I tried to ajust the tx- and rxgain for the sip peer in sip.conf. And
restarted the asterisk. But it takes no effect. Any suggestion?
2011/3/4 Danny Nicholas da...@debsinc.com
Defaults are 0.0 (leave volume unchanged) +values make volume louder, -
softer.
--
This settings are for ISDN configurations I think.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong
Sent: Monday, March 07, 2011 6:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
it should work for sip channel too. I recorded the downlink channel in
wav-format. Does the rx or txgain ajusting only work with alaw or ulaw?
2011/3/7 Faisal Hanif fai...@vopium.com
This settings are for ISDN configurations I think.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
This is a problem in chan_sip.c
After REFER asterisk does not notify dialplan or AGI of REFER.
I've tried to convince asterisk developers this is a problem but they only
offered me 3 solutions:
1. Fix it yourself
2. Pay someone to fix it
3. Try to convince enough people that this is a problem and
You could always just use sox to adjust the levels
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On Thu, 3 Mar 2011 08:42:36 -0600, Danny Nicholas
da...@debsinc.com wrote:
Having traversed this rabbit-hole the answer is that it depends on your
carrier. If they offer call-supervision, asterisk can wait for pickup on
the other side. The resolution I came up with for my offering:
I was going
On Mon, Mar 7, 2011 at 2:11 AM, Stuart Longland redhat...@gentoo.orgwrote:
http://mirror.aarnet.edu.au/pub/gentoo/distfiles/asterisk-1.8.3.tar.gz
I haven't checked that URL, but it should be correct. That, and that
mirror should be unmetered if you're on a university network.
Thanks mate,
Hi,
I am using IAXmodem + hylafax to do outgoing and incoming fax with asterisk.
I wonder how to write a dialplan to differentiate incoming call or fax.
I am sharing a line for both voice and fax.
CK
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-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: Monday, March 07, 2011 8:14 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] [1.4] Forcing Asterisk/Zaptel to wait
On 03/04/2011 12:35 PM, Louis Carreiro wrote:
Ha! Thanks Vip!
Sorry about not including my version numbers too. On my production box I'm
using 1.8.3 (that's the debug from the original email). On my demo box I just
build I'm using 1.8 SVN-trunk-r309404 and that's what generated these logs.
Thanks. This comes really close. My asterisk currently has snmp setup
properly and I can see it shows the output when I do snmpwalk command. I am
stuck at Cacti end. Wondering what to do to setup the asterisk remote end.
The tutorial you provided is for Nagios (which I tend to stay away due to
Thanks for your reply - but I did it a slightly different way:
Nevermind - I've re-written my dialplan so that all subs are in one
context. Now I only need 1 more line of code.
Thanks anyway :)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Kevin,
I had no clue! Thanks for the note! I'll be checking out the 1.8 SVN branch
here shortly then for testing!
Thanks again!
Louis
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming
Sent:
On Mon, 7 Mar 2011 08:20:26 -0600, Danny Nicholas
da...@debsinc.com wrote:
#1 is Lazy notation to say ${EXTEN} starts with SIP (as opposed to Local
or DAHDI)
#2 Might just be a typo on my part. I frequently switch usage between Wait()
and WaitExten().
Thanks for the clarification.
--
On Sat, Mar 05, 2011 at 08:14:37PM +0200, Elliot Murdock wrote:
Hello All,
How does one go about creating a dahdi configuration file for multiple
PRI cards?
1. vi
2. dahdi_genconf handles the common case quite well and will normally be
a good start.
--
Tzafrir Cohen
On Monday 07 March 2011 08:20:26 Danny Nicholas wrote:
On Monday 07 March 2011 08:14:27 Gilles wrote:
1. Why use instead of = to compare the extension with SIP?
exten = s,n,Gotoif($[${EXTEN} SIP]?start)
#1 is Lazy notation to say ${EXTEN} starts with SIP (as opposed to
Local or DAHDI)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: Monday, March 07, 2011 11:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] [1.4] Forcing
Basically each PRI card will be configured as g0, g1 and so on. Try this
link http://www.voip-info.org/wiki/view/Asterisk+PRI
http://www.voip-info.org/wiki/view/Asterisk+PRIif you are using sangoma
cards then try http://wiki.sangoma.com
On Mon, Mar 7, 2011 at 10:00 PM, Tzafrir Cohen
Un-top-posting...
On Sat, Mar 05, 2011 at 08:14:37PM +0200, Elliot Murdock wrote:
How does one go about creating a dahdi configuration file for multiple
PRI cards?
On Mon, Mar 7, 2011 at 10:00 PM, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
1. vi
2. dahdi_genconf handles the common
On Sat, Mar 5, 2011 at 8:54 PM, Pezhman Lali l...@lopl.net wrote:
Dear
this note is only for fresh administrators don't think about asterisk
security.
Do you know where you go to 'un-ban' an IP if they made some mistake?
Using webmin I was not able to find the IP address that was was banned.
Hi,
I installed Free FAX for Asterisk on my home Asterisk 1.8(Centos5.5)
server. Everything seems fine but I just saw this WARNING shows up in
the log every time I start the asterisk:
/[2011-03-07 10:50:41] WARNING[13429] loader.c: Error loading module
'res_fax_digium.so':
I'm using iaxagent on a Droid X to connect by IAX to 1.8.3 at the
office. 1.8.3 has sip phones. The audio is fine on the Droid X side. On
the office side, they hear an echo of _their_ speech, not mine.
The office uses sip-providers generally without any echo problem.
Where do I start to
On 03/07/2011 12:58 PM, Jian Gao wrote:
Hi,
I installed Free FAX for Asterisk on my home Asterisk 1.8(Centos5.5)
server. Everything seems fine but I just saw this WARNING shows up in
the log every time I start the asterisk:
/[2011-03-07 10:50:41] WARNING[13429] loader.c: Error loading module
On 03/07/2011 03:35 PM, RR wrote:
Hello all,
mmm a bit embarrassing about not having a clue as to why we're getting
this error on make of 1.8.3
[AR] hash/hash.o hash/hash_bigkey.o hash/hash_buf.o hash/hash_func.o
hash/hash_log2.o hash/hash_page.o hash/ndbm.o btree/bt_close.o
btree/bt_conv.o
On 03/07/2011 04:15 PM, sean darcy wrote:
I'm using iaxagent on a Droid X to connect by IAX to 1.8.3 at the
office. 1.8.3 has sip phones. The audio is fine on the Droid X side. On
the office side, they hear an echo of _their_ speech, not mine.
The office uses sip-providers generally without any
On Mon, Mar 7, 2011 at 5:25 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 03/07/2011 03:35 PM, RR wrote:
Hello all,
mmm a bit embarrassing about not having a clue as to why we're getting
this error on make of 1.8.3
[AR] hash/hash.o hash/hash_bigkey.o hash/hash_buf.o
On 03/07/2011 04:31 PM, RR wrote:
On Mon, Mar 7, 2011 at 5:25 PM, Kevin P. Fleming kpflem...@digium.com
mailto:kpflem...@digium.com wrote:
Please do not reply directly to posters on the mailing list unless they
request it.
On 03/07/2011 03:35 PM, RR wrote:
Hello all,
On Mon, Mar 7, 2011 at 5:34 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 03/07/2011 04:31 PM, RR wrote:
On Mon, Mar 7, 2011 at 5:25 PM, Kevin P. Fleming kpflem...@digium.com
mailto:kpflem...@digium.com wrote:
Please do not reply directly to posters on the mailing list unless they
On 03/07/2011 04:41 PM, RR wrote:
Someone with SPARC experience will have to chime in then... for some
reason the configure script has determined that your compiler
provides atomic instructions, but they aren't being found at link time.
Ok...thanks. Is there no way for me to tell
Hello all,
Figured I'd repost this with an edited subject line, to attract attention of
people with Debian On Sparc experience. Apologies in advance if this kind of
thing is frowned upon :)
[AR] hash/hash.o hash/hash_bigkey.o hash/hash_buf.o hash/hash_func.o
hash/hash_log2.o hash/hash_page.o
Okay, so here's the configuration I have for MySQL Realtime (Asterisk
version 1.6.2.17):
In /etc/asterisk/extconfig.conf:
sipusers = mysql,mya2billing,cc_sip_buddies
In /etc/asterisk/res_mysql.conf:
[mya2billing]
dbhost = localhost
dbname = mya2billing
dbuser = a2billinguser
dbpass = REDACTED
On Mon, Mar 7, 2011 at 5:52 PM, Ernie Dunbar maill...@lightspeed.ca wrote:
Okay, so here's the configuration I have for MySQL Realtime (Asterisk
version 1.6.2.17):
In /etc/asterisk/extconfig.conf:
sipusers = mysql,mya2billing,cc_sip_buddies
In /etc/asterisk/res_mysql.conf:
Don't know
On Mon, Mar 7, 2011 at 4:52 PM, Ernie Dunbar maill...@lightspeed.ca wrote:
[mya2billing]
dbhost = localhost
dbname = mya2billing
dbuser = a2billinguser
dbpass = REDACTED
dbport = 3306
Try adding dbsock = /var/lib/mysql/mysql.sock to the end of this config
stanza and see if that helps (or
On 03/08/11 08:49, RR wrote:
Any idea where this is coming from? seems like something is selected
that doesn't have other related stuff unselected? no clue where to start
looking
No SPARC expert, but I seem to recall the lowest-common-denominator
SPARCs lack things like hardware multiply in
Hello Stuart
On Mon, Mar 7, 2011 at 6:09 PM, Stuart Longland redhat...@gentoo.orgwrote:
On 03/08/11 08:49, RR wrote:
Any idea where this is coming from? seems like something is selected
that doesn't have other related stuff unselected? no clue where to start
looking
No SPARC expert, but
On Mon, Mar 07, 2011 at 10:16:52AM -0800, Steve Edwards wrote:
Un-top-posting...
On Sat, Mar 05, 2011 at 08:14:37PM +0200, Elliot Murdock wrote:
How does one go about creating a dahdi configuration file for
multiple PRI cards?
On Mon, Mar 7, 2011 at 10:00 PM, Tzafrir Cohen
On 03/08/11 09:21, RR wrote:
Hello Stuart
On Mon, Mar 7, 2011 at 6:09 PM, Stuart Longland redhat...@gentoo.org
mailto:redhat...@gentoo.org wrote:
Even if it doesn't help fix the problem, you probably will want to use
at least -mcpu=v9 (educated guess looking at the gcc manpage) if
On Mon, Mar 7, 2011 at 4:52 PM, Ernie Dunbar maill...@lightspeed.ca
wrote:
[mya2billing]
dbhost = localhost
dbname = mya2billing
dbuser = a2billinguser
dbpass = REDACTED
dbport = 3306
Try adding dbsock = /var/lib/mysql/mysql.sock to the end of this config
stanza and see if that helps
On Mon, Mar 7, 2011 at 6:31 PM, Stuart Longland redhat...@gentoo.orgwrote:
On 03/08/11 09:21, RR wrote:
Hello Stuart
On Mon, Mar 7, 2011 at 6:09 PM, Stuart Longland redhat...@gentoo.org
mailto:redhat...@gentoo.org wrote:
Even if it doesn't help fix the problem, you probably will
I'm using iaxagent on a Droid X to connect by IAX to 1.8.3 at the
office. 1.8.3 has sip phones. The audio is fine on the Droid X side. On
the office side, they hear an echo of _their_ speech, not mine.
The office uses sip-providers generally without any echo problem.
Where do I start to
http://www.disruptivetelephony.com/2011/03/google-voice-now-offers-sip-addresses-for-calling-directly-over-ip.htmlNice ;)
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On 03/07/2011 05:26 PM, Kevin P. Fleming wrote:
On 03/07/2011 04:15 PM, sean darcy wrote:
I'm using iaxagent on a Droid X to connect by IAX to 1.8.3 at the
office. 1.8.3 has sip phones. The audio is fine on the Droid X side. On
the office side, they hear an echo of _their_ speech, not mine.
Hello,
I have just installed an Asterisk server with a Digium TDM410P card with 3
FXO modules (no module in the 4th slot).
It's lived on two different machines (a test machine, which had Linux kernel
2.6.28, and a new dedicated machine which has Linux kernel 2.6.32).
On the test machine
On Mon, Mar 7, 2011 at 5:45 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 03/07/2011 04:41 PM, RR wrote:
Someone with SPARC experience will have to chime in then... for some
reason the configure script has determined that your compiler
provides atomic instructions, but they
On Mon, Mar 7, 2011 at 9:15 AM, Jamie A. Stapleton
jstaple...@computer-business.com wrote:
iptables -L -v
will give you the IP address that was banned
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
On Mon, Mar 7, 2011 at 6:48 PM, RR ranjt...@gmail.com wrote:
On Mon, Mar 7, 2011 at 6:31 PM, Stuart Longland redhat...@gentoo.orgwrote:
On 03/08/11 09:21, RR wrote:
Hello Stuart
On Mon, Mar 7, 2011 at 6:09 PM, Stuart Longland redhat...@gentoo.org
mailto:redhat...@gentoo.org wrote:
Dean,
Thank you for great news. Let us see how the second SIP GV incarnation
survives.
-Vladimir
On 3/7/2011 6:51 PM, Dean Collins wrote:
http://www.disruptivetelephony.com/2011/03/google-voice-now-offers-sip-addresses-for-calling-directly-over-ip.html
Nice ;)
--
On Tue, Mar 8, 2011 at 1:51 AM, Dean Collins d...@cognation.net wrote:
http://www.disruptivetelephony.com/2011/03/google-voice-now-offers-sip-addresses-for-calling-directly-over-ip.html
Nice ;)
Hi Dean,
What I'm waiting for is when you can send GV calls to a SIP URI
without all the gymnastics
On Mon, 7 Mar 2011 08:50:27 -1000, Matt Darnell
mattdarn...@gmail.com wrote:
On Sat, Mar 5, 2011 at 8:54 PM, Pezhman Lali l...@lopl.net wrote:
Dear
this note is only for fresh administrators don't think about asterisk
security.
Do you know where you go to 'un-ban' an IP if they made some
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