Scott, thank you four your reply.
I had already though about both options, but the problem is, that after an ip
change AND a new registration the ip address of the peer is not updated
automatically. INVITES are answered with 401.
Only after a sip reload the peer works again.
That can't be
Hi Patrick,
You are welcome to try our tools out for active and passive voice quality
measurement tools. It's waveform analysis (like PESQ or POLQA) and VoIP
metrics analysis (like G.107 E-model and other metrics).
You can read more at http://www.sevana.biz
or older site http://www.sevana.fi
Hi all,
since asterisk 11 (1.6 was okay) failed the ReceiveFax-Application
when it called about Dial and a Local-Channel.
Directly from external to FaxReceive is no problem.
Cut from cli:
[...]
[Apr 1 11:12:31] -- Executing [s@macro-redirection:85]
Dial(SIP/access-trunk-0001,
Hello -
I am trying to decide if I have stumbled across a bug in PJSIP or I am just
missing something. My Asterisk has two interfaces, an internal eth0 and
an external eth1. In pjsip.conf, I define the following transports:
[trusted]
type=transport
protocol=udp
bind=10.xx.yy.zz:5060
[untrusted]
If I correctly understand what the problem is, what I did was write a
script that runs out of CRON every 15 minutes. It checks the outside IP address
by querying http://checkip.dyndns.org and compares it to the IP address stored
in the parameter “externip” in the [general] section of
John,
thank you four your answer. I think you have misunderstood the problem. It’s
about a ip address change of the sip trunk, not of my asterisk server.
Kind regards,
Daniel
Am 01.04.2015 um 16:40 schrieb Tech Support aster...@voipbusiness.us
mailto:aster...@voipbusiness.us:
If I
On 4/1/15 10:48 AM, Daniel Heckl wrote:
John,
thank you four your answer. I think you have misunderstood the
problem. It’s about a ip address change of the sip trunk, not of my
asterisk server.
You would probably benefit by enabling the DNS Manager to allow for
dynamic IP changes:
# cat
On 4/1/15 7:50 PM, Andrew Galdes wrote:
Hello all,
I have an Asterisk server (Asterisk 10.12.4) with multiple sip
accounts with the same service provides. We have 8 phone numbers in
total.
Incoming calls from the public are all correctly directed to
appropriate office handsets. However,
This is one of the chronic problems. Try this option in sip.conf:
match_auth_username=yes
Carefully read the description, it is better to test in after hours.
02.04.2015 2:50, Andrew Galdes пишет:
Hello all,
I have an Asterisk server (Asterisk 10.12.4) with multiple sip
accounts with the
Hello all,
I have an Asterisk server (Asterisk 10.12.4) with multiple sip accounts
with the same service provides. We have 8 phone numbers in total.
Incoming calls from the public are all correctly directed to appropriate
office handsets. However, the display on the reception phone (the only one
Can you show us the CDR record for that call?
And maybe what your s priority of your incoming context is?
It should be easy to get what number was dialed, Try:
${CUT(PASSTHRU(${SIP_HEADER(TO):5}),@,1)}
Normally I display the callers number on my phones, Not the number they
dialed?
On Wed, Apr
The Asterisk Development Team has announced the release of Asterisk 13.3.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.3.0 resolves several issues reported by the
community and would have not been possible
The Asterisk Development Team has announced the release of Asterisk 11.17.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 11.17.0 resolves several issues reported by the
community and would have not been possible
On Tue, Mar 31, 2015 at 12:36:34PM +0200, Daniel Heckl wrote:
Hello Sebastian,
I had already seen this list of the hosts, but it is not active. All
servers with which my Asterisk has been communicated are not listed.
A port scan, to eventually update the list, found hundreds of servers
On Wed, Apr 01, 2015 at 11:00:56AM -0400, Andres wrote:
On 4/1/15 10:48 AM, Daniel Heckl wrote:
John,
thank you four your answer. I think you have misunderstood the
problem. It’s about a ip address change of the sip trunk, not of my
asterisk server.
You would probably benefit by
Dan Cropp wrote:
I am running asterisk 13.1.0
In pjsip.conf, the endpoint section has an aors and an auth field.
I can name the auth field anything I want. The key is to set the
auth=field accordingly.
However, when I try this with the aors field, it never works. It seems I
have to name the
Thanks Joshua.
That must be it. I'm using PhonerLite and a Cisco SPA504G phone.
Have a great day!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: Wednesday, April 01, 2015 3:07 PM
To:
Hi,
I'm maintaining the FreeBSD ports for asterisk(With madpi...@freebsd.org
as identity). Here's a link to the
asterisk13 port for your reference:
http://www.freshports.org/net/asterisk13/
I performed some tests with RC1 and am doing some final tests with the
final release before committing
I am running asterisk 13.1.0
In pjsip.conf, the endpoint section has an aors and an auth field.
I can name the auth field anything I want. The key is to set the auth=field
accordingly.
However, when I try this with the aors field, it never works. It seems I have
to name the aors=field to
I just realized that you are asking about dynamic AORs, not static Contacts
in an AOR. That may be the difference. I have never actually tried giving a
dynamic AOR a different name. And you wouldn't want more than one dynamic
AOR, you'd just use an AOR that allowed more than 1 contact.
On Wed,
Thanks Trey.
Have a great day!
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Trey Hilyard
Sent: Wednesday, April 01, 2015 3:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP
I don't know why you have issues using different names. I have multiple
AORs assigned to a single endpoint and it works fine. I have to admit that
my AORs do contain the endpoint name, though. For example, for endpoint
myswitch I have two AORs, myswitch_1 and myswitch_2, and I assign
them to the
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