Displaying 20 results from an estimated 5000 matches similar to: "AGI script VERBOSE cmd"
2011 Jan 24
6
ReceiveFAX issue.
I am testing out inbound faxing using res_fax and res_fax_spandsp.so
My system answers the call but then sets there on the ReseiveFax line then
comes back with an error that it exceeded the maximum retries.
How would I go about debugging this? Below is my very simple dialplan code
I am using, and the fax show version gives the following as well.
FAX For Asterisk Components:
2013 Nov 25
4
Voicemail greeting playback issues?
Hey all
I have been beating on this all weekend long.
Any feed back would be appreciated.
We stood up a 11.6 system. We tested everything we could think of.
We moved over to it and all seemed to be working good than a customer told
us that they were not hearing our vociemail greetings.
When we call into the system and it drops to voicemail we just get a beep
no greeting played. We checked
2011 Jun 14
2
Voicemail issue
Hey all
I am having instances where voicemail boxes will have a 00001 message and
no 00000 message this causes the user to be told that they have a message
that they can't get at. If I renumber the messages manually to start with
the 00000 numbering then the user can get their messages. What could be
causing this and how can I get it out of the system.
Is there a patch I can apply to the
2015 Oct 16
2
pjsip database error when using MS SQL via ODBC
I have a project that is requiring the use of MS SQL from asterisk. I get
an error when the pjsip contact tries to update the contact table.
[Oct 15 21:34:55] WARNING[3033]: res_odbc.c:649
ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 22018:
[FreeTDS][SQL Server]Conversion failed when converting the varchar value
'3.000000' to data type int. (101)
The datatype
2012 Aug 13
8
Asterisk hangs while starting in OpenSuse 12.2
Hi,
I am using OpenSuse 12.2 64bit OS which uses Kernel 3.3.x version and
downloaded Asterisk 1.8 current version, after installing Asterisk, while
starting Asterisk it hangs at the stage of loading modules.conf, I checked
the forum https://issues.asterisk.org/jira/browse/ASTERISK-19245 but still
after updating through yast also i am facing the issue.
Have anybody faced this type of issue with
2017 Sep 14
2
Realtime pjsip issues
We are having an issue where on the latest version of asterisk when
configuration pjsip via realtime.
we do a pjsip list endpoints it shows our endpoints but lists them as
invalid.
When we do the pjsip list endpoints again it shows no objects.
This applies to pjsip list aors as well. We did not have this issue on
our older asterisk 13 installs. My guess is something has changed
2010 Dec 01
6
Issues with 1.8 and BlindTransfer
I am having issues with Blind Transfer on asterisk 1.8
If I call from one Grandstream phone to another and us the transfer key
to do a blind transfer everything works fine.
When calling in on a sip trunk and then trying to use the transfer key
to transfer from Grandstream phone to Grandstream phone the call just hangs
up. It did not do this on Asterisk 1.4.x or 1.6.2.x . If we use
2015 Mar 12
2
GXP 1405 and asterisk
Hi list, someone has successfully change different ringtone from
dialpan with asterisk with this model Granstream?
for example:
exten => 0,1,Playback(pls-wait-connect-call)
same=> n,SIPAddHeader(Alert-Info:;info=ring3)
same=> n,Dial(SIP/310&SIP/318,30,t)
can not get it to work
any idea o tips?
regardss
--
rickygm
http://gnuforever.homelinux.com
2013 Jan 17
2
Mail list settings?
Hey all
For some reason the mailing list is sending all messages from the sending
party.
This makes it less than ideal when responding; as selecting reply goes to
the person and not the list.
Can we have it set back to the old way please?
Thanks Andrew for pointing this out to me.
Bryant
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2011 Mar 09
6
SIPAddHeader not working
Hello list,
I notice that the dialplan method SIPAddHeader is not working :
in dialplan :
/exten => s,n,SIPAddHeader(Privacy: id)/
in SIP invite no trace of this header :
/INVITE sip:0473 at sip.domain.be SIP/2.0
Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97
From: "VC" <sip:voip2 at sip.domain.be>;tag=729476652f511c67o2
To: <sip:0473 at sip.domain.be>
2011 Apr 06
11
Asterisk 1.8.3
I have deployed several 1.8.3.2 systems as upgrades of customers systems
and now I am seeing random crashes. For some reason the builds lock up and
stop taking sip connections. Existing calls stay on but when the user hangs
up no new calls or reg attempts work. In most cases a "core restart now"
cleans things up. Some times I have to kill the asterisk process. The
stability of 1.8.2
2011 Nov 03
15
DID from Direct from Telco
Hello Everyone,
Unlike going through DIDx, DIDLogic etc.., we have an option of
getting DIDs directly
from local telco Bell Canada. Currently our SIP Trunk provider
assigned a DID to us,
and as you know, they just redirect requests it to our PBX.
However, when dealing directly with a telco, what equipment will we
need? Basically
giving us the same capability as a DID provider. If someone can
2010 Sep 13
5
Force ip disconnect after register?
Is there a way to drop a ip connection to asterisk after a number of
register attempts.
I have been having issues with hackers doing registration scanning against
our server. We block their address at the fire wall but since asterisk does
not force a drop of the connect after so many bad reg attempts I can't
enforce the block until they drop and try again. This allows them to run
the box
2015 Mar 12
2
Unstable phone connection
This is driving me to distraction. I have a switch with multiple
clients who are all working fine except for one and I can't figure out
what makes them different. I have tried every NAT setting in the ATA
(SPA112 ATA with 2 x FXS, 1 x LAN), stun server on and off, different
sip ports, different RTP ports and it still fails. I have left the
location with it working only to have it fail
2015 Oct 16
2
pjsip show xxxx like endpoint?
Is there a way to limit the items returned by pjsip show [type] using like
chan_sip allowed for sip show peers like xxxx, but I can't seem to figure
out how to lookup or limit my returns with pjsip
Thanks
Bryant
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2011 Jul 28
5
MoH - conversion command
Hi,
I've been trying to get MoH files to sound decent. I've got a hold of
Royalty-free Classical music (a safe choice for most of my customers) and
I`ve been trying to convert them to the normal telephony/Asterisk format
using sox. Unfortunately, it sounds really bad. I don't expect concert hall
quality of course, 8000KHz being what it is, but is there a better way to
convert
2010 Dec 20
3
cdr_mysql stopped working
I did an upgrade to the SVN trunk on the 12/9 and when I looked in my mysql
table for CDR's today there are no entries since the update.
I have rebuilt and re-installed and re-started asterisk still no CDR's
flowing to mysql. I did not change any configs. I checked to make sure that
the cdr_mysql option was selected under the make menu options. The module
shows it is there when I do a
2016 Mar 31
2
Asterisk 13 - Call Bridge issue.
I have the following senerio.
Call file calls 1st party.
When connected give called party option to connect to second party.
Issue Dial to second party. Caller answers and the two are bridged
together.
My issue is that 4 out of 5 calls fail to bridge the audio.
Am I missing something or is there some kind of bug? Here is my test
dialplan
;Dialer Base Code Files.
;Variables
2012 Feb 09
4
checking if a phone number is UP
hi,
We have a phone number from third party provider which is used for inbound
calls. How could I monitor if this *phone number* is reachable?
the initial idea doesn't sound elegant:
- on my SIP server I set couple seconds of ringing before Answer().
- the monitoring server calls to that phone number for few seconds, checks
if it "hears" the ringing and hangs up the call.
**
I use
2011 Dec 21
3
Suppress -- Remote UNIX connection message
We have written some monitoring and stat collection scripts that use
asterisk -rx "command" The script runs once a min and logs data and posts
any critical notifications. Everything is working well with this method
but we get the -- Remote UNIX connection / disconnect message once a min
and we would like to suppress it. Is it possible without reducing the
verbose logging level.