On 15-03-20 6:42 AM, thufir wrote:
I wasn't able to get much out of babytel, beyond the fact that I was,
apparently, sending options which is why I'm not getting 200 OK.
How can I, generally speaking, ping/telnet or otherwise test the
connection to get more data?
A connection to this peer
On 15-03-20 6:55 AM, dotnetdub wrote:
Turn on sip debugging for this peer and watch for the options sending
and response.
If you are getting a response to your options asterisk shouldn't be
marking the peer as unavailable.
is your asterisk behind a firewall?
this fits with tech support
On 15-03-20 6:55 AM, dotnetdub wrote:
Turn on sip debugging for this peer and watch for the options sending
and response.
If you are getting a response to your options asterisk shouldn't be
marking the peer as unavailable.
is your asterisk behind a firewall?
I have a Cisco DPC3825 DOCSIS
Turn on sip debugging for this peer and watch for the options sending
and response.
If you are getting a response to your options asterisk shouldn't be
marking the peer as unavailable.
is your asterisk behind a firewall?
On 20 March 2015 at 13:42, thufir hawat.thu...@gmail.com wrote:
I wasn't
In article b89aa57122d745a099f87ff3e9260...@cm-ex-v01.cm.local,
Grant Bagdasarian g...@cm.nl wrote:
Is it possible to log the raw signaling of Dahdi channels to a log file?
Try googling for: dahdi pcap
It should be possible to log to a pcap file that you can later examine
using Wireshark. I
I wasn't able to get much out of babytel, beyond the fact that I was,
apparently, sending options which is why I'm not getting 200 OK.
How can I, generally speaking, ping/telnet or otherwise test the
connection to get more data?
A connection to this peer directly from a softphone, Jitsi,
i noticed that when i active the voicemail in the IP-phone where the number
0033149xx is configured i can call this number without issue
Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/FD/0033149xx == Begin MixMonitor Recording
SIP/101-010d
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: 11 March 2015 17:34
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID Names
Are the
/101-0103,
__TIMESTR=20150320-164340) in new stack
-- Executing [s@sub-record-check:20] Set(SIP/101-0103,
__FROMEXTEN=101) in new stack
-- Executing [s@sub-record-check:21] Set(SIP/101-0103,
__CALLFILENAME=out-0149xx-101-20150320-164340-1426869820.301) in new
stack
From a softphone (x-lite) the caller id information comes through as
anonymous@anonymous.invalid
These are also valid calls - If I disable outbound CLID on my mobile and call
in -
this happens. However it works fine on calls where I send caller id
information.
Okay, just figured this
-record-check:18] Set(SIP/101-0103,
__YEAR=2015) in new stack
-- Executing [s@sub-record-check:19] Set(SIP/101-0103,
__TIMESTR=20150320-164340) in new stack
-- Executing [s@sub-record-check:20] Set(SIP/101-0103,
__FROMEXTEN=101) in new stack
-- Executing [s@sub-record-check:21
So you are saying that it resolved the issue to activate voicemail on the
device that sits past your trunk provider? That confuses me a little, but
if your calls are working, that's great news.
On Fri, Mar 20, 2015 at 1:44 PM Salaheddine Elharit
salah.elharit...@gmail.com wrote:
i noticed that
Hello list,
I'm hoping that you could read through this mail and give me some tips
on how to improve my setup (functionality, security, really anything).
It's my first Asterisk installation and meant for simple home use.
I installed Asterisk 11 on an OpenWrt Barrier Breaker router. Currently
Hello,
Is it possible to log the raw signaling of Dahdi channels to a log file?
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