Re: [Asterisk-Users] Route SIP calls to provider
Your SIP provider doesn't need registration? Sounds good. Can you share the
IP address please?
Regards
Cameron
- Original Message -
From: iMRAN [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, April 21, 2005 4:35 AM
Subject: [Asterisk-Users] Route SIP calls to provider
Dear Pros,
Can anyone be kind enough to guide me to route calls to my SIP carrier.
I have configured * to as local PBX from Softphones to hardphones and
vice versa, the hardphone i have is AudioCodec MP108 8 FXS port
gateway.
SIP.conf
[general]
port = 5060
bindaddr = 0.0.0.0
disallow=all
allow=gsm
allow=g729
[1000]
type=friend
username=1000
host=dynamic
context=internal
;canreinvite=yes
dtmfmode=rfc2833
[2000]
type=friend
username=2000
secret=password2
host=dynamic
context=internal
;canreinvite=yes
dtmfmode=rfc2833
extension.conf
[general]
static=yes
writeprotect=yes
[globals]
PHONE1=SIP/1000
PHONE2=SIP/2000
[international]
ignorepat = 88
exten= _1N1NXXNXX,1,Dial ???
[internal]
include = local-sip
[local-sip]
exten = 1000,1,Dial(${PHONE1},40,t)
exten = 1000,2,Hangup
exten = 2000,1,Dial(${PHONE2},40,t)
exten = 2000,2,Hangup
exten = 1001,1,Dial(${PHONE3},40,t)
exten = 1001,2,Hangup
i want user to dial 88 and they will get a tone and dial US or UK
number from local-sip context.
the provider only gave me a IP to route my SIP traffic and needs no
registration, can any please help me how to write the International
context in extension.conf also what i shld do in sip.conf..
my audiocodec supports g729 and g723 codec so do i need to aqquire
license for G729 from digium, if yes then why?
last if possible can you also please tell me wht i need to add on my
context so user can while in calling put the call on hold and transfer
to another sip phone.
I thankyou all for reading this mail and i hope someone will be kind
enough to help.
Best regards,
Imran
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[Asterisk-Users] Route SIP calls to provider
Dear Pros,
Can anyone be kind enough to guide me to route calls to my SIP carrier.
I have configured * to as local PBX from Softphones to hardphones and
vice versa, the hardphone i have is AudioCodec MP108 8 FXS port
gateway.
SIP.conf
[general]
port = 5060
bindaddr = 0.0.0.0
disallow=all
allow=gsm
allow=g729
[1000]
type=friend
username=1000
host=dynamic
context=internal
;canreinvite=yes
dtmfmode=rfc2833
[2000]
type=friend
username=2000
secret=password2
host=dynamic
context=internal
;canreinvite=yes
dtmfmode=rfc2833
extension.conf
[general]
static=yes
writeprotect=yes
[globals]
PHONE1=SIP/1000
PHONE2=SIP/2000
[international]
ignorepat = 88
exten= _1N1NXXNXX,1,Dial ???
[internal]
include = local-sip
[local-sip]
exten = 1000,1,Dial(${PHONE1},40,t)
exten = 1000,2,Hangup
exten = 2000,1,Dial(${PHONE2},40,t)
exten = 2000,2,Hangup
exten = 1001,1,Dial(${PHONE3},40,t)
exten = 1001,2,Hangup
i want user to dial 88 and they will get a tone and dial US or UK
number from local-sip context.
the provider only gave me a IP to route my SIP traffic and needs no
registration, can any please help me how to write the International
context in extension.conf also what i shld do in sip.conf..
my audiocodec supports g729 and g723 codec so do i need to aqquire
license for G729 from digium, if yes then why?
last if possible can you also please tell me wht i need to add on my
context so user can while in calling put the call on hold and transfer
to another sip phone.
I thankyou all for reading this mail and i hope someone will be kind
enough to help.
Best regards,
Imran
___
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RE: [Asterisk-Users] Route SIP calls to provider
From your descriptions of your needs, you would be better served with an
AAH installation. Easier to understand than hand coding your contexts.
That aside, here are few answers...
Look here for more...
www.voip-info.org
Routing to the VoIP is just a matter of dial plan matching (see dial
plan at voip-info)
Codecs on Asterisk require the license. Your phone may support the
codec but your server needs a license to do so.
Hold and transfer are usually part of the phone itself. For example, my
Polycom holds the line and lets me transfer from itself.
Otherwise, I can transfer a call via * by dialing # and the extension
target.
This is a blind transfer so the target extension just gets the call
without you even talking to them.
Consultative transfer is different, details located at www.voip-info.org
Cheers,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of iMRAN
Sent: Wednesday, April 20, 2005 9:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Route SIP calls to provider
Dear Pros,
Can anyone be kind enough to guide me to route calls to my SIP carrier.
I have configured * to as local PBX from Softphones to hardphones and
vice versa, the hardphone i have is AudioCodec MP108 8 FXS port gateway.
SIP.conf
[general]
port = 5060
bindaddr = 0.0.0.0
disallow=all
allow=gsm
allow=g729
[1000]
type=friend
username=1000
host=dynamic
context=internal
;canreinvite=yes
dtmfmode=rfc2833
[2000]
type=friend
username=2000
secret=password2
host=dynamic
context=internal
;canreinvite=yes
dtmfmode=rfc2833
extension.conf
[general]
static=yes
writeprotect=yes
[globals]
PHONE1=SIP/1000
PHONE2=SIP/2000
[international]
ignorepat = 88
exten= _1N1NXXNXX,1,Dial ???
[internal]
include = local-sip
[local-sip]
exten = 1000,1,Dial(${PHONE1},40,t)
exten = 1000,2,Hangup
exten = 2000,1,Dial(${PHONE2},40,t)
exten = 2000,2,Hangup
exten = 1001,1,Dial(${PHONE3},40,t)
exten = 1001,2,Hangup
i want user to dial 88 and they will get a tone and dial US or UK number
from local-sip context.
the provider only gave me a IP to route my SIP traffic and needs no
registration, can any please help me how to write the International
context in extension.conf also what i shld do in sip.conf..
my audiocodec supports g729 and g723 codec so do i need to aqquire
license for G729 from digium, if yes then why?
last if possible can you also please tell me wht i need to add on my
context so user can while in calling put the call on hold and transfer
to another sip phone.
I thankyou all for reading this mail and i hope someone will be kind
enough to help.
Best regards,
Imran
___
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Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [Asterisk-Users] Route SIP calls to provider
Hi,
I hope this will help. I will give you my configuration. I'm using nikotel
to make international calls thru
My SIP provider.
Sip.conf
[general]
port=5060
bindaddr=10.0.0.10
disallow=all
allow=g726
allow=alaw
reinvite=no
register = myid:[EMAIL PROTECTED]/myid
[2000]
type=friend
username=2000
secret=test
host=dynamic
context=from-sip
mailbox=100
disallow=all
allow=g729 ; Using g729, to get better sound quality on my
Zyxel P2000W
dtmfmode=rfc2833
[2001]
type=friend
username=2001
secret=test
host=dynamic
context=from-sip
mailbox=101
disallow=all
allow=g729; Using g729, to get better sound quality on my
Zyxel P2000W
dtmfmode=rfc2833
[2002]
type=friend
username=2002
secret=test
host=dynamic
context=from-sip
mailbox=102
dtmfmode=rfc2833 ; I'm not using g729 here, because I have a
excelent voice quality with the alaw or g726 codec
; This is a wired IP phone
[nikotel]
type=friend
context=sip-dial
username=myid
secret=mypasswd
host=calamar0.nikotel.com
fromuser=myid
fromdomain=nikotel.com
canreinvite=no
qualify=yes
disallow=all
allow=gsm
insecure=very
nat=yes
dtmfmode=info
tos=0x18
extentions.conf :
[general]
static=yes
writeprotect=yes
;Local SIP phones
[from-sip]
exten = 2000,1,Dial(SIP/2000,20)
exten = 2000,2,Voicemail(u2000)
exten = 2000,102,Voicemail(b2000)
exten = 2000,103,Hangup
exten = 2001,1,Dial(SIP/2001,20)
exten = 2001,2,Voicemail(u2001)
exten = 2001,102,Voicemail(b2001)
exten = 2001,103,Hangup
exten = 2002,1,Dial(SIP/2002,20)
exten = 2002,2,Voicemail(u2002)
exten = 2002,102,Voicemail(b2002)
exten = 2002,103,Hangup
exten = 2003,1,Dial(SIP/2003,20)
exten = 2003,2,Voicemail(u2003)
exten = 2003,102,Voicemail(b2003)
exten = 2003,103,Hangup
;Call to Hell
exten = 666,1,Answer
exten = 666,2,MP3Player(/home/asterisk/satani.mp3)
exten = 666,202,Hangup
;Mailbox
exten = ,1,VoiceMailMain2(s${CALLERIDNUM})
exten = ,2,Hangup
;Request your phonenumber
exten = 5595,1,Answer
exten = 5595,2,SayNumber(${CALLERIDNUM})
exten = 5595,3,Hangup
;Outgoing calls thru SIP Provider
exten = _00.,1,SetCallerID(004935179799272)
exten = _00.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _00.,3,Hangup
;Incoming calls thru SIP Provider
[sip-dial]
exten = myid,1,Dial(SIP/2000SIP/2001SIP/2002)
exten = myid,102,Hangup
1.This works for incoming and outgoing calls thru your SIP provider. It was
hard working to get it fixed, but
Now I have no probs anymore.
2.Why g729 : I bought a few licenses, to use with my Zyxel phones (better
voice quality). It's not required,
but it's usefull to use with your sip phones, almost not required in your
SIP provider part.
3.Oh yeah, to make outgoing calls thru your SIP provider you just have to
dial 00 + phonenumber, ie.
To make a call to germany : 004912321212. (but you can change this as
required)
Cheers,
Alexander Scheerschmidt, TnmT Org.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of iMRAN
Sent: Wednesday, 20 April 2005 18:36
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Route SIP calls to provider
Dear Pros,
Can anyone be kind enough to guide me to route calls to my SIP carrier.
I have configured * to as local PBX from Softphones to hardphones and vice
versa, the hardphone i have is AudioCodec MP108 8 FXS port gateway.
SIP.conf
[general]
port = 5060
bindaddr = 0.0.0.0
disallow=all
allow=gsm
allow=g729
[1000]
type=friend
username=1000
host=dynamic
context=internal
;canreinvite=yes
dtmfmode=rfc2833
[2000]
type=friend
username=2000
secret=password2
host=dynamic
context=internal
;canreinvite=yes
dtmfmode=rfc2833
extension.conf
[general]
static=yes
writeprotect=yes
[globals]
PHONE1=SIP/1000
PHONE2=SIP/2000
[international]
ignorepat = 88
exten= _1N1NXXNXX,1,Dial ???
[internal]
include = local-sip
[local-sip]
exten = 1000,1,Dial(${PHONE1},40,t)
exten = 1000,2,Hangup
exten = 2000,1,Dial(${PHONE2},40,t)
exten = 2000,2,Hangup
exten = 1001,1,Dial(${PHONE3},40,t)
exten = 1001,2,Hangup
i want user to dial 88 and they will get a tone and dial US or UK number
from local-sip context.
the provider only gave me a IP to route my SIP traffic and needs no
registration, can any please help me how to write the International context
in extension.conf also what i shld do in sip.conf..
my audiocodec supports g729 and g723 codec so do i need to aqquire license
for G729 from digium, if yes then why?
last if possible can you also please tell me wht i need to add on my context
so user can while in calling put the call on hold and transfer to another
sip phone.
I thankyou all for reading this mail and i hope someone will be kind enough
to help.
Best regards,
Imran
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
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