Displaying 20 results from an estimated 1000 matches similar to: "Call picked up from queue and transferred gets disconnected - about 0.01% of calls"
2019 Jan 09
2
Switched from Asterisk 1.8 to 13 - CDR ringtime now always zero (Joshua C. Colp)
Regarding this I've read the specs linked to in detail, but I can find no mention anywhere of any change that implies or states that no ring time will be recorded anymore in Asterisk 13 and that all times in start and answer columns will now be equal for all calls.
Can this be because I nowhere use the Answer() application in my dialplan when dialing out?
-----Original Message-----
From:
2018 Jul 27
3
SHELL() function Asterisk 13 - can only accept one paramter in string?
Hi all
This is a followup on my post "Asterisk 13 - system() dialplan app cannot call bash scripts" from yesterday
I've given up trying to use system() to call BASH scripts with parameters from Asterisk 13.
Turned out under Asterisk 13.22.0 System() DOES work, but only if you do NOT attempt to pass any parameters to the called script.
This works, and reliably calls the script:
2018 May 11
2
Passing parameter to Queue-called macro
Hi Marie
Thanks!
I was just worried about thread safety if I had to use a global variable, e.
g. it might be set to a value by one call (since I'm using the same global
for every incoming call to transfer the accountcode gotten from my HTTP
endpoint to the same macro, and there can be several calls simultaneously
all inserting HTTP-sourced values at more or less the same instant) and then
2020 Jul 01
3
13.22.0 - HTTP session count exceeded 100 sessions - instance unusable
Hi Joshua
HTTP is used on in our setup on
127.0.0.1/mxml?<command>
to send commands to the server, such as
http://127.0.0.1/mxml?action=login&username=myuser&secret=thesecret
to log in and then
http://127.0.0.1/mxml?ActionID=123&Action=BlindTransfer&Channel=Channel&Context=local&Exten=123&Priority=1
etc. to control transfers, for example.
ARI is not being
2020 Apr 21
3
Asterisk 13.22.0 under very high load conditions - freezes in H exten and blocks new calls
Hi all
I'm running an Asterisk on an Intel XEON E5-2660 virtual with Centos 7 -
32GB RAM.
When I approach about 320 channels, I -sometimes- get thousands of these
messages suddenly streamed in the CLI / Asterisk log:
WARNING[60753][C-00022cb9] channel.c: Exceptionally long voice queue length
queuing to Local/xxxxxxxxxx at local-0002dbea;2
WARNING[71993][C-00022dcc] channel.c: Exceptionally
2020 Feb 14
1
Predictive call - agent talking to a customer, then suddenly talking to another customer
Hi, do you have NAT between Asterisk and agent phones?
S pozdravem
Tomáš Holý
Hi Tomas
Thanks for replying.
Yes, the phones are in one location in a LAN and are then NATed to enable them to contact the Asterisk which is hosted in the cloud.
A typical sip.conf phone configuration on the remote server for the site is
[general]
session-timers=refuse
disallow=all
allow=g729:20
allow=ulaw
2018 May 08
2
Passing parameter to Queue-called macro
Hi all
I need to pass a parameter in a thread-safe manner to the Queue pickup
macro. This is to know when (and who) picked up an incoming call to a queue
and log that to my back-office system with a CURL to a HTTP endpoint.
However, the Queue application does not appear to allow passing of
parameters to the called queue pickup macro.
E. g. non-working code is:
[queuetest]
timeout = 60
retry =
2017 Jun 30
3
asterisk.conf ignored?
Hi all
I'm trying to limit the maximum concurrent calls on my Asterisk to try and
mitigate another problem I posted about earlier.
I've edited
/etc/asterisk/asterisk.conf
And uncommented this line, and put a value of 60 in there:
maxcalls = 60
in an effort to limit my Asterisk to 60 simultaneous calls.
I did a
core reload
in the CLI after doing that.
2020 Feb 12
1
Predictive call - agent talking to a customer, then suddenly talking to another customer
Hi all
Asterisk 13 instance - I've got a situation in an agent queue that an agent
will be talking to one person, then suddenly the same agent will be talking
to another person who was talking to another agent.
The calls do not switch around between the two agents, the "losing" agent
will just suddenly have silence in his handset and the other agent will now
be talking to
2015 May 27
2
Strange and complete failure of Asterisk 1.8
Hi all
We've had a very strange failure on an Asterisk 1.8 install that has been
running for about a year at a customer site.
The physical hardware is fine, all other services off the Centos 6.5 server
are running. Only Asterisk is not working...
The first symptom was that no calls can be made over the SIP phones used
with it, and no calls could be received over the SIP trunk connected to
2015 Jan 07
3
Asterisk executable suddenly about 40KB larger - modules not working
Hi all
I have a strange issue with 1.8.11.0 on a production Asterisk machine at our
head office, and the same issue with a production machine at a branch
office.
Every now and then, on the head office machine, ODBC CEL and CDR logging
will stop working. On examination in the CLI, Asterisk behaves as if the
config files for ODBC in the /etc directory are just gone.
Repeated tests have then
2015 Mar 02
1
System() command refuses to execute bash script
Hi All
I'm using this extension to try and get Asterisk 1.8.11.0 to run a bash
script:
exten=>802,n,System(/bin/sh -f /root/wireless.sh)
This file is
-rwxr-xr-x 1 root root 171 Mar 2 16:23 wireless.sh
e.g. root owns the file, and it has execute permissions for all users.
Asterisk runs as root as well.
Asterisk executes the command without any errors at max verbosity.
The file
2020 Jul 01
1
13.22.0 - HTTP session count exceeded 100 sessions - instance unusable
Hi Joshua
No back-off, but I am caching the last 5000 results and and first hitting the cache to see if a recent command already provided the information I'm seeking for a particular request.
I'll see if I can do some simulation and see if I'm effectively DDOSing the local HTTP interface.
I'll have to see if I maybe have a resource leak in my code that makes the HTTP request,
2014 Dec 12
1
Corrupt MixMonitor recordings - .gsm format
Hi all
Asterisk 1.8.11.0 on Centos 6.5
My VOIP phones are using G729 to a G729 trunk from a vendor (Centracom,
South Africa). Unlicensed G729 codec version on server.
75% of my .gsm files from MixMonitor are coming up corrupted about 3 minutes
into the recording.
The server has been up for 7 months beforehand with no problems with
recordings to .gsm format files.
I noted
2020 Jul 01
2
13.22.0 - HTTP session count exceeded 100 sessions - instance unusable
Hi all
I'm running an Asterisk 13.22.0 instance on Centos 7 - I7-8700 12 core HT
with 16GB of RAM.
The server maintains a total active call count of approx 285 calls with 440
channels at any one time. The totals never go below 200 calls concurrently
active.
For the kernel, top reports load levels at around 5.0 to 6.0 constantly.
I'm having issues where at random intervals, the CLI is
2015 Aug 13
2
One way audio - doesn't seem to be NAT issue
Hi D'arcy
Have you checked your RTP port ranges (I'm sure you have), and also that the
server IP for RTP as specified in the initial SIP is correct?
Not sure how this will relate to your setup, but we had something similar
here using Asterisk 1.8.11.0 on both sides of the connection, via a VOIP
service provider in the middle.
We had slightly different parameters, e. g. that we would
2018 Jun 09
3
getting real sip status after dial
Hi,
Is there any way I can get exact sip status from pjsip after a dial ?
or all we can
get is asterisk hangup causes ?
Thanks in advance.
KKh
2018 Jun 09
2
getting real sip status after dial
I think HANGUPCAUSE is channel agnostic.
See: core show function HANGUPCAUSE
Some thing like this IIRC:
Set(my_cause=${HANGUPCAUSE(${CHANNEL(name)},tech)})
Remember the incoming leg of the call and the outgoing leg of the call
are different channels. Make sure you are giving HANGUPCAUSE the
correct channel.
On 06/09/2018 02:01 PM, Khalil Khamlichi wrote:
> It seems very weird to me
2017 Dec 19
2
asterisk queues in off-hook mode ?
Hi,
I am looking to configure asterisk queues in off-hook mode, that is,
the agent calls into the system and stays connected to this call, when
new customer calls, he is redirected to the queue which should
distribute to connected agents. is this possible on teh actual
app_queue or we would need to implement it using ARI.
Thanks in advance.
2014 Oct 13
3
samaba 4 vs active directory
Hello Sir ;
I am very interested for samba 4 news and want to run it in production environment to replace active directory
But first I appreciate if you could help me to get comparison between active directory feature and samba feature that working correctly
I mean if I go to samba which feature may I lose or still have bugs
Thanks in advance
Kind regards,
Ahmed Hassanean Khalil
Customer Service