similar to: Problem with R2 signaling

Hello, The annoying one for two questions: 1) when I recover messages, * jumps of extension.conf toward Voicemail.conf and access to some options default of *. The question is: Can I modify those options? Where ? How? 2) If I want to make a change of password in extension.conf like it would be if I don't possess external BD? [ Authenticate] 7500 =>

hi! What's the libr2 status for Asterisk ? I've got R2 E1 delivered to my * box. I have TE410P digium quad card with newest CVS. How much % is completed with libr2 ? what's completed ? & What's missing ? Thanks, bit123.

I just came across libr2 anyone using it in its current state. Specifically someone from India or around India using it. Also does it work with the digium e1 cards or only the Dialogic cards. http://digium-cvs.netmonks.ca/viewcvs.cgi/libr2/ -- regards Vikram (http://www.vicramresearch.com)

hi! What's the libr2 status for Asterisk ? I've got R2 E1 delivered to my * box. I have TE410P digium quad card with newest CVS. How much % is completed with libr2 ? what's completed ? & What's missing ? Thanks, bit123. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello All I need to use Asterisk with an E1 sangoma card with CAS R2 signalling for Vietnam what is difference between libr2 of CVS and libmfc2 of soft-switch.org ? how to compile chan_unicall.c on asterisk. asterisk update CVS-head- May 27 2005. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi to all. I've put on line a cvs viewer for asterisk source code. Is based onto the suite horde+chora. The website is http://asterisk.espia-net.net The cvs modules shown are * asterisk * asterisk-addons * zaptel * zapata * libpri * libr2 * libiax * libiax2 * gnophone * phpconfig * gastman all revisions, branch , comments & whatever cvs is has been preserved. this could be a sort of

Anyone using the recently MAC OS X ? Version of asterisk ? Thanks, Francisco Perez-Landaeta > From: asterisk-users-request@lists.digium.com > Reply-To: asterisk-users@lists.digium.com > Date: Fri, 27 Aug 2004 13:08:24 -0500 (CDT) > To: asterisk-users@lists.digium.com > Subject: Asterisk-Users Digest, Vol 1, Issue 5082 > > Send Asterisk-Users mailing list submissions to >

Hi folks, where can I find the R2 beta code for Asterisk? Best, PauloHM -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030806/9c7a0660/attachment.htm

https://bugzilla.samba.org/show_bug.cgi?id=2913 ------- Additional Comments From wayned@samba.org 2005-07-26 09:27 ------- Created an attachment (id=1336) --> (https://bugzilla.samba.org/attachment.cgi?id=1336&action=view) Use lchmod and lutimes, if present Please test this patch and see if it works for you. You will need to run "autoconf; autoheader; make proto" (or

https://bugzilla.samba.org/show_bug.cgi?id=2913 Summary: rsync of symlinks on >=bsd needs lutimes(2) and lchmod(2) Product: rsync Version: 2.6.5 Platform: All OS/Version: FreeBSD Status: NEW Severity: normal Priority: P3 Component: core AssignedTo: wayned@samba.org

Hi, How would one activate/deactivate hardware echo cancellation on the TE406 card? Can it be done per channel? I'm going to run TE406 in the following scenario: ISDN -> TE406 -> PABX I understand from Steve Underwood's site that echo cancellation is not good for faxes (and they do that themselves). So what I want to do and bypass echo cancellation for selected extensions before

Hello, I have a busy problem with Asterisk when I try to transfer a call from PRI directly to IVR. This problem appear sometime after 2 hours or 2 minutes. The log file contain : Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) When this problem appear I must restart Asterisk to solve it. Another thing, I don't know why the alarm is set to NOP on SPAN

I have a two port TE205P Digium card. I have set everything up to create a native zap bridge between the two spans. Everything works perfectly except one thing. Our telco has a "password" that has to be entered as soon as a long distance call is made. So if I dial a long distance call from my meridian system, asterisk bridges the call between two channels, my telco picks up and gives

Hi below are my configs: pstn(e1)--->asterisk (span1)----->legacy pbx(connected via span2)-----> legacy pbx analog extensions. my dial plan is like callers dial into asterisk(span1) , hear an IVR option and they are connected to the agents via the legacy pbx (which is in sync with asterisk on span2)....This works perfectly fine until about 200 calls or so...After that time when asterisk

hello, I have been searching the mailing list for info on R2 signalling support in * The best I came up with was a post by Steve as far back as 18 Jul 2003. This post indicates that some work was being done on it. Has anyone got any info if the current cvs does have support for R2 on E100P as yet? Thank you. Anthony

asterisk-users Any help will be appreciated. This card did not connect with E1 line how to loop E400P card to test ? now I loop the card. span 1 ---span2 RJ45 pins 1--4 2--5 but show : When calling ,showing error: app_dial.c:764 dial_exec: Unable to create channel of type 'Zap' Asterisk Ready. *CLI> -- Registered SIP '2002' at 192.168.139.59 port 3289 expires 120

Hi I have upgraded my Dom0 from lenny to sqeeze and xen to 4.0 after that there were some errors when i tried to reboot, but i could fix most of them my dom0 is running fine now, but when i try to start a guest, i get an error: darwin:/# xm create infopool.cfg Using config file "/etc/xen/infopool.cfg". Error: Device 0 (vif) could not be connected. Hotplug scripts not working. i

Hi, I was playing with the munin plugin in nsd4 beta4, and saw some strange errors. Directly after starting nsd on linux, I'm seeing: $ ps ax -o pid,ppid,user,args | grep nsd 1638 1 nsd /usr/sbin/nsd -c /etc/nsd/nsd.conf 1641 1638 nsd /usr/sbin/nsd -c /etc/nsd/nsd.conf 1647 1641 nsd /usr/sbin/nsd -c /etc/nsd/nsd.conf $ sudo munin-run nsd_munin_memory

Hi, I am having issues sending and receiving fax on my asterisk setup. Currently I have a server that has 2 x E1 TDM cards one is sangoma and the other one is openvox. Both support echo cancellation. One of the e1 is connected to our telco provider via mfcr2 where all our incoming calls originate. On the other end is a pri connection going to HICOM PABX where the local attached to a fax is