Hello all,
I installed yesterday the most recent AsteriskNow distro and used the
built-in upgrade options to bring it up to date.
My problem is that CDRs are not stored. I went through dozens of forums and
sites I found on Gogle relating to that issue (very common), but I couldn't
find a
Chris Datfung wrote:
Hi,
Hola,
I'm trying to get Incoming Google Voice calls to ring on my Iaxy. I'm
using Asterisk 11.0.1. Based on the the following configurations can
someone help me figure out why incoming Google voice calls are
not ringing on the Iaxy?
Did chan_motif successfully
Richard Kenner wrote:
I have a peculiar RTP issue. I'm experimenting with Jitsi as a softphone
on one of my desktop Windows machines. That machine can either be connected
to Asterisk via an VPN connection (with a static IP address) or not (via NAT).
When it's connected via NAT, all is OK.
Jerry Geis wrote:
I am running asterisk 1.4.43 on a really small network for testing, all
on same switch.
I launch a meetme between my server and 5 asterisk clients that
are all on 10 foot network cables all connected to the same switch.
The meetme is fine everything is in sync
Then I reboot
Amit Salunkhe wrote:
Hi All,
Hola,
I'm using 1.8 Asterisk and i havet set DTMF mode=rfc2833 in SIP global
default settings.
but when user sending DTMf event with SIP info method my asterisk
accepting that DTMF. If default or global setting is rfc2833 then how
come asterisk accepting SIP
On Mon, Nov 26, 2012 at 2:19 PM, Joshua Colp jc...@digium.com wrote:
I'm trying to get Incoming Google Voice calls to ring on my Iaxy. I'm
using Asterisk 11.0.1. Based on the the following configurations can
someone help me figure out why incoming Google voice calls are
not ringing on the
Chris Datfung wrote:
On Mon, Nov 26, 2012 at 2:19 PM, Joshua Colp jc...@digium.com
mailto:jc...@digium.com wrote:
I'm trying to get Incoming Google Voice calls to ring on my
Iaxy. I'm
using Asterisk 11.0.1. Based on the the following configurations can
someone
On Mon, Nov 26, 2012 at 3:53 PM, Joshua Colp jc...@digium.com wrote:
Chris Datfung wrote:
On Mon, Nov 26, 2012 at 2:19 PM, Joshua Colp jc...@digium.com
mailto:jc...@digium.com wrote:
Hi Joshua,
How can I verify that chan_motif successfully loaded? I didn't see any
errors during the
Hi list,
I face the following problem on incoming calls from my provider which
uses Asterisk 1.6.1.25, our asterisk being 1.8.17.0. Incoming calls are
not sended to the context set in provider sip.conf definition, but are
going to the default context setted in [general].
Provider uses few
On 26/11/2012 10:14 AM, Klaverstyn, David C wrote:
Hi All,
I’m new to Queues and I have created one as follows which seems to work ok.
[david-test]
strategy = rrmemory
timeout = 10
retry = 0
maxlen = 0
announce-frequency = 0
announce-holdtime = no
member = SIP/121
member = SIP/122
Chris Datfung wrote:
Hi Joshua,
I can confirm that chan_motif succesfully loaded:
asterisk*CLI module load chan_motif.so
Unable to load module chan_motif.so
Command 'module load chan_motif.so' failed.
[Nov 26 09:04:33] WARNING[28686]: loader.c:868 load_resource: Module
'chan_motif.so' already
Administrator TOOTAI wrote:
Hi list,
Hola,
I face the following problem on incoming calls from my provider which
uses Asterisk 1.6.1.25, our asterisk being 1.8.17.0. Incoming calls are
not sended to the context set in provider sip.conf definition, but are
going to the default context setted
Hi Dear List members ,
this is coming rather late but I took your advice and went ahead to install
Dahdi before installing libpri-1.4.13
and the error messages are now different.(see attachment)
Kindly help .
I have tried this several times and I get stuck on Libpri installation.
Your input is
Hi,
If were on this subject I'll throw in my question
Does named acl lists in asterisk 11 help for this or only for registrations?
Thanks,
-Original Message-
From: Joshua Colp jc...@digium.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Mon, 26 Nov 2012 10:28:05
To: Asterisk
isr...@gmail.com wrote:
Hi,
If were on this subject I'll throw in my question
Does named acl lists in asterisk 11 help for this or only for registrations?
ACLs don't control SIP peer matching, so no.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW -
Thought so but hoped other wise
Thanks
--Original Message--
From: Joshua Colp
To: ? ??
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] No matching peer for 'callerID' from
'85.xx.xx.2:5060'
Sent: Nov 26, 2012 4:40 PM
What's the configuration like for Jitsi in sip.conf?
Just fullname and md5secret plus a phones section that reads:
[phones](!)
type=friend
host=dynamic
context=SIP_Phones
cc_agent_policy=generic
cc_monitor_policy=generic
disallow=all
allow=gsm
allow=ulaw
allow=g729
allow=h264
What version of
Richard Kenner wrote:
What's the configuration like for Jitsi in sip.conf?
Just fullname and md5secret plus a phones section that reads:
[phones](!)
type=friend
host=dynamic
context=SIP_Phones
cc_agent_policy=generic
cc_monitor_policy=generic
disallow=all
allow=gsm
allow=ulaw
allow=g729
What NAT settings are globally in use?
nat=yes
Do you have directmedia turned off or on?
I've tried both ways, but I normally have it off.
This really does indeed feel like a weird NAT issue that is probably
configuration related (probably both in Jitsi and Asterisk).
Except that:
(1)
Richard Kenner wrote:
What NAT settings are globally in use?
nat=yes
Do you have directmedia turned off or on?
I've tried both ways, but I normally have it off.
This really does indeed feel like a weird NAT issue that is probably
configuration related (probably both in Jitsi and
Yeah this is so weird that packet captures are really needed. A working
call and a non-working call, along with what IP ranges are what.
There are *tremendous* numbers of RTP packets, of course. Are those
captures really going to be useful? That's the problem. If they
*are* going to be
Richard Kenner wrote:
Yeah this is so weird that packet captures are really needed. A working
call and a non-working call, along with what IP ranges are what.
There are *tremendous* numbers of RTP packets, of course. Are those
captures really going to be useful? That's the problem. If they
Not that many RTP packets are required. It's just important to see the
SIP signaling and where traffic is coming/going from with the network
topology in mind. That way a clearer picture of where it's saying media
should go to, where it's sending media from, etc can be gleamed. Once
that
Le 26/11/2012 15:28, Joshua Colp a écrit :
Administrator TOOTAI wrote:
Hi list,
Hola,
I face the following problem on incoming calls from my provider which
uses Asterisk 1.6.1.25, our asterisk being 1.8.17.0. Incoming calls are
not sended to the context set in provider sip.conf definition,
Not that many RTP packets are required. It's just important to see the
SIP signaling and where traffic is coming/going from with the network
topology in mind. That way a clearer picture of where it's saying media
should go to, where it's sending media from, etc can be gleamed. Once
that
Richard Kenner wrote:
Not that many RTP packets are required. It's just important to see the
SIP signaling and where traffic is coming/going from with the network
topology in mind. That way a clearer picture of where it's saying media
should go to, where it's sending media from, etc can be
1. Remove allow=gsm from your sip.conf and reload
That did it! Thanks!
But why should that have been an issue?
2. Disable ZRTP in Jitsi by going into Options - Accounts - Selecting
account - Edit - Security - Uncheck Enable support to encrypt calls.
That was one of the first things I
Richard Kenner wrote:
1. Remove allow=gsm from your sip.conf and reload
That did it! Thanks!
But why should that have been an issue?
The way you had things configured Asterisk was prioritizing GSM over
ULAW, so until Jitsi started responding it sent GSM. This apparently
upset Jitsi a
this is coming rather late but I took your advice and went ahead to
install Dahdi before installing libpri-1.4.13
and the error messages are now different.(see attachment)
This is compile error is reported by newer gcc compiler versions.
It is already fixed in libpri SVN.
The way you had things configured Asterisk was prioritizing GSM over
ULAW, so until Jitsi started responding it sent GSM.
I thought I might have seen something like that in the packets, but it
didn't look like it showed up in the SDP negotiations, so seemed
peculiar to me. Unclear why this
By not in sync do you mean that there is a delay between when the
speaker speaks and when the client hears it?
There's always going to be some amount of delay. It takes time to encode
the audio, send it, mix it (in this case), receive it, decode it, and
have it pass through a jitterbuffer
Hi List,
Until recently, I've been running an Asterisk server behind an MS ISA 2004
firewall. In general, this has worked fine - I've been able to connect to my
SIP provider to make/receive calls (sipgate.co.uk in the UK and
callcentric.com in the US), and DHADI runs the one traditional analogue
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On 11/24/12 4:07 PM, Richard Kenner wrote:
I have a peculiar RTP issue. I'm experimenting with Jitsi as a
softphone on one of my desktop Windows machines. That machine can
either be connected to Asterisk via an VPN connection (with a
static IP
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Hash: SHA1
Le 26/11/2012 04:26, Joshua Colp a écrit :
To others using chan_motif - are you experiencing the same issue?
I didn't use chan_motif since testing a few weeks ago, so I may I have
broke my configuration, but Google Voice seems to be broken now.
Jean-Denis Girard wrote:
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Hash: SHA1
Le 26/11/2012 04:26, Joshua Colp a écrit :
To others using chan_motif - are you experiencing the same issue?
I didn't use chan_motif since testing a few weeks ago, so I may I have
broke my configuration, but Google Voice
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