Ronald Wiplinger
2004-Oct-23 23:08 UTC
[Asterisk-Users] Asterisk, ATA-186 & Sipgate.de / sipgate.co.uk
I try to get the following to work:
Sipgate.de and sipgate.co.uk are configured as gateway, while the
ATA-186 has two phone sets attached.
I tried:
ATA settings as described at:
http://www.loligo.com/asterisk/Cisco/ATA-186-guide.v20030628.txt
(just with a fixed IP)
sip.conf:
=====
[general]
context=default
port=5060
bindaddr=0.0.0.0
srvlookup=yes
register => 5552220:my_password1@sipgate.de/5552220
register => 4782156:my_password2@sipgate.co.uk/4782156
externip = 61.220.121.xx
localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
[601]
type=friend
username=601
secret=my_password1
canreinvite=no
host=dynamic
dtmfmode=rfc2833
mailbox=601
nat=yes
[602]
type=friend
username=602
secret=my_password2
canreinvite=no
host=dynamic
dtmfmode=rfc2833
mailbox=601
nat=yes
[sipgate.de]
type=friend
username=5552220
secret=my_password3
host=sipgate.de
fromuser=5552220
fromdomain=sipgate.net
nat=yes
context=incomingsipgate
canreinvite=no
[sipgate.co.uk]
type=friend
username=4782156
secret=my_password4
host=sipgate.co.uk
fromuser=4782156
fromdomain=sipgate.net
nat=yes
context=incomingsipgate
canreinvite=no
extensions.conf:
==========
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
[incomingsipgate]
exten => h,1,Hangup
exten => 800XXXX,1,Dial(SIP/internestelefon,20,tr)
[sipgate.de]
exten => _0049.,Dial(SIP/${EXTEN:1}@sipgate.de,30,tr)
exten => _0049.,2,Playback(invalid)
exten => _0049.,3,Hangup
[sipgate.co.uk]
exten => _0044.,Dial(SIP/${EXTEN:1}@sipgate.co.uk,30,tr)
exten => _0044.,2,Playback(invalid)
exten => _0044.,3,Hangup
I did not understand the paragraph of [incomingsipgate].
I also do not understand EXTEN:1 (should the second phone be EXTEN:2 ???)
starting the server with asterisk -vvvcg brings a lots of lines ;-)
sip show users:
sipgate.co.uk my_password2 incomingsipgate No Alway
sipgate.de my_password1 incomingsipgate No Alway
602 my_password4 incomingsipgate No Alway
601 my_password3 incomingsipgate No Alway
sip show registry:
sipgate.co.uk:5060 4782156 105 Registered
sipgate.de:5060 5552220 105 Registered
Tests:
601 calls 602 busy
004910000 busy (10000 at sipgate.de should play a tape)
No info on the screen (asterisk: *CLI> )
What have I forgotten / made wrong?
bye
Ronald
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BetaTeilchen
2004-Oct-23 23:46 UTC
[Asterisk-Users] Asterisk, ATA-186 & Sipgate.de / sipgate.co.uk
Ronald Wiplinger schrieb:> > [sipgate.de] > type=friend > username=5552220 > secret=my_password3 > host=sipgate.de > fromuser=5552220 > fromdomain=sipgate.netshould be "fromdomain=sipgate.de"> nat=yes > context=incomingsipgate > canreinvite=no > > > [sipgate.co.uk] > type=friend > username=4782156 > secret=my_password4 > host=sipgate.co.uk > fromuser=4782156 > fromdomain=sipgate.netshould be "fromdomain=sipgate.co.uk"> nat=yes > context=incomingsipgate > canreinvite=no > > > > > extensions.conf: > ==========> > [general] > static=yes > writeprotect=no > > [globals] > CONSOLE=Console/dsp ; Console interface for demo > > [incomingsipgate] > exten => h,1,Hangup > exten => 800XXXX,1,Dial(SIP/internestelefon,20,tr) >should be [incomingsipgate] exten => 5552220,1,Dial(SIP/internestelefon,20,r) exten => 4782156,1,Dial(SIP/internestelefon,20,r)> [sipgate.de] > exten => _0049.,Dial(SIP/${EXTEN:1}@sipgate.de,30,tr) > exten => _0049.,2,Playback(invalid) > exten => _0049.,3,Hangup >should be (you forgot to number prio 1 !) exten => _0049.,1,Dial(SIP/0${EXTEN:4}@sipgate.de,30,r) ; do not dial international prefix 0049 with Sipgate, if you call from same national net ! exten => _0049.,2,Playback(invalid) exten => _0049.,3,Hangup> [sipgate.co.uk] > exten => _0044.,Dial(SIP/${EXTEN:1}@sipgate.co.uk,30,tr) > exten => _0044.,2,Playback(invalid) > exten => _0044.,3,Hangup >exten => _0044.,1,Dial(SIP/0${EXTEN:4}@sipgate.co.uk,30,tr) ; do not dial international prefix 0044 with Sipgate, if you call from same national net ! exten => _0044.,2,Playback(invalid) exten => _0044.,3,Hangup> I did not understand the paragraph of [incomingsipgate].This is the context to which all incoming calls from Sipgate will be sent to be handled.> I also do not understand EXTEN:1 (should the second phone be EXTEN:2 > ???) >please regard correct expression ${EXTEN:1} ! This means "take the variable ($) called {EXTEN} (this is the dialed number) and cut the FIRST digit (:1)" So - if you dial 0049341234567 and then DIAL(SIP/${EXTEN:1}...) this will result in dialing 0493411234567 which is not a valid number. Regards -------------------------- Please visit http://www.ip-phone-forum.de --------------------------
Ronald Wiplinger
2004-Oct-24 00:43 UTC
[Asterisk-Users] Asterisk, ATA-186 & Sipgate.de / sipgate.co.uk
BetaTeilchen wrote:> Ronald Wiplinger schrieb:Thanks for helping me, but it still does not work.> >> >> [sipgate.de] >> type=friend >> username=5552220 >> secret=my_password3 >> host=sipgate.de >> fromuser=5552220 >> fromdomain=sipgate.net > > > > should be "fromdomain=sipgate.de" > >> nat=yes >> context=incomingsipgate >> canreinvite=no >> >> >> [sipgate.co.uk] >> type=friend >> username=4782156 >> secret=my_password4 >> host=sipgate.co.uk >> fromuser=4782156 >> fromdomain=sipgate.net > > > > should be "fromdomain=sipgate.co.uk" > >> nat=yes >> context=incomingsipgate >> canreinvite=no >> >> >> >> >> extensions.conf: >> ==========>> >> [general] >> static=yes >> writeprotect=no >> >> [globals] >> CONSOLE=Console/dsp ; Console interface for demo >> >> [incomingsipgate] >> exten => h,1,Hangup >> exten => 800XXXX,1,Dial(SIP/internestelefon,20,tr) >> > should be > > [incomingsipgate] > exten => 5552220,1,Dial(SIP/internestelefon,20,r) > exten => 4782156,1,Dial(SIP/internestelefon,20,r) >What is the difference between tr and r ? What does the 20 mean?> >> [sipgate.de] >> exten => _0049.,Dial(SIP/${EXTEN:1}@sipgate.de,30,tr) >> exten => _0049.,2,Playback(invalid) >> exten => _0049.,3,Hangup >> > should be > (you forgot to number prio 1 !) > > exten => _0049.,1,Dial(SIP/0${EXTEN:4}@sipgate.de,30,r) > ; do not dial international prefix 0049 with Sipgate, if you call from > same national net ! > exten => _0049.,2,Playback(invalid) > exten => _0049.,3,Hangup > > >> [sipgate.co.uk] >> exten => _0044.,Dial(SIP/${EXTEN:1}@sipgate.co.uk,30,tr) >> exten => _0044.,2,Playback(invalid) >> exten => _0044.,3,Hangup >> > exten => _0044.,1,Dial(SIP/0${EXTEN:4}@sipgate.co.uk,30,tr) > ; do not dial international prefix 0044 with Sipgate, if you call from > same national net ! > exten => _0044.,2,Playback(invalid) > exten => _0044.,3,Hangup > > >> I did not understand the paragraph of [incomingsipgate]. > > > > This is the context to which all incoming calls from Sipgate will be > sent to be handled. > >> I also do not understand EXTEN:1 (should the second phone be >> EXTEN:2 ???) >> > please regard correct expression ${EXTEN:1} ! > > This means "take the variable ($) called {EXTEN} (this is the dialed > number) and cut the FIRST digit (:1)" > So - if you dial 0049341234567 and then DIAL(SIP/${EXTEN:1}...) this > will result in dialing 0493411234567 which is not a valid number. >BTW, when I stop Asterisk with "stop now" I get a Yuck! Error in buffer handling ...: What does this mean? bye Ronald -------------- next part -------------- A non-text attachment was scrubbed... Name: ronald.wiplinger.vcf Type: text/x-vcard Size: 142 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041024/c970409c/ronald.wiplinger.vcf
BetaTeilchen
2004-Oct-24 00:55 UTC
[Asterisk-Users] Asterisk, ATA-186 & Sipgate.de / sipgate.co.uk
Maybe you should start reading here: http://www.voip-info.org/wiki-Asterisk+introduction to get basic knowledges of Asterisk Ronald Wiplinger schrieb:> BetaTeilchen wrote: > >> Ronald Wiplinger schrieb: > > > > Thanks for helping me, but it still does not work. > >> >>> >>> [sipgate.de] >>> type=friend >>> username=5552220 >>> secret=my_password3 >>> host=sipgate.de >>> fromuser=5552220 >>> fromdomain=sipgate.net >> >> >> >> >> should be "fromdomain=sipgate.de" >> >>> nat=yes >>> context=incomingsipgate >>> canreinvite=no >>> >>> >>> [sipgate.co.uk] >>> type=friend >>> username=4782156 >>> secret=my_password4 >>> host=sipgate.co.uk >>> fromuser=4782156 >>> fromdomain=sipgate.net >> >> >> >> >> should be "fromdomain=sipgate.co.uk" >> >>> nat=yes >>> context=incomingsipgate >>> canreinvite=no >>> >>> >>> >>> >>> extensions.conf: >>> ==========>>> >>> [general] >>> static=yes >>> writeprotect=no >>> >>> [globals] >>> CONSOLE=Console/dsp ; Console interface for demo >>> >>> [incomingsipgate] >>> exten => h,1,Hangup >>> exten => 800XXXX,1,Dial(SIP/internestelefon,20,tr) >>> >> should be >> >> [incomingsipgate] >> exten => 5552220,1,Dial(SIP/internestelefon,20,r) >> exten => 4782156,1,Dial(SIP/internestelefon,20,r) >> > > What is the difference between tr and r ? What does the 20 mean? > > >> >>> [sipgate.de] >>> exten => _0049.,Dial(SIP/${EXTEN:1}@sipgate.de,30,tr) >>> exten => _0049.,2,Playback(invalid) >>> exten => _0049.,3,Hangup >>> >> should be >> (you forgot to number prio 1 !) >> >> exten => _0049.,1,Dial(SIP/0${EXTEN:4}@sipgate.de,30,r) >> ; do not dial international prefix 0049 with Sipgate, if you call >> from same national net ! >> exten => _0049.,2,Playback(invalid) >> exten => _0049.,3,Hangup >> >> >>> [sipgate.co.uk] >>> exten => _0044.,Dial(SIP/${EXTEN:1}@sipgate.co.uk,30,tr) >>> exten => _0044.,2,Playback(invalid) >>> exten => _0044.,3,Hangup >>> >> exten => _0044.,1,Dial(SIP/0${EXTEN:4}@sipgate.co.uk,30,tr) >> ; do not dial international prefix 0044 with Sipgate, if you call >> from same national net ! >> exten => _0044.,2,Playback(invalid) >> exten => _0044.,3,Hangup >> >> >>> I did not understand the paragraph of [incomingsipgate]. >> >> >> >> >> This is the context to which all incoming calls from Sipgate will be >> sent to be handled. >> >>> I also do not understand EXTEN:1 (should the second phone be >>> EXTEN:2 ???) >>> >> please regard correct expression ${EXTEN:1} ! >> >> This means "take the variable ($) called {EXTEN} (this is the dialed >> number) and cut the FIRST digit (:1)" >> So - if you dial 0049341234567 and then DIAL(SIP/${EXTEN:1}...) this >> will result in dialing 0493411234567 which is not a valid number. >> > > BTW, when I stop Asterisk with "stop now" I get a > Yuck! Error in buffer handling ...: > What does this mean? > > bye > > Ronald > >_______________________________________________ >Asterisk-Users mailing list >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 >