I am using asterisk 1.8 from a few days ago and it goes into some kind
of loop after maybe a couple of days of use. I compiled with debugging
flags on and no optimize and I then attached gdb to the process.
I did a backtrace and one for all threads and put it at
http://pastebin.com/AGDsLdr7 .
Hi,
it is a known problem, one of the worst. To avoid it:
- do not use urls, only ip addresses in sip.conf
or put your urls inside /etc/hosts (is what I do especially sip
providers urls)
or install a dns-cache on your pbx (maybe the best solution)
Giorgio
On 05/30/2011 03:10 AM, nhadie
On 05/30/2011 02:44 AM, gincantalupo wrote:
- do not use urls, only ip addresses in sip.conf
or put your urls inside /etc/hosts (is what I do especially sip
providers urls)
Definitely don't put URLs in /etc/hosts. I assume you meant URIs, but
either way, neither one belongs there. That
Hello List,
What version of DAHDi should be installed for CentOS Kernel version
2.16.18-194.el5.
We do not have access to yum in our network, so we need to install a specific
version with respect to kernel version.
Or, what update to be downloaded and applied to CentOS kernel to install a
Yes Riddell, I am not aware of TFM.
But all you guys give me lot of information but my question . thanks for
all.
On Mon, May 30, 2011 at 9:22 AM, Matt Riddell li...@venturevoip.com wrote:
On 27/05/11 1:08 PM, Cobra 2 wrote:
I was trying really hard to not say RTFM.
Some people might not
Thanks a lot all,
Now my view is clear ...
On Sun, May 29, 2011 at 3:15 PM, Gordon Henderson
gordon+aster...@drogon.net wrote:
On Sun, 29 May 2011, virendra bhati wrote:
Hi List,
I have stupid question but I want to know it. Why we use the PRI insted of
BRI ? Just for the sake of number
While playing with DB function in Dialplan, I have added some garbage in
AstDB. These are some family names with space in them.
See this,
demo*CLI database show
/18-05-2011 00:00:0052011175221575/TESTDATE: 2011-05-14 21:33:46
/18-05-2011 00:00:0052011175221575/TEST1 : 410
/18-05-2011
On 05/29/2011 04:57 AM, virendra bhati wrote:
And why SIP is used for making calls rather then IAX? Even we know
IAX takes 1 channel for making calls?
IAX is an Asterisk-centered phenomenon and has no currency outside of
it. In the broader VoIP world, interoperability with a wide range of
Hi Satish
Try to do something like this way
CLI database deltree 18-05-2011 00:00:0052011175221575 TESTDATE
I have done like this way hope it works for you.
--
Regards,
Chandrakant Solanki
On Mon, May 30, 2011 at 2:53 PM, Satish Barot satish4aster...@gmail.comwrote:
While playing with DB
Thanks Chandrakant,
It worked!
[SATISH]
On Mon, May 30, 2011 at 3:21 PM, Chandrakant Solanki
solanki.chandrak...@gmail.com wrote:
Hi Satish
Try to do something like this way
CLI database deltree 18-05-2011 00:00:0052011175221575 TESTDATE
I have done like this way hope it works for you.
Thank you for the information. I will try to install a dns-cache.
Regards,
Ron
On 5/30/11, Alex Balashov abalas...@evaristesys.com wrote:
On 05/30/2011 02:44 AM, gincantalupo wrote:
- do not use urls, only ip addresses in sip.conf
or put your urls inside /etc/hosts (is what I do especially
Hi Rajib,
Comments inline.
On 05/30/2011 10:03 AM, Deka, Rajib IN MAA SL wrote:
Hello List,
What version of DAHDi should be installed for CentOS Kernel version
2.16.18–194.el5.
I would use the latest DAHDI version which is currently:
DAHDI tools: 2.4.1
DAHDI linux: 2.4.1.2
You can find
On Sun, May 29, 2011 at 3:18 PM, Ian S. Worthington
ianworthing...@usa.net wrote:
And f/w POS3-07-4-00
That is strange that Asterisk is not sending anything back in response
to the register. Have you looked at the Asterisk console or logs to
see why it is rejecting the register. You might have
Hi List,
Asterisk 's *ControlPlayback* will used for play any recorded file as an
audio player. Is it possible that we can use it for multiple forward and
rewind ?
ex:-
original: ControlPlayback(filename,skipms,ff,rew,stop,pause)
expected
i implemented voicemail with ODBC.
I can now setup mailboxes in voicemail table
and can save recordings in voiemailmessages table in blog field.
Still have issues with join in voicemails and cdrs.
How can we identify the voicemail with corresponding cdr?
Has any one tested?
On Fri, May 27,
On Mon, May 30, 2011 at 02:29:37PM +0200, Patrick Lists wrote:
On 05/30/2011 10:03 AM, Deka, Rajib IN MAA SL wrote:
Hello List,
What version of DAHDi should be installed for CentOS Kernel version
2.16.18–194.el5.
I would use the latest DAHDI version which is currently:
DAHDI tools: 2.4.1
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roger Burton
West
Sent: 28 May, 2011 23:50
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Audio dropping
On Fri, May 27, 2011 at 10:31:57AM
Hello list
i have configured astersik 1.4 with sip i have a question
when i put in dial plan.conf
exten = _0678922645.,1,Set(CALLERID(number)=520460587)
exten = _0678922645
.,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = _0678922645
Did you try different number in place of 5? I meant 1 2 etc..
Also check cli logs on console
Are you dialing from softphone or hardphone because some phone has
dialing regex for security.
--
Sent from my iPhone
On May 30, 2011, at 1:30 PM, salaheddine elharit salah.elharit...@gmail.com
On Mon, 30 May 2011, salaheddine elharit wrote:
exten = _0678922645.,1,Set(CALLERID(number)=520460587)
exten = _0678922645,2,Hangup()
A better subject will get better responses.
Just a quick glance shows that you either mistyped your dial plan or you
need to read up on dial plan pattern
Remove the trailing period after the 5 if that's your whole number
-Original Message-
From: Satish Patel satish...@hotmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Mon, 30 May 2011 14:09:56
To: Asterisk Users Mailing List - Non-Commercial
Ah-ha! Progress at last.
(I'd actually tried debug mode before and wondered why I got no output. Any
harm in leaving that console = etc enabled?)
Console is showing the following. Looks like it doesn't like the format of the
REGISTER message???
--- SIP read from UDP:192.168.1.114:5060 ---
On Mon, May 30, 2011 at 2:45 PM, Ian S. Worthington
ianworthing...@usa.net wrote:
Console is showing the following. Looks like it doesn't like the format of the
REGISTER message???
--- SIP read from UDP:192.168.1.114:5060 ---
REGISTER sip:192.168.1.41 SIP/2.0
Via: SIP/2.0/UDP
Many thanks for that.
I tried pedantic=no (adding it directly to the [702] section in
sip_additional.conf: I'm using the freepbx frontend and it doesn't seem to
have a way to enter that through the gui), but it didn't fix it: same console
log.
Where might I find a reliable source for f/w 8.12?
On Mon, 2011-05-30 at 13:57 +0530, virendra bhati wrote:
Thanks a lot all,
Now my view is clear ...
On Sun, May 29, 2011 at 3:15 PM, Gordon Henderson gordon
+aster...@drogon.net wrote:
On Sun, 29 May 2011, virendra bhati wrote:
Hi List,
On Mon, May 30, 2011 at 5:18 PM, Ian S. Worthington
ianworthing...@usa.net wrote:
Many thanks for that.
I tried pedantic=no (adding it directly to the [702] section in
sip_additional.conf: I'm using the freepbx frontend and it doesn't seem to
have a way to enter that through the gui), but it
Hi All;
From the CLI, if I typed pri then I can find the command and the relative
commands for it .. does this mean that the libpri is installed well? How can I
be sure that Asterisk took the libpri and it is functioning?
Now, regarding to the PRI configurations:
The provider is using: ISDN
By the way, is this only an issue for asterisk 1.4? or is it the same with
1.6 and/or 1.8?
TIA.
Regards,
Ron
On Mon, May 30, 2011 at 2:50 PM, Alex Balashov abalas...@evaristesys.comwrote:
On 05/30/2011 02:44 AM, gincantalupo wrote:
- do not use urls, only ip addresses in sip.conf
or put
Sincere thanks Ryan: all is working at long last.
I risked the f/w upgrade path in the end rather then something which will be
blown away at the next upgrade and leave me scratch me noggin in confusion.
Couldn't have done it without your insight. Thanks again.
i
-- Original Message --
On Mon, May 30, 2011 at 2:44 AM, gincantalupo
gincantal...@fgasoftware.comwrote:
Hi,
it is a known problem, one of the worst. To avoid it:
- do not use urls, only ip addresses in sip.conf
or put your urls inside /etc/hosts (is what I do especially sip providers
urls)
or install a
True, but with all due respect, if the cache's TTL expires and the OP's PBX
cannot reach an external DNS server, they have bigger problems ;-)
Slainte all!
The Mick
Sent from my iPhone
On May 30, 2011, at 9:41 PM, Mark Deneen mden...@gmail.com wrote:
On Mon, May 30, 2011 at 2:44 AM,
On Mon, 30 May 2011, Sherwood McGowan wrote:
True, but with all due respect, if the cache's TTL expires and the OP's
PBX cannot reach an external DNS server, they have bigger problems ;-)
Slainte all!
The Mick
I couldn't disagree more. In fact I think this problem is more serious
than
Thank you very much Ruffell and Patrick.
The problem was basic. The OS was missing with correct kernel headers.
We installed correct kernel headers and its working fine now.
Regards,
Rajib
On Mon, May 30, 2011 at 02:29:37PM +0200, Patrick Lists wrote:
On 05/30/2011 10:03 AM, Deka, Rajib IN
Remove the _ in front of your dialplan,like
exten = 0678922645,1,--
On Mon, May 30, 2011 at 11:00 PM, salaheddine elharit
salah.elharit...@gmail.com wrote:
Hello list
i have configured astersik 1.4 with sip i have a question
when i put in dial plan.conf
exten =
Hi sir,
I was installed Goautodial server and I have b410p BRI card. BRI card
showing OK with dahdi_tool, this NT mode.
whenever I am dialing from server i am not able to connect the call . in Cli
below mention warning is comming .
please what is the mistake with me . help me
Executing
On Tue, May 31, 2011 at 10:48:08AM +0530, mahesh katta wrote:
Hi sir,
I was installed Goautodial server and I have b410p BRI card. BRI card
showing OK with dahdi_tool, this NT mode. whenever I am dialing from
server i am not able to connect the call . in Cli below mention
warning is
On Tue, May 31, 2011 at 11:08 AM, Shaun Ruffell sruff...@digium.com wrote:
On Tue, May 31, 2011 at 10:48:08AM +0530, mahesh katta wrote:
Hi sir,
I was installed Goautodial server and I have b410p BRI card. BRI card
showing OK with dahdi_tool, this NT mode. whenever I am dialing from
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