[asterisk-users] as soon as Phone rings I'm disconnected yet phone rings two more times‏
One of the things I'm trying to do it to connect my 8x8 DTA 310 terminal
adapter onto my asterisk. I have the 8x8 box connected to the Internet, and
the phone line connected to an fxo port on a Cisco router:
voice-port 0/2/0
connection plar opx 5000
caller-id enable
dial-peer voice 200 voip
destination-pattern 5...
session protocol sipv2
session target sip-server
codec g711ulaw
!
sip-ua
sip-server ipv4:172.16.200.212 -- Asterisk server
When I make a call from the PSTN to the 8x8 box, it does send ring back to the
asterisk server and the Digium phone does ring. However, as soon as the phone
rings the call disconnects yet the actual phone, extension 5000, rings two
times before it hangs up, also.
The following output is what I see on the Asterisk console:
asterisk*CLI
== Using SIP RTP CoS mark 5
[Oct 18 16:27:46] NOTICE[1513]: chan_sip.c:23352 handle_request_invite: Call
from '' (172.16.200.1:65451) to extension '5000' rejected because extension not
found in context 'default'.
== Using SIP RTP CoS mark 5
-- Executing [5000@pstn-incoming:1] Dial(SIP/172.16.200.1-0006,
SIP/5000,20|p) in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/5000
-- SIP/5000-0007 is ringing
== Spawn extension (pstn-incoming, 5000, 1) exited non-zero on
'SIP/172.16.200.1-0006'
The 172.16.200.1 is my router.
sip.conf excerpt:
[5000]
type=friend
context=phones
host=dynamic
disallow=all
allow=ulaw
secret=cisco123
mailbox=5000@phones
[172.16.200.1]
context=pstn-incoming
type=friend
host=172.16.200.1
dtmfmode=rfc2833
disallow=all
allow=ulaw
[phones]
exten = 5000,1,Dial(SIP/${EXTEN},20|p)
exten = 5000,n,Hangup
[pstn-incoming]
include=phones
Any help would be greatly appreciated,
Thanks,--
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Re: [asterisk-users] IAX Trunk issue. (Dale Noll
Dale,
Sorry for taking so long to answer, I've been traveling.
Thanks so much for the suggestion, your solution worked perfectly. I'm not
sure why I didn't notice that the IAX trunk was working in the other direction.
Once again, thanks for your help.
Mitch
Date: Mon, 25 Jun 2012 05:44:37 -0500
From: Dale Noll dn...@wi.rr.com
Subject: Re: [asterisk-users] IAX Trunk issue.
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: 4fe84115.60...@wi.rr.com
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
On 06/24/2012 07:53 PM, Mitchell Johnson wrote:
I'm testing a few IAX trunk scenarios in a controlled lab. From server2
extension 5000 (Server IP Address 172.16.200.212) I dial 6001 which goes
across the IAX trunk to server 1 (IP address 172.16.200.210). Instead of
ringing the 6001 phone, it plays tt-weasels (the s extension). When I dial
6099 it also plays tt-weasels as it's supposed to, but it's not the
tt-weasels under its extension. It also dials the s extension.
I only placed the s extension in the dial plan to verify that the traffic was
going across the IAX trunk and hitting the correct context.
Any help would be greatly appreciated.
Thanks Mitch
[phones]
exten = _60XX,1,Dial(IAX2/trunk-1)
exten = _X.,1,Dial(IAX2/trunk-1)
exten = 5000,1,Dial(SIP/${EXTEN})
exten = 5000,n,Hangup
same = n,Hangup()
exten = 5099,1,Playback(tt-monkeys)
exten = 5099,n,HangUp
You are not telling asterisk-1 where you want the call to go, so it is going to
's'.
Try adding the extension to the Dial() command on asterisk-2. Change
Dial(IAX2/trunk-1)
to
Dial(IAX2/trunk-1/${EXTEN})
Note: It appears that you are doing it correctly from asterisk-1
towards asterisk-2
exten = _5XXX,1,Dial(${IAXTrunk}/${EXTEN})
Assuming, of course, that the variable IAXTrunk is properly set.
Dale
--
The truth speaks for itself. I'm just the messenger.
Lyta Alexander - Babylon 5
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[asterisk-users] IAX Trunk issue.
I'm testing a few IAX trunk scenarios in a controlled lab. From server2
extension 5000 (Server IP Address 172.16.200.212) I dial 6001 which goes across
the IAX trunk to server 1 (IP address 172.16.200.210). Instead of ringing the
6001 phone, it plays tt-weasels (the s extension). When I dial 6099 it also
plays tt-weasels as it's supposed to, but it's not the tt-weasels under its
extension. It also dials the s extension.
I only placed the s extension in the dial plan to verify that the traffic was
going across the IAX trunk and hitting the correct context.
Any help would be greatly appreciated.
Thanks Mitch
Asterisk-1
IP Address 172.16.200.210
SIP.CONF
[6001]
type=friend
host=dynamic
context=internal_users
secret=xxx
nat=yes
[6002]
type=friend
host=dynamic
context=internal_users
secret=xxx
nat=yes
extensions.conf
[internal_users]
exten = 6000,1,Answer()
exten = 6000,2,Playback(hello-world)
exten = 6000,3,Hangup()
exten = 6001,1,Dial(SIP/6001)
exten = 6002,1,Dial(SIP/6002)
exten = 6099,1,Playback(tt-weasels)
exten = 6099,n,HangUp
exten = _5XXX,1,Dial(${IAXTrunk}/${EXTEN})
same = n,Hangup()
exten = s,1,Answer()
exten = s,n,Playback(tt-weasels)
exten = s,n,Hangup()
IAX.conf
[trunk-1]
type=friend
username=trunk-1
trunk=yes
requiretoken=no
secret=password
host=172.16.200.212
context=internal_users
auth=plaintext
disallow=all
;allow=ulaw
;allow=alaw
allow=gsm
Asterisk-2
IP Address 172.16.200.212
sip.conf
[5000]
type=friend
context=phones
host=dynamic
disallow=all
allow=ulaw
secret=xxx
extensions.conf
[phones]
exten = _60XX,1,Dial(IAX2/trunk-1)
exten = _X.,1,Dial(IAX2/trunk-1)
exten = 5000,1,Dial(SIP/${EXTEN})
exten = 5000,n,Hangup
same = n,Hangup()
exten = 5099,1,Playback(tt-monkeys)
exten = 5099,n,HangUp
iax.conf
[trunk-1]
type=friend
username=trunk-1
trunk=yes
requiretoken=no
secret=password
host=172.16.200.210
context=phones
auth=plaintext
disallow=all
;allow=ulaw
;allow=alaw
allow=gsm
--
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[asterisk-users] Advice on Asterisk Conference
We're looking into using Asterisk to do our conferencing. Currently we do all our conferencing using Cisco, we have a router with PVDM modules so we can offload the hardware resources. I'm looking for some best practices on how to set it up. 1. DO I need a separate server for the conference server? 2. Do I need to offload the actual conference to a router with PVDM modules. 3. Does anyone have experience with transitioning from Cisco conferencing to Asterisk? 4. How many participants can participate in a conference? Thanks, Mitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
