Re: [asterisk-users] Outside lines are *STILL* just not happening...
Hi David - Sometimes (about half the time) the phone I'm calling (in this case, a cell) will give part of one ring, then report a missed call. The SIP phone hangs up after about 5 seconds. But not always. The rest of the time, the SIP phone just eventually (15 or 20 secs) hangs up on its' own, and the cell never reports a missed call. Have you tried using a regular analog phone on the PSTN lines (without using asterisk at all)? If you're getting partial connections that are failing, it could be a line quality issue. For testing, you might also try turning off as many options as possible in zapata.conf (particularly the echo cancel). - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outside lines are *STILL* just not happening...
In your zaptel.conf you need to use fxsks rather than fxoks
hth
Harvey
- Original Message -
From: J. David Bavousett [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 16, 2007 3:59 PM
Subject: [asterisk-users] Outside lines are *STILL* just not happening...
From yesterday:
-Original Message-
Here's the configuration...my Asterisk box has a TDM844B in it; port 1-4
are FXS, 5-8 FXOs.
Here are the config files:
/etc/zaptel.conf:
fxoks=1
fxoks=2
fxoks=3
fxoks=4
fxsks=5
fxsks=6
fxsks=7
fxsks=8
loadzone= us
defaultzone = us
/etc/asterisk/zapata.conf:
[channels]
language=en
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=no
transfer=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
canpark=yes
rxgain=0.0
txgain=0.0
context=internal
signalling=fxo_ks
channel = 1-4
context=external
signalling=fxs_ks
channel = 5-8
A snippet from /etc/asterisk/extensions.conf:
[internal]
ignorepat = 9
exten = _9NXX,1,Dial(Zap/5/${EXTEN:1})
exten = _9NXX,2,Congestion()
exten = _9NXX,102,Congestion()
The SIP phone is also in the internal context, and other things below
that in the context work just fine on the internal network.
Problem B: Dialing out. From the SIP phone, if I dial out, here's the
transcript:
-- Executing Dial(SIP/102-081854e0, Zap/5/6653674) in new stack
-- Called 5/6653674
-- Zap/5-1 answered SIP/102-081854e0
-- Hungup 'Zap/5-1'
== Spawn extension (internal, 96653674, 1) exited non-zero on
'SIP/102-081854e0'
Sometimes (about half the time) the phone I'm calling (in this case, a
cell) will give part of one ring, then report a missed call. The SIP
phone hangs up after about 5 seconds. But not always. The rest of the
time, the SIP phone just eventually (15 or 20 secs) hangs up on its'
own, and the cell never reports a missed call.
I know this has been long, and wordy...hope someone can help. We're
newbs around here, and trying to get things working. My boss is *very*
impressed with the menus and such I've got set up and working, and we've
used soft phones via VPN and it works great...now we just need our
outside lines working!
Thanks a million!
J. David Bavousett
System Administrator
Abilene Library Consortium
One other symptom: I never, ever hear any ringing on the SIP phone
while it's dialing out. No indication is ever given that it's ringing
on the other end.
Can *anyone* shed some light on this?
--David
___
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Re: [asterisk-users] Outside lines are *STILL* just not happening...
in zaptel.conf use fxsks
- Original Message -
From: J. David Bavousett [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 16, 2007 3:59 PM
Subject: [asterisk-users] Outside lines are *STILL* just not happening...
From yesterday:
-Original Message-
Here's the configuration...my Asterisk box has a TDM844B in it; port 1-4
are FXS, 5-8 FXOs.
Here are the config files:
/etc/zaptel.conf:
fxoks=1
fxoks=2
fxoks=3
fxoks=4
fxsks=5
fxsks=6
fxsks=7
fxsks=8
loadzone= us
defaultzone = us
/etc/asterisk/zapata.conf:
[channels]
language=en
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=no
transfer=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
canpark=yes
rxgain=0.0
txgain=0.0
context=internal
signalling=fxo_ks
channel = 1-4
context=external
signalling=fxs_ks
channel = 5-8
A snippet from /etc/asterisk/extensions.conf:
[internal]
ignorepat = 9
exten = _9NXX,1,Dial(Zap/5/${EXTEN:1})
exten = _9NXX,2,Congestion()
exten = _9NXX,102,Congestion()
The SIP phone is also in the internal context, and other things below
that in the context work just fine on the internal network.
Problem B: Dialing out. From the SIP phone, if I dial out, here's the
transcript:
-- Executing Dial(SIP/102-081854e0, Zap/5/6653674) in new stack
-- Called 5/6653674
-- Zap/5-1 answered SIP/102-081854e0
-- Hungup 'Zap/5-1'
== Spawn extension (internal, 96653674, 1) exited non-zero on
'SIP/102-081854e0'
Sometimes (about half the time) the phone I'm calling (in this case, a
cell) will give part of one ring, then report a missed call. The SIP
phone hangs up after about 5 seconds. But not always. The rest of the
time, the SIP phone just eventually (15 or 20 secs) hangs up on its'
own, and the cell never reports a missed call.
I know this has been long, and wordy...hope someone can help. We're
newbs around here, and trying to get things working. My boss is *very*
impressed with the menus and such I've got set up and working, and we've
used soft phones via VPN and it works great...now we just need our
outside lines working!
Thanks a million!
J. David Bavousett
System Administrator
Abilene Library Consortium
One other symptom: I never, ever hear any ringing on the SIP phone
while it's dialing out. No indication is ever given that it's ringing
on the other end.
Can *anyone* shed some light on this?
--David
___
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RE: [asterisk-users] Outside lines are *STILL* just not happening...
Really, Harvey?
Lines 1-4 are FXS ports, lines 5-8 are FXO... My outside line is
plugged into port 8. Ports 1-4 are inside, and work fine.
--David
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wireless
Sent: Wednesday, May 16, 2007 10:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Outside lines are *STILL* just not
happening...
In your zaptel.conf you need to use fxsks rather than fxoks
hth
Harvey
- Original Message -
From: J. David Bavousett [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 16, 2007 3:59 PM
Subject: [asterisk-users] Outside lines are *STILL* just not
happening...
From yesterday:
-Original Message-
Here's the configuration...my Asterisk box has a TDM844B in it; port 1-4
are FXS, 5-8 FXOs.
Here are the config files:
/etc/zaptel.conf:
fxoks=1
fxoks=2
fxoks=3
fxoks=4
fxsks=5
fxsks=6
fxsks=7
fxsks=8
loadzone= us
defaultzone = us
/etc/asterisk/zapata.conf:
[channels]
language=en
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=no
transfer=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
canpark=yes
rxgain=0.0
txgain=0.0
context=internal
signalling=fxo_ks
channel = 1-4
context=external
signalling=fxs_ks
channel = 5-8
A snippet from /etc/asterisk/extensions.conf:
[internal]
ignorepat = 9
exten = _9NXX,1,Dial(Zap/5/${EXTEN:1})
exten = _9NXX,2,Congestion()
exten = _9NXX,102,Congestion()
The SIP phone is also in the internal context, and other things below
that in the context work just fine on the internal network.
Problem B: Dialing out. From the SIP phone, if I dial out, here's the
transcript:
-- Executing Dial(SIP/102-081854e0, Zap/5/6653674) in new stack
-- Called 5/6653674
-- Zap/5-1 answered SIP/102-081854e0
-- Hungup 'Zap/5-1'
== Spawn extension (internal, 96653674, 1) exited non-zero on
'SIP/102-081854e0'
Sometimes (about half the time) the phone I'm calling (in this case, a
cell) will give part of one ring, then report a missed call. The SIP
phone hangs up after about 5 seconds. But not always. The rest of the
time, the SIP phone just eventually (15 or 20 secs) hangs up on its'
own, and the cell never reports a missed call.
I know this has been long, and wordy...hope someone can help. We're
newbs around here, and trying to get things working. My boss is *very*
impressed with the menus and such I've got set up and working, and we've
used soft phones via VPN and it works great...now we just need our
outside lines working!
Thanks a million!
J. David Bavousett
System Administrator
Abilene Library Consortium
One other symptom: I never, ever hear any ringing on the SIP phone
while it's dialing out. No indication is ever given that it's ringing
on the other end.
Can *anyone* shed some light on this?
--David
___
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Re: [asterisk-users] Outside lines are *STILL* just not happening...
sorry should have read your complete post!
- Original Message -
From: J. David Bavousett [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 16, 2007 6:58 PM
Subject: RE: [asterisk-users] Outside lines are *STILL* just not
happening...
Really, Harvey?
Lines 1-4 are FXS ports, lines 5-8 are FXO... My outside line is
plugged into port 8. Ports 1-4 are inside, and work fine.
--David
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wireless
Sent: Wednesday, May 16, 2007 10:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Outside lines are *STILL* just not
happening...
In your zaptel.conf you need to use fxsks rather than fxoks
hth
Harvey
- Original Message -
From: J. David Bavousett [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 16, 2007 3:59 PM
Subject: [asterisk-users] Outside lines are *STILL* just not
happening...
From yesterday:
-Original Message-
Here's the configuration...my Asterisk box has a TDM844B in it; port 1-4
are FXS, 5-8 FXOs.
Here are the config files:
/etc/zaptel.conf:
fxoks=1
fxoks=2
fxoks=3
fxoks=4
fxsks=5
fxsks=6
fxsks=7
fxsks=8
loadzone= us
defaultzone = us
/etc/asterisk/zapata.conf:
[channels]
language=en
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=no
transfer=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
canpark=yes
rxgain=0.0
txgain=0.0
context=internal
signalling=fxo_ks
channel = 1-4
context=external
signalling=fxs_ks
channel = 5-8
A snippet from /etc/asterisk/extensions.conf:
[internal]
ignorepat = 9
exten = _9NXX,1,Dial(Zap/5/${EXTEN:1})
exten = _9NXX,2,Congestion()
exten = _9NXX,102,Congestion()
The SIP phone is also in the internal context, and other things below
that in the context work just fine on the internal network.
Problem B: Dialing out. From the SIP phone, if I dial out, here's the
transcript:
-- Executing Dial(SIP/102-081854e0, Zap/5/6653674) in new stack
-- Called 5/6653674
-- Zap/5-1 answered SIP/102-081854e0
-- Hungup 'Zap/5-1'
== Spawn extension (internal, 96653674, 1) exited non-zero on
'SIP/102-081854e0'
Sometimes (about half the time) the phone I'm calling (in this case, a
cell) will give part of one ring, then report a missed call. The SIP
phone hangs up after about 5 seconds. But not always. The rest of the
time, the SIP phone just eventually (15 or 20 secs) hangs up on its'
own, and the cell never reports a missed call.
I know this has been long, and wordy...hope someone can help. We're
newbs around here, and trying to get things working. My boss is *very*
impressed with the menus and such I've got set up and working, and we've
used soft phones via VPN and it works great...now we just need our
outside lines working!
Thanks a million!
J. David Bavousett
System Administrator
Abilene Library Consortium
One other symptom: I never, ever hear any ringing on the SIP phone
while it's dialing out. No indication is ever given that it's ringing
on the other end.
Can *anyone* shed some light on this?
--David
___
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asterisk-users mailing list
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http://lists.digium.com/mailman/listinfo/asterisk-users
--
This message has been scanned for viruses and
dangerous content by ESVA, and is
believed to be clean.
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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