Hi Everyone,
I have setup an OpenVPN tunnel between Server A (running Asterisk) and
Server B suppling it's SIP Phones with DHCP pool of IPs.
So, the tunnel is established nicely and everyone can ping others. "sip show
peers" shows the local subnet of the SIP Phones registered (192.168.100.0/24
).
On Tue, Sep 7, 2010 at 5:36 PM, Fabiano Carlos Heringer <
b...@grupoheringer.com.br> wrote:
> Em 07/09/2010 17:15, Miguel Molina escreveu:
>
> El 07/09/10 14:49, Fabiano Carlos Heringer escribió:
>
> Is there a way to solve the mess on CDR caused by CDR Transfer? anyway, by
> paid support, no pai
Hi,
I fixed it in the end by adding the sip headers I was interested in as extra
"x" headers in the openser config. Then just capturing these in the asterisk
dialplan as variables. Simples.
Regards
Jon
On 21 Sep 2010 16:03, "Jonas Kellens" wrote:
> On 09/21/2010 04:22 PM, Jon Farmer wrote:
>> O
> I use realtime on 1.4 and 1.6 servers but always with rtcachefriends=yes in
> sip.conf
I already use that and it doesnt seem to re-register when a call comes in.
Only when the registration period expires, or the peer dials out.
--
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On 09/21/2010 06:09 PM, Marcus Vinicius wrote:
> I tried relaxdtmf = yes but has not worked.
>
> If I type very slowly digits are recognized normally. But if I dial a
> number and enter the redial button, the digits are recognized in the
> asterisk. It appears that:
>
> [Sep 21 19:20:24] DEBUG [4
On Tue, 2010-09-21 at 19:04 -0400, Dan Journo wrote:
> I checked the bug reports and all I could find was similar issues with the
> Asterisk 1.6 which (according to the reports) have been resolved.
> I couldnt find anyone talking about 1.4 so I created a new issue and someone
> closed the case an
> I tried relaxdtmf = yes but has not worked.
>
> If I type very slowly digits are recognized normally.
Then indeed it won't make a difference. If that were your problem, it
likely wouldn't work at any speed.
--
_
-- Bandwidt
Hello,
thanks for the reply.
I tried relaxdtmf = yes but has not worked.
If I type very slowly digits are recognized normally. But if I dial a number
and
enter the redial button, the digits are recognized in the asterisk. It appears
that:
[Sep 21 19:20:24] DEBUG [4751] chan_dahdi.c: waitfordi
I checked the bug reports and all I could find was similar issues with the
Asterisk 1.6 which (according to the reports) have been resolved.
I couldnt find anyone talking about 1.4 so I created a new issue and someone
closed the case and added this note:-
> This does not appear to be a bug, but
I have a project where I need to connect two E1 links from different
providers. One will be PRI ISDN (Telefonica) and the other MFC/R2
(Telmex). There should not be any problem supporting both types of link
on a single TE220B card but my concern is more about who will be the
primary timin
> I dial 12345678, but only '16 'is received by the asterisk.
You may want to try
relaxdtmf=yes
in chan_dahdi.conf. That fixed a similar problem for me.
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-- Bandwidth and Colocation Provided by http://www.api-digital
Hello
I use Asterisk with FXS extensions in chan_dahdi and I'm having trouble
detecting the digits in dahdi.
I dial 12345678, but only '16 'is received by the asterisk. The following
appears in the logs:
[Sep 21 18:11:44] DTMF [8536] channel.c: DTMF end '1 'received on DAHDI/10-1,
duration 0
On Tue, 21 Sep 2010, Shaun Ruffell wrote:
> On 09/21/2010 10:48 AM, Jeff LaCoursiere wrote:
>> I have several servers with Sangoma A104d cards, and the Sangoma driver
>> has a debug mode that lets me see the RBS bit transitions. I have used
>> this in the past to prove that the T1 provider is ac
Did you get this to work? If not, shoot me an email. We use the Polycom's,
and I can send you our config file.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of colin mcdermott
Sent: Friday, September 10, 201
On 09/21/2010 10:48 AM, Jeff LaCoursiere wrote:
> I have several servers with Sangoma A104d cards, and the Sangoma driver
> has a debug mode that lets me see the RBS bit transitions. I have used
> this in the past to prove that the T1 provider is actually triggering
> the hangup from their side.
Every time I start Asterisk or do a simple reload I see this message:
“Cannot open maximum file descriptor 32767 at boot? No such file or
directory”.
It only works if I set 1024 in asterisk.conf "maxfiles"
However, my
sysctl fs.file-max
fs.file-max = 65535
and my ulimits are
ulimit -a
core file s
Hi,
I have an asterisk 1.4.35 server with a Digium TE410P (1st gen) four
port T1 card. Only one RBS T1 plugged into it right now.
I have been getting complaints about random hangups. Endpoints are all
remote, but I track very closely the latency (by graphing the output of
"sip show peers") whic
Personally, I would like to see less commercial marketing on
http://asterisk.org. I count 5 separate marketing ads on the download
page alone. This is just my opinion.
The level of commercialism on the Asterisk.org download page does not
bother me at all. Seems eminently fair for Digium to a
On Tuesday 21 September 2010 08:42:04 CDR wrote:
> Every time I start Asterisk or do a simple reload I see this message:
> "Cannot open maximum file descriptor 32767 at boot? No such file or
> directory"
> Does anybody have some idea of what can it be? It did not happen in version
> 1.4.
> Philip
Hi!
> Could somebody tell me how to use SHARED function?
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+shared
> I want to get RTCP stats from SIP, but current channel is DAHDI.
> How can I get SIP channel?
If you have one DADHI and one SIP channel bridged together, then only for
On 09/21/2010 04:22 PM, Jon Farmer wrote:
> On 16 September 2010 22:23, Barry Miller wrote:
>
>
>> For an interim fix, setting res_agi=1.4 in the [compat] section of
>> asterisk.conf should work. See UPGRADE-1.6.txt .
>>
> I have tried this but it still complains about the pipe not bein
On 16 September 2010 22:23, Barry Miller wrote:
> For an interim fix, setting res_agi=1.4 in the [compat] section of
> asterisk.conf should work. See UPGRADE-1.6.txt .
I have tried this but it still complains about the pipe not being a comma.
Regards
Jon
--
Jon Farmer
Tel 07795 118140
--
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of CDR
Sent: Tuesday, September 21, 2010 8:42 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Unexplained message in 1.6.2
Every time I start Asterisk or do a simple reload
Every time I start Asterisk or do a simple reload I see this message:
"Cannot open maximum file descriptor 32767 at boot? No such file or
directory"
Does anybody have some idea of what can it be? It did not happen in version
1.4.
Philip
--
__
I wholly disagree. Open-source does not imply not-for-profit at all. Look at
Red Hat. They sell open source software, by way of selling support, and access
to stable repositories for updates. So this line does not need to exist. If the
line does exist, then I agree it should be well defined.
T
Failed to grab lock, is usually used when referencing that it can't lock the
port (in this case the sip port you use), because the port is used by
another app/service. Just a tip.
And by the way, *DO* upgrade.
On Tue, Sep 21, 2010 at 12:08 PM, dashy dude wrote:
> I understand the point.
> Howev
On 09/21/2010 04:26 AM, t. k wrote:
>
> Hi
>
> Thanks for help.
>
>
>> I will try to help. But others might know more. What SIP client are you
>> using - a softphone, a hardphone? It looks like the client is sending
>> the full " at 192.168.0.1" instead of just "" as the username.
> Sebas
On 09/21/2010 03:41 AM, Rod Montgomery wrote:
[/snip]
>
> Does anyone reading this have an opinion on whether commercial
> listings for complementary products and services should appear
> directly on Asterisk.org?
Just my two cents - but I prefer that organisations keep a clear line
between ope
On Tue, Sep 21, 2010 at 8:33 AM, Andrew Thomas wrote:
> I was wondering what happened if YOU put that number in. Does it put
> everyone in to the same conference?
>
> That would, at least, prove that the MeetMe app was working as it should
> (unless you've tried this already).
>
>
> yes, as i sai
I was wondering what happened if YOU put that number in. Does it put
everyone in to the same conference?
That would, at least, prove that the MeetMe app was working as it should
(unless you've tried this already).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[ma
I understand the point.
However, till the time I upgrade, I need to figure out how to stop this.
Also, I checked the bug ID 6181. but could not find something like a version in
which this is closed. So unable to decide as to which version I need to go to.
Regards
--- On Mon, 9/20/10, dotnetdub
Hi Friends,
LOCAL/*89/9875784578
I want to match above dialstring into dialplan context.
How can i match dialplan extension pattern matching for "*89/9875784578"
with including '/' character.
Thanks in advance.
--
Best Regards,
Rajnikant Vanza
--
__
Hello!
Could somebody tell me how to use SHARED function?
I want to get RTCP stats from SIP , but current channel is DAHDI.
How can I get SIP channel?
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