Hi @ll,
I just moved my Asterisk Box and changed the Provider and Internet Access to a
full IP Access by Deutsche Telekom.
I set up my sip.conf as I found various examples throughout the Net. Calls and
some other stuff is basically working.
The problem I ran into is, that the outgoing and
Try canreinvite=yes in sip trunk
-Original Message-
From: Florian Wolters flor...@florian-wolters.de
Sender: asterisk-users-boun...@lists.digium.com
Date: Thu, 21 Mar 2013 08:31:54
To: asterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
2013/3/21 Florian Wolters flor...@florian-wolters.de:
Hi @ll,
I just moved my Asterisk Box and changed the Provider and Internet Access to
a full IP Access by Deutsche Telekom.
I set up my sip.conf as I found various examples throughout the Net. Calls
and some other stuff is basically
On Mon, 2013-03-11 at 14:34 +0100, Patrick Lists wrote:
On 03/11/2013 12:53 PM, termo termosel wrote:
Hi,
I have Ubuntu and Asterisk 11.2.1 in a boot USB. When I put it in
desktop computer, asterisk starts without problem but if I insert
the
same USB in a laptop computer Asterisk
I had this exact problem with my voip provider a few years ago.
It was disconnecting at exactly 5 minutes.
I solved it by moving Asterisk 1.6 to Asterisk 1.4.
Try asterisk 1.4 or 1.8 on a test box and see how it goes.
Peter
On 21/03/2013 09:31, Florian Wolters wrote:
Hi @ll,
I just moved
I am having the same problem with Asterisk 11.2.0 and Linphone and it is
exactly 15 minutes and occurring with SIP running on our LAN.
On Thu, Mar 21, 2013 at 3:31 AM, Florian Wolters flor...@florian-wolters.de
wrote:
Hi @ll,
I just moved my Asterisk Box and changed the Provider and Internet
Hello,
I solved it by moving Asterisk 1.6 to Asterisk 1.4.
Try asterisk 1.4 or 1.8 on a test box and see how it goes.
I did try the latest 1.8.2x release already without any improvement.
Currently running is a Asterisk PBX 1.8.10.1~dfsg-1ubuntu1 as the tcpdump
says (little mistake to my last
Florian Wolters wrote:
So I turned on SIP debug for this host and analyszed it with wireshark.
The last packets show an INVITE from my provider, that is answered by my
Asterisk with 200 OK, with session description. What follows is an ACK
by the provider and immediately a BYE sent by the
Mitch Claborn wrote:
Thank you for that most excellent post. I had guessed at most of the
SDP fields and meaning.
No problem. I actually like looking at SIP traces for some reason.
I have wireshark traces from the client and the RTP packets are not in
the trace, which I think means
hello list,
i have installed 2 diguim cards in my server using asterisk 1.4 (i use the
old version with zapata.conf and zaptel.conf)
i want to use the span 1 for group 1 and span 2-span 6 for the group 2 (i
want to active the round-robin for span 2 and 6) in order to activate the
WIMAX and FH
On 3/21/2013 12:31 AM, Florian Wolters wrote:
Hi @ll,
I just moved my Asterisk Box and changed the Provider and Internet Access to a
full IP Access by Deutsche Telekom.
I set up my sip.conf as I found various examples throughout the Net. Calls and
some other stuff is basically working.
The
What do you mean by roundrobin here
On Mar 21, 2013 8:27 PM, Salaheddine Elharit salah.elharit...@gmail.com
wrote:
hello list,
i have installed 2 diguim cards in my server using asterisk 1.4 (i use the
old version with zapata.conf and zaptel.conf)
i want to use the span 1 for group 1 and
i mean the burden-sharing between Wimax and FH
2013/3/21 Bharat Lalcheta bharatlalch...@gmail.com
What do you mean by roundrobin here
On Mar 21, 2013 8:27 PM, Salaheddine Elharit salah.elharit...@gmail.com
wrote:
hello list,
i have installed 2 diguim cards in my server using asterisk 1.4
If u want to dial in round robin use Dial(zap/r2/2) . It dials using
channel in round robin
On Mar 21, 2013 9:37 PM, Salaheddine Elharit salah.elharit...@gmail.com
wrote:
i mean the burden-sharing between Wimax and FH
2013/3/21 Bharat Lalcheta bharatlalch...@gmail.com
What do you mean by
On Thu, 21 Mar 2013, Salaheddine Elharit wrote:
i have installed 2 diguim cards in my server using asterisk 1.4 (i use
the old version with zapata.conf and zaptel.conf)
question 2: what is difference between etc\zapataa.conf and
etc\asterisk\zapata.conf
There is no /etc/zapata.conf.
The 2
File is ok there is no etc/zapata file.
On Mar 21, 2013 9:42 PM, Steve Edwards asterisk@sedwards.com wrote:
On Thu, 21 Mar 2013, Salaheddine Elharit wrote:
i have installed 2 diguim cards in my server using asterisk 1.4 (i use
the old version with zapata.conf and zaptel.conf)
question
On Tue, 19 Mar 2013 02:15:10 +, Jaap Winius wrote:
Following an upgrade from Debian squeeze to wheezy, and Asterisk 1.6.2.9
to 1.8.13, my server is no longer able to register a connection to a SIP
account at my ISP (XS4ALL in the Netherlands). At the same time, it is
still able to
Hi folks,
Following an upgrade to Debian wheezy, I'm now running Asterisk 1.8.13.1.
As opposed to Asterisk 1.6.2.9 that I ran with squeeze, this version can
support IPv6. However, it seems that I can't get it to support both IPv4
and IPv6 at the same time. For example, if in sip.conf I set the
please see,
http://lists.digium.com/pipermail/asterisk-users/2013-March/278130.html
On Thu, Mar 21, 2013 at 5:47 PM, Jaap Winius jwin...@umrk.nl wrote:
Hi folks,
Following an upgrade to Debian wheezy, I'm now running Asterisk 1.8.13.1.
As opposed to Asterisk 1.6.2.9 that I ran with squeeze,
how can i use Dial(zap/r2/2)
below an exemple from my extensions.conf
exten = _0612.,1,Set(CALLERID(number)=520460587)
exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten =
Use r2 instead of g2 in dial
Dial(Zap/r2/${EXTEN}
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
hi,
exten = _0612.,1,Set(CALLERID(number)=520460587)
exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten =
_0612.,n,Dial(Zap/r2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
exten = _0612.,n,Hangup()
Note r
I did open a ticket with SFL support and sent them the packet trace.
Interestingly, using Bria we sometimes see similar, though not exactly
the same, symptoms. That would make me wonder about the TCP stack on
the client machine, or similar.
Bria on Ubuntu is not terribly stable. Bria on
Hello,
I have a variable created like
... Set(__myVar=${ARG1})
... Set(__${myVar}STATUS=)
If ARG1 is abcd, variable is abcdSTATUS and should be empty. This is OK.
Now I would like to get the value of abcdSTATUS. How to do it?
${${myVar}STATUS}} isn't working, nor ${{myvar}STATUS}
Thanks
All other phones we work with will auto-answer when we do this:
[macro-paging1way]
exten = s,1,SIPAddHeader(Call-Info: answer-after=0)
exten = s,n,Page(${PAGINGLIST})
exten = s,n, Hangup
The SPA phones simply ring. I have verified that Auto Answer Page is set
to yes (the default). We've tried
On Thu, Mar 21, 2013 at 2:48 PM, Carlos Alvarez car...@televolve.comwrote:
All other phones we work with will auto-answer when we do this:
[macro-paging1way]
exten = s,1,SIPAddHeader(Call-Info: answer-after=0)
exten = s,n,Page(${PAGINGLIST})
exten = s,n, Hangup
The SPA phones simply ring.
hi,
${myVar}STATUS is empty you have not assign any value here your var
Set(__${myVar}STATUS=) is empty.
use instead Set(__myVar=${ARG1}STATUS) and remove second line.
On Thu, Mar 21, 2013 at 7:45 PM, Administrator TOOTAI ad...@tootai.netwrote:
Hello,
I have a variable created like
...
Hello Everyone,
I have disallow=all and allow=g729 set in sip.conf however, it seems
that asterisk still thinks it support other codecs:
Capabilities: us - 0x8008000e (gsm|ulaw|alaw|h263|testlaw). How
can I disable gsm,ulaw,alaw.
Thanks in Advance,
Nick.
--
On Thu, Mar 21, 2013 at 11:58 AM, Optical Phoenix
opticalphoe...@gmail.comwrote:
Hi Carlos,
According to this site,
http://community.linksys.com/t5/VoIP-Phones/SPA942-auto-answer-page-intercom-beeping-loudly/td-p/215064the
sip string should be
Call-Info:\;answer-after=0. I have not
please post sip.conf.
On Thu, Mar 21, 2013 at 8:01 PM, Nick Khamis sym...@gmail.com wrote:
Hello Everyone,
I have disallow=all and allow=g729 set in sip.conf however, it seems
that asterisk still thinks it support other codecs:
Capabilities: us - 0x8008000e
- Original Message -
From: Jaap Winius jwin...@umrk.nl
To: asterisk-users@lists.digium.com
Sent: Thursday, March 21, 2013 12:47:57 PM
Subject: [asterisk-users] Asterisk 1.8 and dual stack support
Hi folks,
Following an upgrade to Debian wheezy, I'm now running Asterisk
1.8.13.1.
On Thu, 21 Mar 2013, Administrator TOOTAI wrote:
I have a variable created like
... Set(__myVar=${ARG1})
... Set(__${myVar}STATUS=)
If ARG1 is abcd, variable is abcdSTATUS and should be empty. This is OK.
Now I would like to get the value of abcdSTATUS. How to do it?
${${myVar}STATUS}}
Hi there
Michael L. Young wrote:
How are you determining that it is not listening on IPv4?
bindaddr=:: should allow you to support dual stack.
Which is the way it works with 1:1.8.11.1-1digium1~squeeze.
I even use Asterisk as a RTP audio IPv4 - IPv6 proxy.
Regards,
Rob
--
Hello Asghar,
I fixed the issue after I realized that I was specifying allow before
disallow. Sorry for the noise!!!
Nick.
On 3/21/13, Asghar Mohammad asghar...@gmail.com wrote:
please post sip.conf.
On Thu, Mar 21, 2013 at 8:01 PM, Nick Khamis sym...@gmail.com wrote:
Hello Everyone,
I
On 03/21/13 14:14, Gerard wrote:
I think a simple tcpdump of the traffic will show the mystery. It can
be your provider doing something nasty. Have you tried using some
other cheap SIP termination? or arrange a fake termination yourself
on another server?
Leandro
I thought so too, but it
On Thu, 21 Mar 2013 12:21:44 -0700, Michael L. Young wrote:
How are you determining that it is not listening on IPv4?
bindaddr=:: should allow you to support dual stack.
That's what I thought would happen. When I set bindaddr=:: and use
'netstat -lpn |grep 5060' it shows:
udp6 0 0
Le 21/03/2013 20:27, Steve Edwards a écrit :
On Thu, 21 Mar 2013, Administrator TOOTAI wrote:
I have a variable created like
... Set(__myVar=${ARG1})
... Set(__${myVar}STATUS=)
If ARG1 is abcd, variable is abcdSTATUS and should be empty. This is OK.
Now I would like to get the value of
:)
On Thu, Mar 21, 2013 at 10:27 PM, Jaap Winius jwin...@umrk.nl wrote:
On Thu, 21 Mar 2013 12:21:44 -0700, Michael L. Young wrote:
How are you determining that it is not listening on IPv4?
bindaddr=:: should allow you to support dual stack.
That's what I thought would happen. When I
hi,
exten 000,1.Progress() work in some situation.
On Thu, Mar 21, 2013 at 9:30 PM, Gerard gsara...@rarcoa.com wrote:
On 03/21/13 14:14, Gerard wrote:
I think a simple tcpdump of the traffic will show the mystery. It can
be your provider doing something nasty. Have you tried using some
On Thu, 21 Mar 2013, Administrator TOOTAI wrote:
I have a variable created like
... Set(__myVar=${ARG1})
... Set(__${myVar}STATUS=)
If ARG1 is abcd, variable is abcdSTATUS and should be empty. This
is OK.
Now I would like to get the value of abcdSTATUS. How to do it?
- Original Message -
From: Jaap Winius jwin...@umrk.nl
To: asterisk-users@lists.digium.com
Sent: Thursday, March 21, 2013 5:27:37 PM
Subject: Re: [asterisk-users] Asterisk 1.8 and dual stack support
That's what I thought would happen. When I set bindaddr=:: and use
'netstat -lpn
On Thu, 21 Mar 2013 16:02:17 -0700, Michael L. Young wrote:
Let me try to understand this. With bindaddr set as bindaddr=::, upon
starting Asterisk, you are fine and all your IPv4 peers connect
properly. Therefore, dual stack is working at this point. ...
You minunderstand. When I start
On Thu, 21 Mar 2013 16:35:16 +, Jaap Winius wrote:
Hopefully, my ISP will see fit to squash this bug ASAP.
Well, I got my answer from them quickly enough: Nope.
Luckily, somebody was kind enough to suggest a workaround. Unfortunately,
it involves, downloading the source code and making a
I found the problem, it's (I think) a bug with queue command. My
dialplan:
[context]
...
exten = 33123,n,macro(unpauseQueueMembers,q820,104,105,136,,)
exten = 33123,n(back2Queue),Queue(${myQueue},nit,,,14400)
exten = 33123,n,NoOp(Queue ${myQueue} call status is ${QUEUESTATUS}
-
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