Hi
I add the details.
The error seems that UAC set different username of digest.
But UAC cannot send same username of digest and from for specification.
So I want to know how to solve with Asterisk.
Register
From: ;tag=644056924
To:
Call-ID: 2457796...@192.168.0.2
CSeq: 125 REGISTE
Hi guys,
Interesting discussion - I learnt quite a bit. Thanks.
That said, no one's yet answered my two original questions. Anyone know?
To repeat:
1. When I used the line "dateformat=%F %T" in the general section of
logger.conf, the format in /var/log/asterisk/full did change, but the
rou
Any Idea??
On Mon, Aug 30, 2010 at 11:11 AM, Pratik Shrestha wrote:
> Oh so sorry.
> Yes you are right, the 'callee'.
>
> We have one soft switch somewhere located in US.
> When the call comes, then asterisk has to see the callee number in the sip
> extensions. If the number is not in its ext
On 1 Sep 2010, at 10:30, Pratik Shrestha wrote:
> Any Idea??
Read
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf
But I'm guessing you knew that and are just after getting someone else to do
the work...
Just create a catch-all pattern to match anything yo
Quoting Matt Riddell :
> Maybe you could do:
>
> Set(CDR(userfield)=${CALLERID(num)})
>
> Before dialing SIP/1000
That looks so simple -- and it actually works! -- although exactly not
in the way that I was expecting. Instead of replacing the contents of
one of the existing fields, a new fiel
Hello,
After installing on Ubuntu 10.04 using the tutorial on
http://hmontoliu.blogspot.com/2010/03/installing-asterisk-and-freepbx-on.html
I have a running instance of Asterisk.
PROBLEM: result of dahdi_cfg:
DAHDI Tools Version - 2.2.1
DAHDI Version: 2.2.1
Echo Canceller(s): MG2
Configuration
Hi there,
Your problem is with the Tiger ISDN kernel module claiming the digium card. I
have the following in /etc/modprobe.d/blacklist.conf
blacklist hisax
blacklist netjet
blacklist isdn
blacklist mISDN_core
blacklist mISDN_ipac
Once the netjet driver isn't claiming the card, the dahdi module
On Wed, Sep 01, 2010 at 02:58:59PM +0300, Mehmet Kuzulugil wrote:
> Hello,
> After installing on Ubuntu 10.04 using the tutorial on
> http://hmontoliu.blogspot.com/2010/03/installing-asterisk-and-freepbx-on.html
> I have a running instance of Asterisk.
>
> PROBLEM: result of dahdi_cfg:
> DAHDI To
Has any advancement been made to get 3102 operational in either a SIP or
H323 asterisk environment.
A post back in time mentioned a downloader service.
>From the posts and articles I have read, the NCP is acting like a bootp
and tftp server which uploads the configuration to the phone??
Am I close
Hi list,
Im using asterisk 1.6.0.10 and have following dialplan for doing chanspy
[app-chanspy]
include => app-chanspy-custom
exten => 555,1,Read(SPYNUM,extension)
exten => 555,2,ChanSpy(SIP/${SPYNUM},q)
exten => 555,n,Hangup
but if the channel is hang up or even destroyed the chanspy is not
chanspy as best I can tell from the code will not lock on a single device and
when that device goes away exit. What is passed to chanspy is a template for a
channel name. I submitted a patch to add option s so that chanspy would stop
when the one channel I wanted to watch went away or I used * t
On Wed, Sep 1, 2010 at 6:09 AM, Barry Fawthrop wrote:
> Has any advancement been made to get 3102 operational in either a SIP or
> H323 asterisk environment.
> A post back in time mentioned a downloader service.
> >From the posts and articles I have read, the NCP is acting like a bootp
> and tftp
On Wednesday 01 September 2010 09:01 PM, Jim Dickenson wrote:
> chanspy as best I can tell from the code will not lock on a single device and
> when that device goes away exit. What is passed to chanspy is a template for
> a channel name. I submitted a patch to add option s so that chanspy would
Hello all,
Has anyone have magicjack working with their asterisk? I had patched
chan_sip.c with some code that allows asterisk to do the md5 hash that mjmd5
proxy does. * shows that it is registered with magicjack, but incoming calls
are not even hitting my * box and outgoing calls get congesti
I had the same need which is why I submitted the patch. I think the feature
might finally be added to 1.8, it I remember correctly. I am not aware of any
other way around this.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Sep 1, 2010, at 9:12 AM, Rushikesh wrote:
Looks to me like the problem is with your dial command;
You're trying to do dial(sip/a,30,r)
Instead of
Dial(sip/a/b,30,r)
In simple terms, if I have extension sip/170 and I do
Dial(sip/170,30,r)
That's ok because sip/170 is an extension
But the magicjack is a trunk, so I have to do
Jeff Jones wrote:
> Has anyone have magicjack working with their asterisk? I had patched
> chan_sip.c
I highly recommend running 'mjproxy' on a host with a public IP instead of
patching asterisk; I can confirm that it is a good solution and asterisk
can be configured as a trunk for incoming and
On Tue, Aug 31, 2010 at 9:34 AM, Danny Nicholas wrote:
> Why not just copy the _1NXXNXX line into the remote context?
>
Well, that could be done, and probably would be a good tactic if you have
lots of DID's
and want to do db lookup or something to direct the next call leg.
But, if you onl
Hi, I have a very strange problem. In the middle of the call the MOH starts
for 30 seconds approximately.
After this the call run normally.
Anybody have an ideia or has some similar problem?
Thanks in advance!!
--
Atenciosamente,
---
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dario Quiroz
Subject: [asterisk-users] MOH in the middle of the call
>Hi, I have a very strange problem. In the middle of the call the MOH starts
for 30 seconds approximately.
>After this
The Asterisk Development Team has announced the release of libpri 1.4.11.4.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/libpri/
The release of libpri 1.4.11.4 resolves several issues reported by the
community and would have not been possible with
Anyone know of an ITSP that can offer DDIs (or DIDs) from India?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http:
Danny Nicholas schrieb:
>
> *From:* asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dario
> Quiroz
> *Subject:* [asterisk-users] MOH in the middle of the call
>
> >Hi, I have a very strange problem. In the middle of the call the MOH
> sta
Thanks Steve..
On Wed, Sep 1, 2010 at 5:13 PM, Steve Howes wrote:
>
> On 1 Sep 2010, at 10:30, Pratik Shrestha wrote:
>
> > Any Idea??
>
> Read
> http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf
>
> But I'm guessing you knew that and are just after getting some
Hi all, I am configuring asterisk in a cable modem network, using a
motorola TM401A.
I can make calls from the MTA but I can receive, display the following
error:
-- Executing [1...@alberti:1] Dial("OSS/dsp",
"MGCP/aaln/1...@0-13-11-82-bd-a.ssw.intercal.net|30") in new stack
[Sep 2 00:10
Matt Riddell wrote:
> On 25/08/10 8:54 PM, cov...@ccs.covici.com wrote:
> > Hi. I have a soft phone -- expresstalk-- on a computer in my network
> > and I use the internal ip address of the asterisk box to register the
> > phone. But using asterisk-1.8 between revisions 281912 and 281982 it
> >
Hi Everyone,
I have two servers as the following that are trunked with each other via
IAX2 trunk:
Server A:
Asterisk 1.4.21.2 (Elastix Flavor)
Server B (IP # 72.72.72.72):
Asterisk 1.6.2.0 (Vanilla)
Server B can place calls to Server A but when trying to place calls from
Server A to Server B t
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