2 same sip extension number on 2 asterisk - call not passing on certain condition

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Hi list, something crazy here. 2 asterisk on 2 different place (1.4 and 1.8) both
having an extension [115], one as type peer (caller side 1.4) and one as
friend (callee side 1.8). Phones from both location connect to Asterisk
from LAN. Router are Linux boxes. Connection between the 2 sites is done like this: On the callee side [115] ;callee
type=friend
host=dynamic
secret=otherSecret
context=local
nat=no
canreinvite=no
qualify=no
dtmfmode=rfc2833
allow=all
call-limit=1
busy-level=1
allow=all [Caller]
type=peer
host=voip1.domain.net
deny=0.0.0.0/0.0.0.0
permit=xxx.xxx.xxx.xxx
context=myAccess

Asterisk Users 3.8 years ago 0 Answers

Question about Read() application

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Hello again list, Did the following: (on 1.4.42 installation) asterisk -rx "core show application read" -= Info about application 'Read' =- [Synopsis] Read a variable [Description] Read(variable[|filename][|maxdigits][|option][|attempts][|timeout]) Reads a #-terminated string of digits a certain number of times from the user in to the given variable. filename -- file to play before reading digits or tone with option i maxdigits -- maximum acceptable number of digits. Stops reading after maxdigits have been entered (without requiring the user to press the '#' key). Defaults to 0 - no limit - wait for the user press the '#'
key. Any value…

Asterisk Users 3.8 years ago 8 Answers

DTMF dropping in Read Command

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Hello listers, I have a couple of 1.4.37 through 1.4.42 boxes
running at different sites. These systems run a fairly simple IVR that uses
waitexten and Read to get credentials and plow on through a set of contexts.
I am experiencing two problems in my setup: 1. In environments where DAHDI is the trunk of choice, this snippet
drops digits, especially if the user attempts to key them quickly. It also
mysteriously hangs up (autofallthrough=yes) if they key them too slowly exten => s,1,Set(chktime=0) exten => s,n,Read(digitacct,entacct,17,skip,1,10) exten => s,n,Gotoif($[${LEN(${digitacct})} < 1]?checking,t,1) exten => s,n,Read(digitpass,entpin,5,skip,1,10)…

Asterisk Users 3.8 years ago 0 Answers

how to debug dropped calls?

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I've been experiencing a number of dropped calls - both where I'm
calling out and the call drops before answer, and where it's inbound and
the call drops while I'm talking (usually at almost exactly 5 minutes). I'm using dahdi 2.5.0.1 with a TDM400P connected to PSTN. The console doesn't show anything ( dahdi 4 is PSTN):

Asterisk Users 3.8 years ago 0 Answers

How to unregister a sip trunk

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Hi
In our asterisk we have more than 3 sip trunks.When we do "sip
reload" from CLI/AMI we need to unregister all 3 trunks and register
available trunks only. Now in asterisk unregisteration is not
happening.Please help on this Thanks
NIkhil

Asterisk Users 3.8 years ago 1 Answer

ast_debug messages not showing up

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Hi, I'm running Asterisk 1.8.7.1 on Gentoo. I set `core set debug 9` but don't
see any debug messages on the console. I do get the verbose messages from
ast_verbose. Is there something I need to configure to see these messages? Regards,
Yahya

Asterisk Users 3.8 years ago 0 Answers

Fax not detected by Asterisk

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Hello List, I have an Elastix 2 machine with digium fax modules (with license).
When I try to create an extension that also works with FAX, Asterisk does
not detect any incoming fax. Even when I use 'fax set debug', it does not
display anything.
It's Asterisk 6.2.x . Any ideas what can I do to figure out why it does not
detect the arriving faxes ? Thanks, Ido

Asterisk Users 3.8 years ago 5 Answers

Use Polycom FX with Asterisk

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Hi List,
I have a Polycom FX video unit and I'm thinking maybe I can integrate it
on our Asterisk Server to be able to do teleconference and video as well
via Polycom FX. I already have oh323 configured on my Asterisk box and I just no idea on
how to let them work.Any help please? Regards,
Malvin

Asterisk Users 3.8 years ago 0 Answers