similar to: *** Yet another boring weekend? Test new Asterisk features in development!

Friends, The developer team for Asterisk not only consists of coders - a very important part are the testers, those that test new code and give feedback. For a few weeks, I've been maintaining a large number of branches with various stuff in them and have gotten very little feedback, not enough to judge whether or not to move forward with these patches. Some, but not all, code is

Anyone have any info on the date (or bug tracker number) of the change from incominglimit to call-limit, and is it call_limit or call-limit? Does it work with SIP friends? Running CVS head 8/24 (right before 1.2 release). -------------- next part -------------- An HTML attachment was scrubbed... URL:

Your log file almost looks like a bug in Asterisk doesn't it? Why call poll() with a zero timeout while passing only one FD? and then why do the read when there is no data? Read the man pages for all the system calls Take a look at the source chan_sip.c /* Wait for sched or io */ res = ast_sched_wait(sched); if ((res < 0) || (res > 1000))

Friends, At the first Astricon I was very happy to see Marc Blanchet as one of the attendees. I knew he was one of the IPv6 gurus and wanted someone to show some interest in Asterisk and IPv6. Well, he did not only get interested in it, but started coding on it. The results have been available for quite some time at http://www.asteriskv6.org/ and Marc has tested it at several SIPits for

The Asterisk Development Team has announced the release of Asterisk 1.6.2.18. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.18 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

The Asterisk Development Team has announced the release of Asterisk 1.6.2.18. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.18 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

Actually, I have tested this here with an Aastra 9133i and an Asterisk@Home server, and the 9133i will re-subscribe on its own after an Asterisk reboot, if you wait long enough. It took on the order of an hour to do so. Of course, a phone reboot will get it done faster, if necessary, but it _will_ eventually re-subscribe on its own. > Thanks Olle, > > So am I to understand that you

Trouble getting chan_sip2 to compile below is what I have done -download acl.c.patch,acl.h.patch,chan2s_sip.c to /root/software cp /root/software/chan_sip2s.c /usr/src/asterisk/channels cd /usr/src/asterisk/ patch -p0 acl.c /root/software/acl.c.patch cd /usr/src/asterisk/include/asterisk patch -p0 acl.h /root/software/acl.h.patch - added the follow to /usr/src/asterisk/channels/Makefile

hi All, We are using Asterisk server with sip phones (SJPhone). On the local LAN, when we use the SJPhone as the SIP client, communication works fine with no disturbances and noices. But when it comes to dialup connection we harldy hear anything except a rough noice. We have included G.729 Codec (Annex B) with the Asterisk server, and we added the G.729 Codec to the SJPhone too. But it seems

Hi all, Regular and avid readers of this column will know that Don Driscoll and I have recently posted two messages requesting assistance concerning an apparent failure of "se.contrast" to produce an se for a contrast. So far, an ominous silence rings in our ears, but read on Gentle Reader, and see if even the machinations of "debug" doesn't stimulate you to respond with a

I am trying to use the "shared mem size" parameter in Samba 2.0.4b, but the size reported by smbstatus does not match the value I am using. Without the parameter, the default is 1MB, and the size reported by smbstatus is indeed 1048576. However, when I set the value of "shared mem size" to 5242880 (5MB), smbstatus says 879608. Is smbstatus not reporting the actual size, or

--- "Daniel B. Miller" wrote: It looks to me like patch is trying to put each .cvsignore in the vp32 directory rather than the specified subdirectories such as Codecs/quicktime/. It seems like I need some option to get patch to use the 'Index:' commands. I looked at the man page, it goes on about how patch tries to derive the filenames, but I am still confused. Any help would

I think I'll start with the simple stuff, just to be safe. :-) Here are some cvsignore updates, and a fix for the inter-project dependencies in the MSVC *.dsw file. Cheers, Eric -------------- next part -------------- A non-text attachment was scrubbed... Name: vp32-cvs-ignore-patch.diff Type: application/octet-stream Size: 1524 bytes Desc: vp32-cvs-ignore-patch.diff Url :

Dudes and/or Dudettes: HEEELLLPPP! (such a newbie statement) I can't get my remote SMB mounts to work on my Linux box. duke was happily running Slackware Linux 2.0.36 and Samba 1.9.18p8 . I upgraded to Linux 2.2.0pre9 (2.2.0final), which seemed to require upgrading to Samba 2.0.0 . (I don't remember the exact errors) Now i can't get my NT and '95 disks to stay mounted on

When creating users in the sip.conf file, they do not appear when running the "sip show users" command from the CLI until i restart. A reload doesnt make them appear. As i said, I am new to the whole Asterisk thing, however have worked with IP/SIP PBX's for a few years - its most likely a user problem though! Check it out and let me know what you get. Cheers Chris PS - I would try

Friends in the Asterisk community, I've been talking for years about the new version of the SIP channel. I've been trying to get funding and get going. Well, the funding part remains to be handled, but I have other news - if you kan keep it to yourself. ...I've began coding. Finally. With a happy smile on my face I removed "pedantic=yes" the other day. After years of

<P>folks,<BR><BR>Sorry to disturb you with such a basic question, but I am new to bandwidth control , shaper etc...&nbsp; I tried reading some doc about it, but couldn´t apply them to solve my (following problem):<BR><BR>I have this..<BR><BR>[internet]------[eth0][linux box][eth1]--------[my lan]<BR><BR><BR>ADSL (256k)<BR>eth0:

Hello Asterisk users around the world! Recently, I have been working with pretty large Asterisk installations. 300 servers running Asterisk and Kamailio (OpenSER). Replacing large Nortel systems with just a few tiny boxes and other interesting solutions. Testing has been a large part of these projects. How much can we put into one Asterisk box? Calls per euro invested matters. So far,

Friends in the Asterisk.org community, There is a lot of cool stuff going on in Asterisk development, things that will change Asterisk and make it work better in your organisation, make it easier to sell in your area or give you more consulting oppurtunities - in short, functionality that will make a lot of sense for you users. However, developers can't really get anywhere without a

The Asterisk Development Team has announced the final maintenance release of Asterisk, version 1.6.2.19. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ Please note that Asterisk 1.6.2.19 is the final maintenance release from the 1.6.2 branch. Support for security related issues will continue until April 21, 2012. For more information