What version of asterisk are you using?
Ish
On 17 February 2014 20:49, Mike Diehl mdiehlena...@gmail.com wrote:
Hi all,
I'm trying to build a fax relay mechanism where faxes come in and get
relayed out to their final destination. I'm using the h extension to store
various results from
Hello
I have a problem where I would like to be able to send an arbitrary SIP
domain when sending a call to a registered friend. By default the from
domain is set to the IP of the Asterisk server, but I would like to set it
to something else.
The case is that when a call from a foreign
Thanks a lot Patrick.
Regards
Rajib Deka
Siemens Ltd.
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Message: 7
Date: Mon, 17 Feb 2014 10:22:02 +0100
From: Patrick Laimbock patr...@laimbock.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk crashes at meetme kick all
Message-ID:
- Original Message -
On 13 Feb 2014, at 18:10, Tim Nelson tnel...@rockbochs.com wrote:
I recently experienced an odd situation. I have an Asterisk 11.5.0
system (Box A) with a SIP peering to another Asterisk 1.8.23.0
system (Box B). At some point, Box A started sending over 65Mbps
hello,
try to use failed instead of h
exten = failed,1,
best regards.
2014-02-18 9:09 GMT+00:00 Ishfaq Malik i...@pack-net.co.uk:
What version of asterisk are you using?
Ish
On 17 February 2014 20:49, Mike Diehl mdiehlena...@gmail.com wrote:
Hi all,
I'm trying to build a fax
I am using Realtime on Asterisk 11.5 with a SQLite3 backend. While
everything seems to be working fine I keep getting this error on my log
files:
[2014-02-17 19:55:18] WARNING[20569] res_config_sqlite3.c: Could not
execute 'UPDATE sip_buddies SET ipaddr = '192.168.2.23', port =
'5060',
I can't imagine it working any other way.
Either your phones are on static IP addresses or they must register to inform
Asterisk the IP associated with the peer entry in sip.conf.Unless you have
chan_psychic.so Asterisk won't know the IP of the phones unless you tell it.
-Original
Il 18/02/2014 19.15, Carlos Chavez ha scritto:
I am using Realtime on Asterisk 11.5 with a SQLite3 backend. While
everything seems to be working fine I keep getting this error on my log
files:
[...]
[2014-02-17 20:55:15] WARNING[20569] res_config_sqlite3.c: Could not
execute 'UPDATE
Attach the packet capture to your Jira bug report or post it online somewhere.
Hopefully someone will look at it.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Tuesday, February 18, 2014
Is Asterisk not able to gather IP and port info on INVITES in a REGISTER free,
host=dynamic setup? As you all know REGISTERS are resourceful and the
phone can be anywhere..
Kind Regards,
Nick.
--
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-- Bandwidth and
No. Asterisk will accept calls from unregistered devices, but you have to
enable guests I sip.conf and hope your dialplan is secure. No sane person does
this.
Asterisk cannot send calls to a device unless it knows the address from a
register or from a host= entry for the peer.
You may
On 2/18/2014 2:09 PM, Eric Wieling wrote:
No. Asterisk will accept calls from unregistered devices, but you have to
enable guests I sip.conf and hope your dialplan is secure. No sane person does
this.
Asterisk cannot send calls to a device unless it knows the address from a
register or
Hello Eric,
On 2/18/14, Eric Wieling ewiel...@nyigc.com wrote:
No. Asterisk will accept calls from unregistered devices, but you have
to enable guests I sip.conf and hope your dialplan is secure. No sane
person does this.
Thank you for your response. Our security layer is abstracted out
autocreatepeer still requires registration, from sip.conf.sample:
;autocreatepeer=no ; Allow any UAC not explicitly defined to
register
; WITHOUT AUTHENTICATION. Enabling this options
poses a high
; potential security
I just want to clarify. We are not creating peers automatically. And
we allowguest=no. We do have peer entries in sip_buddies db as you
would expect. As mentioned, we just don't allow phones to REGISTER
every 3600 (for example). Once a valid peer/phone tries to place a
call, we would like asterisk
We are seeing the message PRI Error on span 1: Received MDL/TEI management
message, but configured for mode other than PTMP on one of our Asterisk boxes
on a PRI. A Google search turns up a number of hits for this error, but they
are all for BRI not PRI.
I'm reasonably sure there are no
MDL/TEI is pretty low level.
Which card are you using?
chan_dahdi.conf?
Card config?
PCAP trace? Q.931 and Q.921 messages are relatively easy to debug with
wireshark.
jg
--
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-- Bandwidth and Colocation Provided by
On Mon, 17 Feb 2014, Mike Diehl wrote:
Is there something I need to do in order to get the h extension to get
called?
Would the 'g' dial() option help?
Proceed with dialplan execution at the current extension if the
destination channel hangs up.
It won't take you to h, but it may allow
I was not aware Wireshark worked on PRI spans. What interface should I tell it
to watch? 8-|
Card is: wanpipe: AFT-A101-SH PCI T1/E1 card found (HDLC (DS) rev.37), cpu(s) 1
chan_dahdi.conf has a timestamp of Jun 25 of last year
# cat /etc/asterisk/chan_dahdi.conf
;autogenerated by
Hi,
http://wiki.sangoma.com/wanpipe-wireshark-pcap-pri-bri-wan-t1-e1-tracing
On 18 February 2014 21:34, Eric Wieling ewiel...@nyigc.com wrote:
I was not aware Wireshark worked on PRI spans. What interface should I
tell it to watch? 8-|
Card is: wanpipe: AFT-A101-SH PCI T1/E1 card found
asterisk-users-boun...@lists.digium.com wrote on 02/18/2014 01:35:13 PM:
From: Nick Cameo sym...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com,
Date: 02/18/2014 01:35 PM
Subject: Re: [asterisk-users] Host = Dynamic in a Register Free
In case anyone is confused about the message reported not showin up in pri
debug. They show up in the Asterisk logs.
[2014-02-13 14:39:31] WARNING[2932] chan_dahdi.c: PRI Error on span 1: Received
MDL/TEI managemement message, but configured for mode other than PTMP!
[2014-02-13 14:39:34]
Nifty! I never knew that.
Here is a pcap:
http://help.nyigc.net/tmp/isdn.pcap
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lukasz Grzywanski
Sent: Tuesday, February 18, 2014 3:37 PM
To: Asterisk Users
Eric,
since you seem to have a Sangoma card, you should contact Sangoma's excellent technical support.
I'll have a look at your pcap trace tomorrow. You can also download the relevant Q.931 and Q.921
docs from ITU's web server.
Sangoma's wanpipe drivers allow you to configure a few hardware
My recent experiences with Sangoma tech support have been less than good. I
admit the issue was rather wieid. We had an issue about a year ago where
Sangoma PCI cards would not work in the servers we were purchasing unless you
used a PCI/PCI-X converter/riser card. Everything would load
Hello, a few days ago I sent a question:
http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html
but no one answered me! I just want to know is it possible or not?
--
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-- Bandwidth and Colocation Provided by
On Wed, 19 Feb 2014, Gholamreza Sabery wrote:
Hello, a few days ago I sent a question:
http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html
but no one answered me! I just want to know is it possible or not?
If it were only so easy...
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