Displaying 20 results from an estimated 5000 matches similar to: "Odd Local Channel and 0 billsec issue"
2014 May 02
1
CDR billsec issue with calls forwarded through the Local channel
Hi
I'm using asterisk 1.8.23.1 but I've seen this same issue in previous
versions of 1.8. I have created some work arounds but the behaviour is
incorrect.
This is the scenario:
Call comes in and goes to appropriate dialplan
In the dialplan the call is forwarded to another number using a Local
channel (and using /n ) e.g.
Dial(Local/<my-number>@outbound-context/n,60)
The number is
2009 Aug 20
8
mysql sip realtime
Hi
I have some question about mysql realtime.
1) Anyone know exactly if there is a specific order to declare sip table
column for realtime ? In which file can I find that order ?
2) In my extconfig.conf, [settings] are :
sipusers => mysql,general,siptable
sippeers => mysql,general,siptable
so means that I use realtime dynamic exactly ?
Is it normal if some parameters from sip.conf still
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
2011 Feb 28
2
Asterisk 1.8.3-rc3 and one way audio
I've just installed 1.8.3-rc3 on a test server as we really needed that
deadlock involving REFER fix on our server but now I'm having an odd
issue with one way audio with a specific type of call.
If I do extension to extension calls there is full 2 way audio.
If I route in an incoming call through numbers provided by our SIP
provider there is no inbound audio (mobile to * SIP extension)
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
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 Aug 11
5
Trouble with *8 Pickup
We have a client that has sporadic problems with the *8 pickup facility.
The server they are using is 1.8.5 and they are using Snom phones.
Every now and then when they try to do a pickup from another phone they
get a forbidden message on the phone and I can see the following in the
logs.
[Aug 8 11:51:53] ERROR[19314] astobj2.c: user_data is NULL
[Aug 8 11:51:53] ERROR[19314] astobj2.c:
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
2009 Nov 23
1
Is Answer really needed
Hi
All my incoming dial plans start of with an Answer which I now know
starts the billing time. Some of the dialplans then get forwarded out to
POTS via a carrier and so the actual amount of time that should be
billed is being distorted.
I've done a few tests this morning and found that if I don't start with
an answer then the billsec of my forwarded call is actually the length
of
2010 Jul 28
2
Answered call not bridged
Hi
I've suddenly started encountering a strange issue. Sometimes, when a
call is made into our system, an extension answered the phone but I can
see no mention of it being bridged in the console. Also, the server does
not seem to think that it is answered and then goes to voicemail. We are
using asterisk 1.4.17
Here is the console output for one of these calls, it was me ringing a
2010 Mar 29
5
Continue a dialplan when the client hang up the call
Hi all,
When a user make a call to Asterisk, and when user hang up the call at any point of the conversation,? Asterisk will stop Diaplan intermediately.
At this situation,? Are there any way to make? Asterisk continue execute the Diaplan ?, so Asterisk can do something like that delete temporary file, .. etc.
Thanks in advance,
Giang
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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
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
2011 Jan 05
1
Blind Transfer not working - 1.4.38
Hi
We've been running asterisk 1.4.17 (deb package) in a production
environment for some while now and are finally taken the plunge to
update to 1.4.38 (Ubuntu servers). All of this is using the RealTime
Architecture
I have upgraded the asterisk version in one of our test environments and
blind transferring seems to have suddenly stopped working. It was
working fine under 1.4.17
So, call
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
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
2011 Feb 11
3
Asterisk 1.8.3
Hi
Does anyone have any rough idea how far away 1.8.3 is?
We can't deploy 1.8 yet because of this issue
https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
2011 Feb 22
3
Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available
The Asterisk Development Team has announced security releases for Asterisk
branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are
released as versions 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4.
These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases
The releases of Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and
2011 Feb 22
3
Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available
The Asterisk Development Team has announced security releases for Asterisk
branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are
released as versions 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4.
These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases
The releases of Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and
2011 Apr 13
11
Realtime SIP & peer status
Hello,
I'm using SIP realtime with MySQL DB.
Is it possible to get the status of the SIP peer (free / calling) from
this realtime DB ?
If not, is there another way to obtain the call state of a SIP peer ?
Kind regards,
Jonas.
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