dim 29/06/2003 à 12:12, philkerr@elec.gla.ac.uk a écrit : > 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

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 Emanuel Gon?alves wrote: > Do not forget to add the "Copying conditions" to the RFC. > > Check

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 everybody! happy new year ;-) I have a problem with inbound calls on my Diva Server PRI This is my problem: 2005-01-05 15:38:04 ERROR[31716]: chan_capi.c:1696 pipe_msg: did not find device for msn = 4132 capi.conf: [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 language=de [interfaces] msn=4132 incomingmsn=* controller=1 context=demo mode=immediate

I have a Diva Server BRI installed in a Debian system running Asterisk 1.2.22. When a call comes in the Background application plays a couple of times. Halfway through the second time, the call is disconnected (total time connected about 25 seconds). I've posted various configs below. Any advice on how to debug would be appreciated. We are located in New Zealand. CLI showing capi debug (just

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

Hi, We've got a dedicated T1 with two trunks running into our ILECs selective router for 911. Split out of the T1 into two MF CAMA trunks on ILEC side. I'm trying to use asterisks e911 signaling, but I'm having trouble with the dial command. (== Everyone is busy/congested at this time (1:1/0/0)) I'm missing something and I'm thinking it has to do with the hookstate

Good day all I've installed asterisk with sip on my LAN,no special cards,if done sip.conf and extensions.conf and all work 100,I'm using x-lite as a client. I'm trying to do conferencing.What I did was to has out the meetme.conf looks like [rooms] conf => 9876 conf => 2345,9938 and extension.conf exten => 9876,1,MeetMe,9876 When I go onto x-lite and type 9876 it gives me

Hi all, does anyone have capiECT working? I can't get it to work - please see my logfiles below. While using an * CVS version of late September and chan_capi-0.2.5 (I guess), it worked!!! (I know, never change a running system ... should've backuped ... etc.) Here's the setup: NT----AgfeoISDNPBX----AgfeoISDNPBX----AnalogPhones in between the two PBX I attached my * server with a

Hi all, i have asterisk 1.0.9 with an Eicon Diva 4bri and chan_capi-cm-0.6. I have 2 NTBAs (one with did and one without). When using the one without did, i am able to place outgoing and incoming calls. When i use the NTBAs with did i have a layer 2 error. Anyone an idea? -- Executing Dial("SIP/2004-9634", "CAPI/g1/43XXXXXX") in new stack > data = g1/43XXXXXX

Prior to this change, some screen models would display a 2px-large purple line along left screen border, likely to complain about invalid InfoFrame content. The following bug report seems to trigger this issue: https://bugs.freedesktop.org/show_bug.cgi?id=75203 Running mmiotrace on a GTX 650 (GK107) produces the following trace, regardless of which output is used (HDMI-1, DVI-D-1, DVI-D-2), to