similar to: No D-channels available! Using Primary on channel 16 anyway!

---------- Forwarded message ---------- From: David Sarmiento Q. <dorphalsig@gmail.com> Date: Jan 8, 2006 3:49 PM Subject: Problems with R2 Support To: asterisk-users@digium.com Hello! I've been trying to compile R2 support into * 1.2 for two days now, and havent been able to do so. I am using: * spandsp 0.0.2pre21c * libsupertone 0.0.2 * libmfcr2 0.02 * libunicall 0.0.2e I compile

I would just like to get a refresh of the situation for Australian users. It would seem that the TE400P is currently available and is likely to acheive approval for use in Australia within the next 2 months ? (when is the end of summer?). Once this is done, it will certainly suit larger installations, but it still leaves a number of 'gaps'. Anyone with analog phone lines will need

Hi, Is there a simply way to allow dialout from ISDN modem to outside number through Asterisk? I've got an server with an Asterisk and the following cards: 1. TE110 -- to telco 2. TE400P with one FXS to analog phone 3. Two HFC-S based cards in NT mode I'd like to connect ISDN modem to one HFC-S card and allow dialout through TE100 to some external number (being more precisely, to some

I have recently installed Asterisk Now Beta 4 (32-bit) with three Digium cards - 2 Wildcards TE400P (one with 4 FXS modules and one with 4 FXO modules) and a TE205P dual E1/T1 card. The problem is with the TE205P and configuring the /etc/zaptel.conf and /etc/asterisk/zapata.conf files. I go in and add the required lines to configure the TE205P and restart Asterisk. All is fine until I reboot the

Hi, im not sure if this is a mongrel or rails question, so im sorry if I got the wrong list. Is there anyway to turn off logging (in production.log) if a request comes from a given IP address. I use monit to watch everything, and I get tons of this in my log file: Processing AccountController#login (for 127.0.0.1 at 2007-07-31 14:06:28) [GET] Session ID: XXXXXXXXXXXXXXXXXX

I''m using monit to restart mongrel processes if they go away. With mongrel 0.3 I could issue a start command and the process would start up even if the .pid file existed (though a warning would appear). Now, with mongrel 1.0.1 the warning says I must clear the .pid file and the process is not started. Is there anyway to force it to start without having to clear the .pid? Regards

I spent quite a bit of time setting up what I wanted to update and install via yumex. I left my system for a few minutes to attend to another computer. A helpful family member logged me off without checking. So can I recapture what I had queued? I cannot find anything that looks like a yumex queue file. yum.log does not have any updates from today.... :(

Tried to convert a 60 gig ext2 disk to ext3. Used tune2fs version 1.22 (it said). The filesystem refused to be converted with that version. Upgraded to 1.25 of e2fsprogs and the problem went away. My two cents. /Benjamin

Signed-off-by: Dor Laor <dor.laor at qumranet.com> --- external-module-compat.h | 1 + hack-module.awk | 7 +++++++ 2 files changed, 8 insertions(+), 0 deletions(-) diff --git a/external-module-compat.h b/external-module-compat.h index 9c93464..ff2f4ca 100644 --- a/external-module-compat.h +++ b/external-module-compat.h @@ -79,6 +79,7 @@ do { \ #define

Hi All; My ISDN was working fine, and suddenly I start getting the below: sig_pri.c:985 pri_find_dchan: Span 1: No D-chanannel anyway! There is a Yellow Alarm, so what it could be the problem? My configuration as following: system.conf span=1,1,0,ccs,hdb3,yellow bchan=1-15,17-31 dchan=16 chan_dahi.conf context=IncomingPSTN group=0 signalling=pri_cpe switchtype=euroisdn

Hi. I am using latest asterisk 1.2 and it would be nice in a meetme conference to be able to mute all but a particular user and then unmute all those users again with one command. Am I missing something or is this not available? Maybe I could write something, but I wanted to check first. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it?

Hi all. I have this problem with my Digium 2E1 card and PRI, for hours It works well, with some meesages... [Aug 10 09:20:31] NOTICE[32270] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 But PRI continue up....hours later... PRI go down. I thought the problem was in the telco, but the strange thing is that I have a loop cable in the second E1 and when I scan both E1

Figure maybe someone on this list would know because I imagine people use faster than gigabit ethernet in xen solutions. I picked up two "Flextronics 24-port 4X 1U SDR InfiniBand Switch" for $285 and $340 or (so about $600 for the two shipped) used on ebay. Everywhere I look I see them listed $2,500 new. I have split emotions on this. I already got one of them, it looked primo

>> >> I highly recommend the following site: I intentionally deleted the dangerous link that was previously posted here. >Sure. >And take the next time more time to explain why you recommend it. >Mostly because to get beyond a moderator of this mailinglist. > >In other words: >Stappers considered the blut URL dropping as spam. >He flagged the posting E-mail

virsh provides net-start <netname-or-id> to activate an inactive network. Does virsh provide any way to do the exact converse? deactivate an active network without destroying, undefining, removing, ... Also - I am curious - what does activate actually do? What change in state of anything occurs? ( I read the document at http://libvirt.org/formatnetwork.html which

How do you retreive the caller id on incoming analog lines and display the id on the sip phones on the network? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060201/076073d1/attachment.htm

I am using an Asterisk box as a mini-softswitch and have run into a minor (hopefully) road block. The far end switch requires CIC (Carrier Identification Code) in the SIP invite like: INVITE sip:+18001234567;cic=+16789@hostname.com;user=phone SIP/2.0 ^^^^^^^^^^^^^^^^^^^^^^^ Is there a way to configure Asterisk to send this in the SIP invite? Any help would be

Hello, On 3/22/06, Dovid Bender <asteriskdigium@yahoo.com> wrote: > I have a rule in my outlook to delete any email that > in the subject it says "asterisk users mailing list > traffic". Get my drift ? This topic has been around > for a long time as others have mentioned and people > keep replying. This useless topic alone adds several > messages to my inbox

Hi all, I am trying to find a way to set the maximum wait time for a user in MeetMe. I am creating a dynamic conference. If there's only 1 user in the conference room, I would like to allow him to wait only until the certain time. eg. may 1 min. If there's no other user joining the conference, I'll hang up his call. Is there any technique to enable this?? Pim

Just saw from build 2036: Starting mini_httpd... WARNING WARNING WARNING YOU STILL HAVE NOT CHANGED YOUR HTTPS ADMIN PASSWORD ANYONE THAT KNOWS YOU ARE USING ASTLINUX CAN DESTROY YOUR SYSTEM. PLEASE CHANGE THIS OR DISABLE THE HTTPS ADMIN INTERFACE IMMEDIATELY! Example: htpasswd /var/www/admin/.htpasswd admin WARNING WARNING WARNING Starting mini_httpd (HTTP only)... cat: can't open