Displaying 20 results from an estimated 10000 matches similar to: "Unable to include switch 'Realtime/@' in context"
2011 Mar 02
1
Asterisk 1.8 SIP realtime and NAT
Hi
After recently upgrading to 1.8.3 I have noticed that the nat setting
for my peer in my sip table is not making it into the realtime cache.
For example
* Name : 501
Realtime peer: Yes, cached
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : pack-local
Subscr.Cont. : <Not set>
Language :
AMA flags :
2010 Dec 17
2
Asterisk Freeze In 1.4 realtime
Has anyone seen the following in 1.4 (1.4.17)
We have istances when the number of sip channels in use multiples up
(eg: we have 40 channels in use, and then it will jump to 80, then 100+
and it will keep going upwards) and in doing this, all the channels
which are in use at that time are simply cut off or frozen.
The only way for us to get everything back to normal is via a hard
restart of
2010 Jun 14
0
Hint priority in RealTime
Hi
I've just had a request from a customer who wants to use Busy Lamp Feed.
I've had a look around and it would appear that you have top use the
'hint' priority. We are using asterisk 1.4.17 with realtime and the
priority column in the extensions table is a tinyint so obviously I
can't put hint in there.
Has anyone any experience of working round this problem?
Thanks
2009 Jul 10
1
Lagged Extension
Hi There
I have an extension which is in a different country and is constantly
lagged (about 800ms). When anyone tries to call this extension we get a
No route to destination message.
Now I would have thought that the server should be able to find a route
to the destination seeing as the peer poke finds it's way there. Or is
that lag too much to create a SIP channel?
Thanks in advance
2011 May 19
3
Manager logged on/off messages
Hi
Is there a way I can stop Manager logged on/off messages from going to
the console/logs without losing all the other information I need?
Regards
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
2010 Feb 10
1
Muted calls occasionally dropping after 30 seconds
Hi
I'm having a very odd phenomenon happening on our production server
(1.4.17 and using realtime). Sometimes a call will disconnect 30 seconds
after the SIP phone hits the mute button but it doesn't happen all the
time. I've done a sip debug while watching this happen and that doesn't
show anything other than a BYE message being sent out of the blue.
The rtptimeout and
2013 Dec 05
1
Lync and Asterisk Realtime Architecture
Hi guys
We're using asterisk 1.8.23.1 on CentOS 5 and are trying to create a trunk
to MS Lync server.
If I create the peer in sip.conf the trunk connects with no problem.
However, we prefer to use ARA.
Whenever we define the peer in our peers table, the trunk does not work,
even if we use sip show peer <peer-name> load.
Has anyone got any experience of connecting to Lync using ARA?
2009 Aug 25
2
Authenticating SIP peer on IP address only
Hi
I know this is far from best practice but is it possible to authenticate
a sip peer on the IP address it's coming from so that it doesn't need to
use a UN and Pass?
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
2009 Nov 10
1
Call audio leaking between calls
Hi
Has anyone ever had experience of phones on the same office network
being able to hear other concurrent call's audio whilst on calls of
their own? We're getting this for the first time and I'm at a bit of a
loss as to where to start to look.
We're using 1.4.17
Any pointers would be much appreciated!
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660
2011 May 23
1
AJAM XML output not valid xml
Hi
I'm using asterisk 1.8.3.2 and have been implementing AJAM. I've noticed
the final '>' is missing from every response I've had so far. Here is an
example
<ajax-response>
<response type='object' id='unknown'><generic response='Success' message='Authentication accepted' /></response>
</ajax-response
Has anyone
2009 Oct 16
1
Check if a variable is set
Hi
Is there any way to check if a variable is set in asterisk? I've had a
look around and can't find a purpose built function for it.
I'm going to be using it to see if an argument has been passed with a
macro or not (e.g. see if ${ARG3} is set or not)
Thanks in advance
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
2010 Dec 15
1
Transferring problem within Queues
Hi
We are using asterisk 1.4.17 for the apt repository on an Ubuntu server
and we're getting an odd problem with one customer using a Queue
The queue is called in the dialplan with the options Tn
The queue only has one member.
Occasionally and starting to get more frequently the caller ends up
being initially answered by the wrong extension (i.e. one that is not a
member of the queue)
Has
2011 May 20
1
*8 pickup and CLI presentation
Hi
When we use the *8 feature to pick up a call on another extension, the
phone will only display *8 and *8 is what is stored in the phones
memory. Is there anything we can do so that when we use *8 the incoming
caller's CLI will be presented on the screen of the phone and in the
phones memory?
We are using Snom phones but I'm sure this is an asterisk rather than
phone issue...
Thanks
2011 Dec 23
1
GotoIfTime days query
Hi
I'm using 1.8. Is there a way you can specify staggered days in a single
GotoIfTime command e.g. mon|wed|fri?
Thanks in Advance
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
2009 Nov 16
2
Odd Local Channel and 0 billsec issue
Hi
I've been noticing an odd issue with our servers (1.4.17) where a large
number of one particular customer's (we operate a hosted VoIP platform)
calls go through a Local channel rather than the SIP channel and
whenever this happens our asterisk CDR is recording a billsec value of 0.
Our outgoing calls to POTS are sent through a separate carrier and we
get a daily CDR off them in
2010 Sep 29
2
Alert-Info advice
Hi guys
I'm using asterisk 1.4 and going on to Snom phones. I'm trying to add a
sip header to make the Snom phone use a different ring tone on one
particular incoming number. I have added the following to the dial plan
of the incoming context
+------+------------------+-------+----------+--------------+-------------------------+
| id | context | exten | priority | app
2010 Apr 13
2
Full transfer details on inbound calls
Hi
We're using asterisk 1.4.17 using RealTime and my boss has decided that
we should keep a track of the full history of incoming calls i.e. who
and when they were transferred to. The asterisk CDR only holds the
initial answering channel for any call and not any further transfers
that may have happened.
The idea we are toying with is getting the time and the originating
channel from the
2010 Jul 01
2
Brute force attacks
Hi
We've just noticed attempts (close to 200000 attempts, sequential peer
numbers) at guessing peers on 2 of out servers and thought I'd share the
originating IPs with the list in case anyone wants to firewall them as
we have done
109.170.106.59
112.142.55.18
124.157.161.67
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
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2011 May 26
3
UK English sounds packs
Hi
Does anyone know if there are any free UK accented English sounds packs?
Thanks
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
2011 Aug 02
3
MixMonitor and attended transfers
Hi
I'm using asterisk 1.8.3.2 (with a couple of patches)
I have the following scenario...
SIP call comes in and gets answered by extension A (MixMonitor is
executed as part of this inbound dial plan of the number being called)
Extension A puts call on hold and calls extension B
Extension A then does an attended transfer of incoming call to extension
B
I'm finding that the recording