Displaying 20 results from an estimated 1000 matches similar to: "asterisk-users Digest, Vol 72, Issue 81"
2010 Aug 05
1
Can ChanIsAvail return status from sip uri using router ip
hello,
Although my previous posts in this forum have not received satisfying
answers, here is another question from me.
my question is can i use ChanIsAvail function to get the status of a user in
the format SPI/user-id if i provide user in sip uri like this
ChanIsAvail(SIP/user at 153.18.x.x:5062)
calling user with this sip uri works fine.
I once tried but status returned was "unknow
2010 Jul 20
0
asterisk-users Digest, Vol 72, Issue 49
sorry for typo mistake in my last post. as from my orignal post two
registration of the same user are as follows
SIP/XYZ at 119.68.0.90:5060
SIP/XYZ at 202.16.34.10:5678
so dial command with unique-id i want to use will be
Dial(SIP/XYZ at 192.168.0.20:5062-096afee8,30,rtT)
Dial(SIP/XYZ at 192.168.0.12:64290-0966ab80,30,rtT)
and not
Dial(SIP/192.168.0.20:5062-096afee8,30,rtT)
2010 Jul 16
0
asterisk-users Digest, Vol 72, Issue 39
yes, actually this scenario is on remote servers. like
SIP/XYZ at 119.18.230.20:5060
SIP/XYZ at 202.68.0.90:5678
audio is ok when dialing without using ip & port as below
SIP/XYZ
but when i dial using below dialstring
SIP/XYZ at 202.68.0.90:5678
or
SIP/XYZ at 119.18.230.20:5060
then the problem arises
hope you got the idea..
Nasir
2010 Jul 29
4
How to extract channel-id of a user or peer
Hi,
my question is how can i get channel-id of a user or peer. I tried using
ChanIsAvail(username). this works correctly when user and asterisk are on
Local LAN. But my asterisk server is on public ip and users are behind nat,
and so this method says unknow host when used on public asterisk server.
I also tried built-in variable ${CHANNEL}, but this returns the channel-id
of the calling channel.
2010 May 11
1
asterisk-users Digest, Vol 70, Issue 24
Yes this scenario works on my 2 systems which are at LAN. I made one system
as server (192.168.0.20) and registered from other system... it is fine but
now there is a different scene.
actually there is a registered user named abc at system1 (192.168.0.20)
having context [payasyougo] which is used to do outbound calls. we want to
use this user's context and account so that when we register
2010 Jul 30
1
asterisk-users Digest, Vol 72, Issue 82
thanks for your reply but i think ${BRIDGEPEER} will work only when both
channels are connected. i want to get channel-id before dialing so that i
can dial using that channel id.
> ${BRIDGEPEER} is probably a good way to do what you want.. if Channel
> A calls Channel B, and you want Channel A to "get" the channelID of
> Channel B, as long as the two channels are bridged,
2010 May 11
1
asterisk-users Digest, Vol 70, Issue 23
Thanks Vardan,
I will like to know if this scenario can work when peer is not having fixed
ip and we use
host = nasir.server.com
?
also I have set insecure=invite,port
what if i use
insecure=no
thanks again.
Message: 24
Date: Tue, 11 May 2010 10:52:14 +0500
From: Vardan <hvardan71 at gmail.com>
Subject: Re: [asterisk-users] Dialing a SIP Peer without using
register strin
To:
2010 Jul 22
0
SIP URI Dial has one way audio
Hi,
I am trying to dial a sip user via his IP:PORT Combination. i am using XYZ
as target user which is registered.
Asterisk server IP: 70.118.x.x
calling user IP: 117.58.x.x
called user IP: 117.58.x.x:5062
First I dialed my registered user in normal way like this,
Dial(SIP/XYZ,30,rtT)
and during conversation audio was OK in both ways. Then I dialed the
registered user via
2010 May 12
2
asterisk-users Digest, Vol 70, Issue 25
Hi Vardan
I did same as you told and deleted the SIP information in Astdb and
restarted asterisk. but the result was same.
as you said there might be mistake in sip.conf so i am pasting both servers
configuration here..
1- nasir.server.com
[abc]
username=abc
type=friend
secret=mysecret
nat=yes
mailbox=12234568
incominglimit=2
outgoinglimit=2
host=dynamic
dtmfmode=rfc2833
context=payasyougo
2010 Aug 03
0
asterisk-users Digest, Vol 73, Issue 5
Hi C F
no asterisk and sip device are not behind same router. actually both are in
different countries. how ever when caller and callee are behind same routers
voice is just fine (both ways) and i can see re-INVITEs too.
but when someone calls from another router then this issue arises. caller
can hear the called party but called party can not hear caller. and there
are no re-invites issued
2010 May 13
0
asterisk-users Digest, Vol 70, Issue 30
sorry, you r right i just checked it with registration so there were astdb
entries for SIP registration.
anyhow after clearing settings frm astdb i tried the same scenario you
advised but no luck.
I think i told that i am not using server as peer but want to use a user
[abc] as peer so that when ever i use
dial(SIP/${EXTEN}@abc) or dial(SIP/abc/${EXTEN})
the call will be out from server using
2010 Jul 15
6
One way audio when dialing multiple registrations
Hi,
I am working on calling 2 registrations of same user on 2 different ip or
ports. It works fine and both phones ring simultaneously. the problem is
that there is one way audio, calling party can hear me but i can't hear
calling party.
here is the scenario..
SIP/XYZ at 192.168.0.20:5060
SIP/XYZ at 192.168.0.10:5678
i dial using following dial string
Dial(SIP/XYZ at
2020 Apr 30
0
Certified Asterisk 16.8-cert1 Now Available
The Asterisk Development Team would like to announce the release of Certified Asterisk 16.8-cert1.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/certified-asterisk
The release of Certified Asterisk 16.8-cert1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following
2020 Apr 30
0
Certified Asterisk 16.8-cert1 Now Available
The Asterisk Development Team would like to announce the release of Certified Asterisk 16.8-cert1.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/certified-asterisk
The release of Certified Asterisk 16.8-cert1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following
2010 May 10
1
Dialing a SIP Peer without using register strin
Hi,
I am new to this list and this is first time i m posting here. please help
me out
currently I am working on dialing a sip peer on an asterisk server from 2nd
asterisk server. scenario is like this.
on my system i am using this peer in sip.conf.
[abc]
type=peer
username=abc
secret=mysecret
host=192.168.0.20
context=default
dtmfmode=rfc2833
;restrictcid=no
canreinvite=yes
2010 Aug 03
2
RTP stream not passing through router with port forwarding
Hi,
I am trying to dial a registered user via his IP:Port mechanism, but problem
is that the audio data is not reaching to dialed user. here is the scenario.
caller and callee both are registered at asterisk server. asterisk server is
on public ip so no port forwarding and natting necessary there. however
caller and callee both are behind router and there is port forwarding
enabled and nat=yes,
2004 Aug 23
1
using ChanIsAvail
Hi
I am trying to use ChanIsAvail to decide if a particular extension is
available in the sip channel
I am using MySQL to hold my SIP friends.
and wy cvs version shows Asterisk CVS-08/02/04
my intention is, that if the extension is not available in Sip channel, I
will send the call somewhere else
my extensions file contains the following:
exten => _[123]XX,1,ChanIsAvail(sip/${EXTEN})
exten
2004 Jun 23
1
Problem with incominglimit and outgoinglimit
Hi,
I seem to have a problem with chanisavail and the call limits on sip
phones(incoming and outgoing)
The problem seems to be that chanisavail when trying create to create
channels and hanging them up afterwards screw up the current usage limit on
the phones.
Example with chanisavail:
Phone A calls voicemail (usage now 1)
Phone B tries to call Phone A and uses ChanIsAvail in the dialplan.
2010 Mar 04
0
Availstatus returns 20 ?
Hello list.
ChanIsAvail returns 20 for ${AVAILSTATUS}. What does this '20' mean ??
...
exten => 1,n,ChanIsAvail(SIP/sin10)
exten => 1,n,NoOp(chanisavail == ${AVAILSTATUS})
...
[Mar 4 15:10:16] -- Executing [1 at sin:7]
ChanIsAvail("IAX2/testlocal-14088", "SIP/sin10") in new stack
[Mar 4 15:10:16] -- Executing [1 at sin:8]
2007 Jan 24
1
ChanIsAvail kills dialplan processing when no Zap available on 1.2.14.
Hi, I'm trying to use ChanIsAvail to build a resilient 'dialout' macro.
The logic is simple; try Zap/g1 (a group of two E1s), and if that
fails, try locating a channel via DUNDi. Here's a massively cut down version
to illustrate the problem I'm having.
macro dialout ( dest ) {
ChanIsAvail(Zap/g1);
noop(Value of AVAILCHAN is ${AVAILCHAN});