Hi list!
I'm trying to connect Analog GSM gateway (2N Ateus) with Asterisk 1.2.5 over
Grandstream HT488 ATA. I have plugged gateway to FXO port of ATA. Incoming calls
work the way I call GSM number and then I get DISA to call inside company.
Outgoing call work well when I call VoIP number of ATA which calls gateway and
then I dial number I wish to call over gateway. As I said, it works fine.
Now I would like to dial ATA_number+number_I_wish_to_call so that I don't
have to dial twice when I'm trying to establish outgoing call from company
thru gateway.
I have tried this but it doesn't work well.
; to dial outside thru GSM gateway
exten => _456.,1,Dial(SIP/${EXTEN},30,tTD(${EXTEN:3}))
exten => _456.,n,Hangup
This is what I see on CLI:
-- Executing Dial("SIP/577-104c",
"SIP/4560989970434|30|tTD(248)") in new stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup("SIP/577-104c", "") in new stack
== Spawn extension (sip, 4560989970434, 2) exited non-zero on
'SIP/577-104c'
Why asterisk thinks that gateway is busy when it's not?
--
Tomislav Par?ina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: tomo@pbx.lama.hr
e-mail: tparcina#lama.hr
http://www.lama.hr
What you need is something like:
exten => _456.,1,Dial(SIP/${EXTEN}@IP.OF.2N.UNIT,30,tTD(${EXTEN:3}))
regards,
PaulH
AsteriskIT
www.asteriskit.com.au
On Tue, 2006-08-22 at 10:59 +0200, Tomislav Par?ina
wrote:> Hi list!
>
> I'm trying to connect Analog GSM gateway (2N Ateus) with Asterisk 1.2.5
over Grandstream HT488 ATA. I have plugged gateway to FXO port of ATA. Incoming
calls work the way I call GSM number and then I get DISA to call inside company.
Outgoing call work well when I call VoIP number of ATA which calls gateway and
then I dial number I wish to call over gateway. As I said, it works fine.
>
> Now I would like to dial ATA_number+number_I_wish_to_call so that I
don't have to dial twice when I'm trying to establish outgoing call from
company thru gateway.
>
> I have tried this but it doesn't work well.
>
> ; to dial outside thru GSM gateway
> exten => _456.,1,Dial(SIP/${EXTEN},30,tTD(${EXTEN:3}))
> exten => _456.,n,Hangup
>
> This is what I see on CLI:
>
> -- Executing Dial("SIP/577-104c",
"SIP/4560989970434|30|tTD(248)") in new stack
> == Everyone is busy/congested at this time (1:0/0/1)
> -- Executing Hangup("SIP/577-104c", "") in new
stack
> == Spawn extension (sip, 4560989970434, 2) exited non-zero on
'SIP/577-104c'
>
> Why asterisk thinks that gateway is busy when it's not?
>
>
>
> --
> Tomislav Par?ina
> Lama Computers Split
> Stinice 12, 21000 Split
> Tel.: +385(21)495148
> Mob.: +385(91)1212148
> SIP: tomo@pbx.lama.hr
> e-mail: tparcina#lama.hr
> http://www.lama.hr
> _______________________________________________
> --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
>
Assuming the 456 is the ATA number and the outside number is always 10
digits.
exten => _456.,1,Dial(SIP/${EXTEN:0:3}/${EXTEN:-10},tT)
but then it might as well be
exten => _456.,1,Dial(SIP/456/${EXTEN:-10},tT)
> ; to dial outside thru GSM gateway
> exten => _456.,1,Dial(SIP/${EXTEN},30,tTD(${EXTEN:3}))
> exten => _456.,n,Hangup
>
M
In article <44EACA39.3000309@mjvn.net>, m@mjvn.net says...> Assuming the 456 is the ATA number and the outside number is always 10 > digits. > > exten => _456.,1,Dial(SIP/${EXTEN:0:3}/${EXTEN:-10},tT) > > but then it might as well be > > exten => _456.,1,Dial(SIP/456/${EXTEN:-10},tT)Paul, Mamus, thank you both! Combining you answers I have found solution. I have forget to exclude other numbers (those after first three). Now this is what works just fine. exten => _456.,1,Dial(SIP/${EXTEN:0:3},30,tTD(${EXTEN:3})) Another question, if I would like to delay sending EXTEN:3 digits, where should I put w option? If I put it after tTD then it's for one touch record. -- Tomislav Par?ina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: tomo@pbx.lama.hr e-mail: tparcina#lama.hr http://www.lama.hr
Inside the brackets for the D flag> exten => _456.,1,Dial(SIP/${EXTEN:0:3},30,tTD(*w*${EXTEN:3})) >M -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060822/cb0fb085/attachment.htm
In article <44EAE487.3010900@mjvn.net>, m@mjvn.net says...> Inside the brackets for the D flag > > exten => _456.,1,Dial(SIP/${EXTEN:0:3},30,tTD(*w*${EXTEN:3}))Thank you Marnus. -- Tomislav Par?ina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: tomo@pbx.lama.hr e-mail: tparcina#lama.hr http://www.lama.hr
Hello
WE can provide you with budget GSM Gateway if you are interested?
Sam
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tomislav
Par?ina
Sent: Tuesday, August 22, 2006 4:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] GSM gateway and FXO ATA
Hi list!
I'm trying to connect Analog GSM gateway (2N Ateus) with Asterisk 1.2.5 over
Grandstream HT488 ATA. I have plugged gateway to FXO port of ATA. Incoming
calls work the way I call GSM number and then I get DISA to call inside
company. Outgoing call work well when I call VoIP number of ATA which calls
gateway and then I dial number I wish to call over gateway. As I said, it
works fine.
Now I would like to dial ATA_number+number_I_wish_to_call so that I don't
have to dial twice when I'm trying to establish outgoing call from company
thru gateway.
I have tried this but it doesn't work well.
; to dial outside thru GSM gateway
exten => _456.,1,Dial(SIP/${EXTEN},30,tTD(${EXTEN:3}))
exten => _456.,n,Hangup
This is what I see on CLI:
-- Executing Dial("SIP/577-104c",
"SIP/4560989970434|30|tTD(248)") in
new stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup("SIP/577-104c", "") in new stack
== Spawn extension (sip, 4560989970434, 2) exited non-zero on
'SIP/577-104c'
Why asterisk thinks that gateway is busy when it's not?
--
Tomislav Par?ina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: tomo@pbx.lama.hr
e-mail: tparcina#lama.hr
http://www.lama.hr
_______________________________________________
--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
On Sat, Aug 26, 2006 at 01:35:36AM +0800, Sam Tam wrote:> WE can provide you with budget GSM Gateway if you are interested?which is commercial nope? wrong list again????? could have been private Email? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN steve@gbnet.net Euro Tech News Blog http://eurotechnews.blogspot.com
On 2006-08-22 01:59:09 -0700, Tomislav Par?ina <tparcina@lama.hr> said:> Hi list! > > I'm trying to connect Analog GSM gateway (2N Ateus) with Asterisk 1.2.5 > over Grandstream HT488 ATA.<snip> Personally I found the FXO port on the HT-488 to unworkable except as a backup for power outages. I found several problems with it. 1) serious echo issues (I have a long loop). 2) If the phone is answered on the first ring the call goes off to la la land. Explaining to users (or myself) that you need to wait for the second audible ring on the handset's before answering isn't acceptable. 3) The device hangs and reboots itself occasionally. This is all just an FYI. Marty PS I did test with the latest HT-488 firmware and all issues were still present.
In article <ecqctl$o8s$2@sea.gmane.org>, ast@stillnewt.org says...> 2) If the phone is answered on the first ring the call goes off to la > la land. Explaining to users (or myself) that you need to wait for the > second audible ring on the handset's before answering isn't acceptable.Hi Marty! Can you tell me more about this? You mean when call from SIP goes to FXO port, if phone attached on FXO port answers after the first ring (before second) ATA will always stop to work? -- Tomislav Par?ina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: tomo@pbx.lama.hr e-mail: tparcina#lama.hr http://www.lama.hr
In article <ed793h$k2m$2@sea.gmane.org>, ast@stillnewt.org says...> Actually it's kind of the opposite... When a call comes in to the FXO, > and it rings the FXS, if the FXS answers on the first ring, the call > goes somewhere but who knows where. > > The picking up party hears a dial tone, and the caller hears dead air.Hi Marty, I can live with that. I don't have anything connected to FXS port :) -- Tomislav Par?ina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: tomo@pbx.lama.hr e-mail: tparcina#lama.hr http://www.lama.hr