Darryl Moore <darryl at moores.ca> schrieb:> Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like?Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register => 00493513333333:MYSECRET at pbxanika/00493513333333 register => 4444444444:MYSECRET at messagenet/4444444444 [pbxluca] type=peer defaultuser=00493511111111 secret= MYSECRET dtmfmode=rfc2833 host=172.16.34.132 context=luca_incoming outboundproxy=172.16.34.132 port=5060 fromuser=00493511111111 fromdomain=172.16.34.132 usereqphone=yes canreinvite=no insecure=invite [pbxfax] type=peer defaultuser=00493512222222 secret= MYSECRET dtmfmode=rfc2833 host=172.16.34.132 context=fax_incoming outboundproxy=172.16.34.132 port=5060 fromuser=00493512222222 fromdomain=172.16.34.132 usereqphone=yes canreinvite=no insecure=invite [pbxanika] type=peer defaultuser=00493513333333 secret= MYSECRET dtmfmode=rfc2833 host=172.16.34.132 context=anika_incoming outboundproxy=172.16.34.132 port=5060 fromuser=00493513333333 fromdomain=172.16.34.132 usereqphone=yes canreinvite=no insecure=invite [messagenet] type=peer defaultuser=4444444444 secret=MYSECRET dtmfmode=rfc2833 host=sip.messagenet.it context=messagenet_incoming outboundproxy=sip.messagenet.it port=5061 fromuser=4444444444 fromdomain=sip.messagenet.it usereqphone=yes canreinvite=no insecure=invite Here my extensions.conf: [stdexten] include => luca_incoming include => fax_incoming include => anika_incoming include => messagenet_incoming [luca_incoming] exten => _00493511111111,1,Verbose(2,Call for Luca) exten => _00493511111111,n,Dial(SIP/00493511111111) exten => _00493511111111,n,Hangup [fax_incoming] exten => _00493512222222,1,Verbose(2,Call for FAX) exten => _00493512222222,n,Dial(SIP/00493512222222) exten => _00493512222222,n,Hangup [anika_incoming] exten => _00493513333333,1,Verbose(2,Call for Anika) exten => _00493513333333,n,Dial(SIP/00493513333333) exten => _00493513333333,n,Hangup [messagenet_incoming] exten => _4444444444,1,Verbose(2,Call from Messagenet) exten => _4444444444,n,Dial(SIP/00493511111111) exten => _4444444444,n,Hangup [myproxy] exten => _X.,1,Verbose(2,Call from ${CALLERID(num)} to ${EXTEN}) exten => _X.,n,GotoIf($["${CALLERID(num)}" = "00493511111111"]?dialluca) exten => _X.,n,GotoIf($["${CALLERID(num)}" = "00493512222222"]?dialfax) exten => _X.,n,GotoIf($["${CALLERID(num)}" = "00493513333333"]?dialanika) exten => _X.,n,Dial(SIP/pbxluca/${EXTEN},30,r) exten => _X.,n,Hangup exten => _X.,n(dialluca),Verbose(2,Outgoing using pbxluca) exten => _X.,n(dialluca),Dial(SIP/pbxluca/${EXTEN},30,r) exten => _X.,n,Hangup exten => _X.,n(dialfax),Verbose(2,Outgoing using pbxfax) exten => _X.,n(dialfax),Dial(SIP/pbxfax/${EXTEN},30,r) exten => _X.,n,Hangup exten => _X.,n(dialanika),Verbose(2,Outgoing using pbxanika) exten => _X.,n(dialanika),Dial(SIP/pbxanika/${EXTEN},30,r) exten => _X.,n,Hangup And here my users.conf: [00493511111111] fullname = luca secret = MYSECRET dahdichan = 1 hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = no context = myproxy host = dynamic dtmfmode=rfc2833 canreinvite=no sendrpid=pai type=friend nat=force_rport,comedia qualify=yes qualifyfreq=60 transport=Auto avpf=no force_avp=no icesupport=no encryption=no callgrouppickupgroupdial=SIP/00493511111111 [00493512222222] fullname = fax secret = MYSECRET dahdichan = 1 hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = no context = myproxy host = dynamic dtmfmode=rfc2833 canreinvite=no sendrpid=pai type=friend nat=force_rport,comedia qualify=yes qualifyfreq=60 transport=Auto avpf=no force_avp=no icesupport=no encryption=no callgrouppickupgroupdial=SIP/00493512222222 [00493513333333] fullname = anika secret = MYSECRET dahdichan = 1 hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = no context = myproxy host = dynamic dtmfmode=rfc2833 canreinvite=no sendrpid=pai type=friend nat=force_rport,comedia qualify=yes qualifyfreq=60 transport=Auto avpf=no force_avp=no icesupport=no encryption=no callgrouppickupgroupdial=SIP/00493513333333 Now I see this: if I call my phone (00493511111111) from Twinkle it works. If I call it from the phone of my wife, logged in on the same AsteriskNOW of Twinkle and able to speak with Twinkle, it does NOT work and I see that in the Log of my Asterisk: == Using SIP RTP CoS mark 5 [May 28 23:05:59] WARNING[4135]: chan_sip.c:12800 check_auth: username mismatch, have <1234>, digest has <luca> [May 28 23:05:59] NOTICE[4135]: chan_sip.c:20083 handle_request_invite: Failed to authenticate device "Test1" <sip:1234 at 172.16.34.132>;tag=as7855ffe5 (the phone of my wife is now logged in on AsteriskNOW with the user "1234" and try to call my phone with the same number I use from Twinkle, which works). Very puzzled... Thanks Luca Bertoncello (lucabert at lucabert.de)
I think your phone may be trying to register with the username '1234', while your sip configuration is expecting 'luca'. Can you try changing your phone registration credentials to use 'luca'? Can you give us a sip transcript when you try to place a call from it? On 15-05-28 05:09 PM, Luca Bertoncello wrote:> Darryl Moore <darryl at moores.ca> schrieb: > >> Ahh. Seen that before! That suggests to me that you don't have your >> sip.conf records setup right. >> >> What's your sip.conf look like? > Well, here what I wrote in my sip.conf: > > register => 00493511111111:MYSECRET at pbxluca/00493511111111 > register => 00493512222222:MYSECRET at pbxfax/00493512222222 > register => 00493513333333:MYSECRET at pbxanika/00493513333333 > register => 4444444444:MYSECRET at messagenet/4444444444 > > [pbxluca] > type=peer > defaultuser=00493511111111 > secret= MYSECRET > dtmfmode=rfc2833 > host=172.16.34.132 > context=luca_incoming > outboundproxy=172.16.34.132 > port=5060 > fromuser=00493511111111 > fromdomain=172.16.34.132 > usereqphone=yes > canreinvite=no > insecure=invite > > [pbxfax] > type=peer > defaultuser=00493512222222 > secret= MYSECRET > dtmfmode=rfc2833 > host=172.16.34.132 > context=fax_incoming > outboundproxy=172.16.34.132 > port=5060 > fromuser=00493512222222 > fromdomain=172.16.34.132 > usereqphone=yes > canreinvite=no > insecure=invite > > [pbxanika] > type=peer > defaultuser=00493513333333 > secret= MYSECRET > dtmfmode=rfc2833 > host=172.16.34.132 > context=anika_incoming > outboundproxy=172.16.34.132 > port=5060 > fromuser=00493513333333 > fromdomain=172.16.34.132 > usereqphone=yes > canreinvite=no > insecure=invite > > [messagenet] > type=peer > defaultuser=4444444444 > secret=MYSECRET > dtmfmode=rfc2833 > host=sip.messagenet.it > context=messagenet_incoming > outboundproxy=sip.messagenet.it > port=5061 > fromuser=4444444444 > fromdomain=sip.messagenet.it > usereqphone=yes > canreinvite=no > insecure=invite > > > Here my extensions.conf: > > [stdexten] > include => luca_incoming > include => fax_incoming > include => anika_incoming > include => messagenet_incoming > > [luca_incoming] > exten => _00493511111111,1,Verbose(2,Call for Luca) > exten => _00493511111111,n,Dial(SIP/00493511111111) > exten => _00493511111111,n,Hangup > > [fax_incoming] > exten => _00493512222222,1,Verbose(2,Call for FAX) > exten => _00493512222222,n,Dial(SIP/00493512222222) > exten => _00493512222222,n,Hangup > > [anika_incoming] > exten => _00493513333333,1,Verbose(2,Call for Anika) > exten => _00493513333333,n,Dial(SIP/00493513333333) > exten => _00493513333333,n,Hangup > > [messagenet_incoming] > exten => _4444444444,1,Verbose(2,Call from Messagenet) > exten => _4444444444,n,Dial(SIP/00493511111111) > exten => _4444444444,n,Hangup > > [myproxy] > exten => _X.,1,Verbose(2,Call from ${CALLERID(num)} to ${EXTEN}) > exten => _X.,n,GotoIf($["${CALLERID(num)}" = "00493511111111"]?dialluca) > exten => _X.,n,GotoIf($["${CALLERID(num)}" = "00493512222222"]?dialfax) > exten => _X.,n,GotoIf($["${CALLERID(num)}" = "00493513333333"]?dialanika) > exten => _X.,n,Dial(SIP/pbxluca/${EXTEN},30,r) > exten => _X.,n,Hangup > exten => _X.,n(dialluca),Verbose(2,Outgoing using pbxluca) > exten => _X.,n(dialluca),Dial(SIP/pbxluca/${EXTEN},30,r) > exten => _X.,n,Hangup > exten => _X.,n(dialfax),Verbose(2,Outgoing using pbxfax) > exten => _X.,n(dialfax),Dial(SIP/pbxfax/${EXTEN},30,r) > exten => _X.,n,Hangup > exten => _X.,n(dialanika),Verbose(2,Outgoing using pbxanika) > exten => _X.,n(dialanika),Dial(SIP/pbxanika/${EXTEN},30,r) > exten => _X.,n,Hangup > > And here my users.conf: > > [00493511111111] > fullname = luca > secret = MYSECRET > dahdichan = 1 > hassip = yes > hasiax = no > hash323 = no > hasmanager = no > callwaiting = no > context = myproxy > host = dynamic > dtmfmode=rfc2833 > canreinvite=no > sendrpid=pai > type=friend > nat=force_rport,comedia > qualify=yes > qualifyfreq=60 > transport=Auto > avpf=no > force_avp=no > icesupport=no > encryption=no > callgroup> pickupgroup> dial=SIP/00493511111111 > > [00493512222222] > fullname = fax > secret = MYSECRET > dahdichan = 1 > hassip = yes > hasiax = no > hash323 = no > hasmanager = no > callwaiting = no > context = myproxy > host = dynamic > dtmfmode=rfc2833 > canreinvite=no > sendrpid=pai > type=friend > nat=force_rport,comedia > qualify=yes > qualifyfreq=60 > transport=Auto > avpf=no > force_avp=no > icesupport=no > encryption=no > callgroup> pickupgroup> dial=SIP/00493512222222 > > [00493513333333] > fullname = anika > secret = MYSECRET > dahdichan = 1 > hassip = yes > hasiax = no > hash323 = no > hasmanager = no > callwaiting = no > context = myproxy > host = dynamic > dtmfmode=rfc2833 > canreinvite=no > sendrpid=pai > type=friend > nat=force_rport,comedia > qualify=yes > qualifyfreq=60 > transport=Auto > avpf=no > force_avp=no > icesupport=no > encryption=no > callgroup> pickupgroup> dial=SIP/00493513333333 > > > Now I see this: if I call my phone (00493511111111) from Twinkle it works. > If I call it from the phone of my wife, logged in on the same AsteriskNOW of > Twinkle and able to speak with Twinkle, it does NOT work and I see that in the > Log of my Asterisk: > > == Using SIP RTP CoS mark 5 > [May 28 23:05:59] WARNING[4135]: chan_sip.c:12800 check_auth: username mismatch, have <1234>, digest has <luca> > [May 28 23:05:59] NOTICE[4135]: chan_sip.c:20083 handle_request_invite: Failed to authenticate device "Test1" <sip:1234 at 172.16.34.132>;tag=as7855ffe5 > > (the phone of my wife is now logged in on AsteriskNOW with the user "1234" and try > to call my phone with the same number I use from Twinkle, which works). > > Very puzzled... > > Thanks > Luca Bertoncello > (lucabert at lucabert.de) >
Darryl Moore <darryl at moores.ca> schrieb:> I think your phone may be trying to register with the username '1234', > while your sip configuration is expecting 'luca'. Can you try changing > your phone registration credentials to use 'luca'? Can you give us a sip > transcript when you try to place a call from it?Well, right now this phone USES the username 1234, on the AsteriskNOW (the "later Telekom"). I really don't know why it tries to authenticate to my "own Asterisk"... What I see right now, if I try to connect the phone of my wife to "my own Asterisk": -- Registered SIP '00493512222222' at 192.168.200.11 port 5060 [May 28 23:46:01] NOTICE[1350]: chan_sip.c:22933 sip_poke_noanswer: Peer '00493512222222' is now UNREACHABLE! Last qualify: 0 But, as I said, right now the phone is connected to the AsteriskNOW... Well, now I must sleep... Hope someone can suggest me something that I can try tomorrow. Thanks a lot Luca Bertoncello (lucabert at lucabert.de)
Darryl Moore <darryl at moores.ca> schrieb:> I think your phone may be trying to register with the username '1234', > while your sip configuration is expecting 'luca'. Can you try changing > your phone registration credentials to use 'luca'? Can you give us a sip > transcript when you try to place a call from it?Well, another information (then I **MUST** go sleep...): I tried to use my mobile phone logging to my "own Asterisk" with the login data of my wife's telefon. Now this user is REACHABLE... So I think, it was a problem on her phone... I can't call and receive calls. I think, that it's a problem of my Dialplan. If I try to call the mobile phone from AsteriskNOW (later: "the world"), I see that in Asterisk's log ("my own Asterisk"): == Using SIP RTP CoS mark 5 [May 29 00:07:49] NOTICE[1106]: chan_sip.c:20163 handle_request_invite: Call from '00493511111111' to extension '00493512222222' rejected because extension not found. That's very strange, since I call from Twinkle and it has the number "1234"... If I call my mobile phone using my VoIP-phone (connected on the same "my own Asterisk") I get that: == Using SIP RTP CoS mark 5 == Call from 00493511111111 to 00493512222222 == Outgoing using pbxluca == Using SIP RTP CoS mark 5 == Using SIP RTP CoS mark 5 [May 29 00:09:25] WARNING[1106]: chan_sip.c:12800 check_auth: username mismatch, have <00493511111111>, digest has <00493512222222> [May 29 00:09:25] NOTICE[1106]: chan_sip.c:20083 handle_request_invite: Failed to authenticate device "00493511111111" <sip:00493511111111 at 172.16.34.132>;tag=as058adbf2 == Everyone is busy/congested at this time (1:0/1/0) == Spawn extension (myproxy, 00493512222222, 9) exited non-zero on 'SIP/00493511111111-00000004' Maybe this is the same problem, since I didn't configured my own Asterisk to manage "internal calls" (since I don't need to call my wife on VoIP... :D) And, last but not least, if I try to call from my mobile phone Twinkle I get this: == Using SIP RTP CoS mark 5 == Call from 00493512222222 to 1234 == Outgoing using pbxanika == Using SIP RTP CoS mark 5 == Everyone is busy/congested at this time (1:0/1/0) == Spawn extension (myproxy, 1234, 15) exited non-zero on 'SIP/00493512222222-00000006' And if I try to call my VoIP-phone I get that: == Using SIP RTP CoS mark 5 == Call from 00493512222222 to 00493511111111 == Outgoing using pbxanika == Using SIP RTP CoS mark 5 == Using SIP RTP CoS mark 5 [May 29 00:12:02] WARNING[1106]: chan_sip.c:12800 check_auth: username mismatch, have <00493512222222>, digest has <00493511111111> [May 29 00:12:02] NOTICE[1106]: chan_sip.c:20083 handle_request_invite: Failed to authenticate device "00493512222222" <sip:00493512222222 at 172.16.34.132>;tag=as193c26b0 == Everyone is busy/congested at this time (1:0/1/0) == Spawn extension (myproxy, 00493511111111, 15) exited non-zero on 'SIP/00493512222222-0000000a' Maybe can these information help someone helping me? Thanks a lot! Luca Bertoncello (lucabert at lucabert.de)