RE: [Asterisk-Users] Pots Extensions
Did you put immediate=yes in your zapata.conf? I had similar problems
previously (I have T100p instead of X100p) and it is fixed when I put
immediate=no.
Lisa
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David J
Carter
Sent: Tuesday, May 04, 2004 12:43 PM
To: Asterisk User Group
Subject: [Asterisk-Users] Pots Extensions
Hi all,
I am either going daft or not reading things right.
I have just installed, (well 10 hours ago) a TDM40B and 3 X100P cards. I
have followed the examples for the conf files to the letter.
I can call the pots extensions OK from IAX clients, SIP clients and from
the
incoming X100P cards.
But, if I pick up the handset to make a call all I get is the engaged
tone
and the following message.
May 4 23:24:24 WARNING[221200]: pbx.c:1812 ast_pbx_run: Channel
'ZAP/5-1'
sent into invalid extension 's' in context 'default' but no invalid
handler.
If I am reading my configs then shouldn't they be going to the internal
context?
Do I need to set-up pots extensions somewhere like IAX Sip extensions?
=
zaptel.conf
fxsks=1-3
fxoks=4-7
loadzone=uk
zapata.conf
signalling=fxs_ks
context=incoming
channel = 1-3
signalling=fxo_ks
context=internal
channel = 4-7
extensions.conf
[internal]
exten = 4090,1,Dial,ZAP/4
exten = 4091,1,Dial,ZAP/5
exten = 4092,1,Dial,ZAP/6
exten = 4093,1,Dial,ZAP/7
exten = _9X.,Dial,ZAP/1,${EXTEN:1}
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[Asterisk-Users] RE: FXS card dial digit wrong
Well, I just figured out that 4 digit dialing plan is used on the other
end so if I just 4 digit extension, i.e. 95222, the call works out fine
for both the sip phone and the pots phone.
However, I still don't understand that when I dial full 7 digit number,
sip phones work but pots does not. I'll ignore it for now:)
-Original Message-
From: Lisa Xie
Sent: Friday, April 30, 2004 9:52 AM
To: '[EMAIL PROTECTED]'
Subject: FXS card dial digit wrong
Hello, everyone,
I am currently trying to get the asterisk server to talk with a 3COM NBX
with T1 connection. My asterisk server has a T100p, TDM20B, a couple of
sip phones. Now the sip phones are calling 3COM NBX phones fine,
however, the analog phone has problem when dialing the NBX phones. The
connection is established and the NBX auto-attendant picks up the call
however the NBX end says that incorrect extension number is dialed,
From 3com NBX end to Asterisk is fine, i.e., 3com NBX phones call both
the sip phone and the analog phone with no problem.
Below is the console output from Asterisk when I tried to call the same
extension using both the sip phone and the analog phone.
Also my configuration files are attached: zapata.conf, zaptel.conf,
extensions.conf.
Thanks for your help.
Lisa
~~~Console output from Asterisk~~~
*CLI
-- Executing Dial(SIP/2001-2445, Zap/g1/222) in new stack
-- Called g1/222
-- Zap/1-1 answered SIP/2001-2445
-- Hungup 'Zap/1-1'
== Spawn extension (internal, 9222, 1) exited non-zero on
'SIP/2001-2445'
-- Starting simple switch on 'Zap/26-1'
-- Executing Dial(Zap/26-1, Zap/g1/222) in new stack
-- Called g1/222
-- Zap/1-1 answered Zap/26-1
-- Attempting native bridge of Zap/26-1 and Zap/1-1
-- Hungup 'Zap/1-1'
== Spawn extension (internal, 9222, 1) exited non-zero on
'Zap/26-1'
-- Hungup 'Zap/26-1'
~~~My configuration files are here~~~
---Zaptel.conf---
#add t100 card
span=1,0,0,esf,b8zs
em=1-24
loadzone = us
defaultzone=us
#add tdm20b card
fxoks=25-26
---Zapata.conf
;add for t100 card
signalling=em_w
context=incoming
group=1
immediate=yes
channel = 1-24
;add for tdm20b card
signalling=fxo_ks
context=internal
channel=25-26
---Extensions.conf---
[incoming]
exten = _XXX2001.,1,Dial(SIP/2001,20)
exten = _XXX2101.,1,Dial,Zap/26
include = internal
[internal]
exten = s,1,Playback(demo-congrats)
exten = 2001,1,Dial(SIP/2001,20)
exten = 2100,1,Dial,Zap/25
exten = 2101,1,Dial,Zap/26
;outbound calls
exten =_9.,1,Dial(Zap/g1/${EXTEN:1})
-Original Message-
From: Lisa Xie
Sent: Wednesday, April 28, 2004 5:51 PM
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] Extra digit needed for outbound call
Here is part of the files: extensions.conf for both of the servers
Asterisk server 1
[incoming]
include = internal
[internal]
exten = s,1,Playback(demo-congrats)
exten = 2000,1,Dial(SIP/2000,20)
exten = 2100,1,Dial,Zap/25
;outbound calls
ignorepat = 9
exten =_9.,1,Dial(Zap/g1/${EXTEN:1})
Asterisk server 2
[incoming]
include = internal
[internal]
exten = s,1,Playback(demo-congrats)
exten = 2000,1,Dial(SIP/2000,20)
;outbound calls
ignorepat = 9
exten =_9.,1,Dial(Zap/g1/${EXTEN:1})
Thanks!
Lisa
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Wednesday, April 28, 2004 4:39 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Extra digit needed for outbound call
On Wed, 2004-04-28 at 14:49, Lisa Xie wrote:
Hi,
I've been working on starting a lab of end to end asterisk system and
now most of pieces seem to be working. The two asterisk servers are
connected by T1. Both servers have a couple of SIP phones connected
and
one of the servers has a FXS card with an analog phone hanging.
I can make calls across the T1 link however there is one thing that I
don't understand. I need to append one extra digit to get the correct
extension number at the other end. For example, when I tried to call
extension 2000 at the other end, I need to dial 92000x, where x can
be
anything between 0-9. Otherwise, if I dial 92000, the console says
something like extension 200 is not found.
Also internal calls are normal.
This looks very bizarre for me... How can I fix it?
Examine your dialplan. Post it here and maybe someone will point out
what you are doing wrong.
--
Steven Critchfield [EMAIL PROTECTED]
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[Asterisk-Users] 3com SIP phone working with asterisk
Hello everyone, I just like to let you know that I tested Asterisk with 3COM SIP phones and it worked fine. The 3Com phones are old ones with the same look of NBX 2102 phone but different product number: P/N: 655005001 Rev B There is no special set up except that I have to specifically put allow=ulaw in sip.conf. Otherwise, there is codec unrecognized error. [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) allow=ulaw; Allow all codecs Lisa ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] avaya and linux
I heard it once that the Avaya's Definity runs linux but I am not familiar with the product so sorry if it was wrong. Lisa -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Glen Ford Sent: Friday, April 02, 2004 2:48 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] avaya and linux Does anyone know if avaya voip product is running linux under the hood? Thanks, /glen -- Glen Ford [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] avaya and linux
FYI. http://www.nwfusion.com/news/2003/1208avaya.html New products on tap from Avaya include: * The S8500 Media Server, a Linux-based call processor that supports up to 3,200 phones. Lisa -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lisa Xie Sent: Friday, April 02, 2004 2:56 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] avaya and linux I heard it once that the Avaya's Definity runs linux but I am not familiar with the product so sorry if it was wrong. Lisa -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Glen Ford Sent: Friday, April 02, 2004 2:48 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] avaya and linux Does anyone know if avaya voip product is running linux under the hood? Thanks, /glen -- Glen Ford [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Any Suggestion for this system?
Dear everyone, I am new to the mailing list and the asterisk system. I am looking for suggestions to start a VoIP test lab and I am seriously interested in the asterisk solution. After some homework, this is the end to end system that I come up in mind: 1. Two PCs running asterisk: Intel Celeron, 833MHz, 256M memory 2. Each PC with one Wildcard T100P Single-Span T1, one Wildcard TDM400P Quad-Port FXS PCI 3. 5 3com SIP phones plus software phones 4. And I like to interconnect these two PC through Ascend 4000 WAN router. Will the system work? The test lab will probably be expanded later and need: 1. WAN simulator 2. Asterisk system to other vendor system: Nortel, Cisco. 3. E1 support Do you see any potential incompatibility in this initial system for future needs? Thanks! Lisa ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
