BJ Weschke wrote:
On 2/2/07, Thomas Winter [EMAIL PROTECTED] wrote:
Hi,
I have an queue stored in relatime and defined members called through
LOCAL/
I found out that if I call the members through the LOCAL think the queue
statistics is not updated.
Any idea, or isnt possible to call
I hate to think of the possible echo if you change the volume and a
sip device didn't know about it. It wouldn't effectively use the echo
cancellation on board, I am not an expert with echo cancellation
however.
It should be possible to just multiply each byte of a waveform by a
percentage if
Thanks, it really was easy.
Unfortunately it only works for MSN's, not for the base number. Oh
well, I'll just stop using the base number, I've got enough MSN's anyway.
Thanks again.
Armin Schindler wrote:
On Thu, 1 Feb 2007, Cosmin Prund wrote:
Any ideas? It should be simple...
What type of line is that? The 'base number' is also a MSN on lines I know.
Or is it PtP with DID?
Armin
On Sat, 3 Feb 2007, Cosmin Prund wrote:
Thanks, it really was easy.
Unfortunately it only works for MSN's, not for the base number. Oh well,
I'll just stop using the base number, I've
I'm still using asterisk 1.0.x bristuffed at one site.. Is there
anything similar for this? When both channels are in use, 3rd call
doesn't recive busy signal, but a message fromt he TelCo (something
like The dialed number is not currently available).
Thanks,
Julián J. M.
On 2/3/07, Armin
On Sat, 3 Feb 2007, Julian J. M. wrote:
I'm still using asterisk 1.0.x bristuffed at one site.. Is there
anything similar for this? When both channels are in use, 3rd call
doesn't recive busy signal, but a message fromt he TelCo (something
like The dialed number is not currently available).
I don't use chan_capi, but bristuff. http://www.junghanns.net/en/download.html
Julian.
On 2/3/07, Armin Schindler [EMAIL PROTECTED] wrote:
On Sat, 3 Feb 2007, Julian J. M. wrote:
I'm still using asterisk 1.0.x bristuffed at one site.. Is there
anything similar for this? When both channels
On Fri, 2 Feb 2007, Remco Barendse wrote:
Would you be willing to share your blacklist for the kernel modules?
Have you considered compiling a custom kernel for your hardware rather
than not loading modules? It's something I've always done from day 1 with
Linux (some 10 years back now!)
Hi,
maybe I am a little bit late with this answer.
I take a look at Your config and the debug output.
snip
Provider --te11xp--- asterisk ---te11xp-- nortel merridian
option 11c
snip
zapata.conf
-
context=from-pstn
switchtype=dms100
Hi
i am upgrading an asterisk server from 1.2.4 to 1.4.
i've installed libpri 1.4
i've installed zaptel 1.4
I've installed the new version of misdn with the script of beronet.
i use this configure script:
./configure --with-postgres=/usr/local/pgsql
then:
make menuselect
[*] 1. cdr_csv
[*]
On 2/3/07, nik600 [EMAIL PROTECTED] wrote:
Hi
i am upgrading an asterisk server from 1.2.4 to 1.4.
i've installed libpri 1.4
i've installed zaptel 1.4
I've installed the new version of misdn with the script of beronet.
i use this configure script:
./configure --with-postgres=/usr/local/pgsql
Hi,
im trying to connect 2 asterisk servers. my server which registers to the
main server keeps on displaying this message:
[Feb 3 20:04:57] NOTICE[10112]: chan_sip.c:7085 sip_reg_timeout:--
Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #84)
why is it not
Hi,
On Fryday, 2007-02-02 François Delawarde wrote :
Don't you think it could be an interesting feature in Asterisk? It
already does transcoding, why not gain when voice flow passes through it?
François.
On a SIP-to-SIP-Call Asterisk is not neccessarily in the voice flow,
so this does not
On 1/31/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Asterisk version 1.4
Queue(queuename[|options[|URL][|announceoverride][|timeout][|AGI])
The optional AGI parameter will setup an AGI script to be executed on the
calling party's channel once they are connected to a queue member.
Ceará
On Sat, 3 Feb 2007, Gordon Henderson wrote:
On Fri, 2 Feb 2007, Remco Barendse wrote:
Would you be willing to share your blacklist for the kernel modules?
Have you considered compiling a custom kernel for your hardware rather than
not loading modules? It's something I've always done from
Armin Schindler wrote:
What type of line is that? The 'base number' is also a MSN on lines I know.
Or is it PtP with DID?
Armin
On Sat, 3 Feb 2007, Cosmin Prund wrote:
The base number works like any other MSN most of the times, but the
busy application doesn't work on it. If I dial the
On Sat, Feb 03, 2007 at 05:52:51PM +0200, Cosmin Prund wrote:
Armin Schindler wrote:
What type of line is that? The 'base number' is also a MSN on lines I
know. Or is it PtP with DID?
Armin
On Sat, 3 Feb 2007, Cosmin Prund wrote:
The base number works like any other MSN most of the
The following strange conditions is happening while I try to dial a
SIP user from another SIp user.
SIP to Zap dialing is fine, as all 4 users can call PSTN.
I'm using Asterisk SVN-branch-1.2-r51359M
Example: extension 3210 calls extension 3213. They are all registered properly:
chrom01*CLI sip
Tzafrir Cohen wrote:
Do you use Busy to send a bus signal to the other party?
I use Busy. I have no idea how it works. When I call from my mobile
phone to my PBX I get a busy signal and it seems I'm not being charged
for the call (so it's not like * opened up the line and played the
while compiling svn 53132 of asterisk branch 1.2
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT
-D_GNU_SOURCE -O6 -march=i586 -DZAPTEL_OPTIMIZATIONS
-DBUSYDETECT_MARTIN -fomit-frame-pointer -fPIC -c -o app_sms.o
Leo Ann Boon wrote:
Good question. Anyone knows if the TDM-400 actually detect loop drops?
Well, that's really what kewlstart (and loopstart) means. If it
couldn't, then Asterisk wouldn't know that the call had been hung up,
and hog the channel.
For loopstart lines, I don't think
On Sat, 3 Feb 2007, Cosmin Prund wrote:
Tzafrir Cohen wrote:
Do you use Busy to send a bus signal to the other party?
I use Busy. I have no idea how it works. When I call from my mobile phone to
my PBX I get a busy signal and it seems I'm not being charged for the call (so
it's not
That depends on your distro. I have tested * with Beronet cards on OpenSuSE
10, Debain Sarge and Ubuntu Edgy. What has to be blacklisted is every
remainder of old ISDN stuff and hotplug modules (*php = * pci hot plug). As
far as I know these cards are not hotpluggable at all and who wants to
Hello all,
I am having trouble getting gtalk to work with my account which is not
using a gmail.com email address. When I do this there an error from the
Jabber module:
[Feb 3 20:51:17] ERROR[6286]: res_jabber.c:573 aji_act_hook: JABBER:
Node Error
[Feb 3 20:51:17] WARNING[6286]:
Stephen Bosch wrote:
The reason we have these complaints is not because Asterisk doesn't
detect the drop -- it's because a great many telephone companies don't
do remote party disconnect signalling, or they don't do it properly.
When people call for technical assistance they usually end up
You need to at least register AFAIK. Download gaim and use its
facilities to rejister. Jabber is not for the faint of heart when it
comes to IM platforms, read up on it if you haven't already.
On 2/3/07, Ian Hailey [EMAIL PROTECTED] wrote:
Hello all,
I am having trouble getting gtalk to work
The fact that all of the phones have the same 'host' is not a good sign.
Also - turn 'qualify' on. It really helps with phone status.
PaulH
On Sat, 2007-02-03 at 12:50 -0500, Erick Perez wrote:
The following strange conditions is happening while I try to dial a
SIP user from another SIp
No such host: 3213)
Look for an extra closing parenthesis in your Dial command:
Dial(SIP/3210-084eaa80, SIP/3213)|30|to)
It should be SIP/3213 rather than SIP/3213).
--Luki
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asterisk-users
Hi, first time poster. I've searched, but find very little on this topic.
My church has only 2 incoming phone lines (copper), being serviced by
Digum TDM400 card. The existng phone system (being replaced) was a key
system, easy for users to tell both lines were in use. We have chosen
hi,
i think the problem is here :
exten = _321[0123],n,Dial(SIP/${EXTEN}),30,to)
|
replace with
exten = _321[0123],n,Dial(SIP/${EXTEN},30,to)
note, i removed the parenthesis ')' after the {EXTEN}
this should do
regards,
jacobson
---
Scarlet ONE -
Hello all,
Witch snmp system do you use to collect info about their asterisk
boxes, for example, uptime, downtime, max load, HD, free memory,
asterisk status, ,etc?
I use Nagios and the extension that logs in to the * manager interface.
On 2/3/07, Jim Karen Ostrosky [EMAIL PROTECTED] wrote:
Hi, first time poster. I've searched, but find very little on this topic.
Welcome!
What I'd really like to do - for now - is to take the hint, which is
currently assigned to the specific Zap channel, and somehow have it
indicate that
Can I disconnect an arbitrary call using a console command? I remember
reading something but can't find any more.
Yuan Liu
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There are several possibilities, however the one that works across just
about every channel type is soft hangup channel
Yuan LIU wrote:
Can I disconnect an arbitrary call using a console command? I
remember reading something but can't find any more.
same with branch revision 53142
On 2/3/07, Erick Perez [EMAIL PROTECTED] wrote:
while compiling svn 53132 of asterisk branch 1.2
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT
-D_GNU_SOURCE -O6 -march=i586
Stop now?
PaulH
On Sat, 2007-02-03 at 21:47 -0800, Yuan LIU wrote:
Can I disconnect an arbitrary call using a console command? I remember
reading something but can't find any more.
Yuan Liu
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