Please look in any conf file that have any relations with sip.conf. I think you have some records. And one also, you take this message when calling in both direction? (server1 call server2 and server2 call server1)
Vardan Vieri wrote: > > > --- On Wed, 5/12/10, Vardan<[email protected]> wrote: > >> I have forget to write for outcall in >> extension >> >> server1: >> [calltoserver2] >> exten => _X.,1,Noop(Call to server2) >> exten => >> _X.,2,Dial(SIP/interboxserver2/${EXTEN}) >> exten => _X.,3,Hangup >> >> server2: >> >> [calltoserver1] >> exten => _X.,1,Noop(Call to server1) >> exten => >> _X.,2,Dial(SIP/interboxserver1/${EXTEN}) >> exten => _X.,3,Hangup >> >> :) >> >> Vardan >> >> >> Vardan wrote: >>> Hello >>> >>> Server1: >>> >>> sip.conf >>> >>> [interboxserver2] >>> type=friend >>> host=192.168.250.112 >>> context=callfromserver2 >>> disallow=all >>> allow=ulaw >>> allow=alaw >>> allow=g729 >>> >>> extensions.conf >>> >>> [callfromserver2] >>> >>> exten => _X.,1,Noop(Call from server2) >>> exten => _X.,2,Dial(SIP/${EXTEN}) >>> exten => _X.,3,Hangup >>> >>> >>> Server2: >>> >>> sip.conf >>> >>> [interboxserver1] >>> type=friend >>> host=192.168.250.111 >>> context=callfromserver1 >>> disallow=all >>> allow=ulaw >>> allow=alaw >>> allow=g729 >>> >>> extensions.conf >>> >>> [callfromserver1] >>> >>> exten => _X.,1,Noop(Call from server1) >>> exten => _X.,2,Dial(SIP/${EXTEN}) >>> exten => _X.,3,Hangup >>> >>> >>> Try so, I think it must work. >>> And also, look and delete any another records in both >> servers in >>> sip.conf about this servers settings. >>> >>> Vardan >>> >>> >>> Vieri wrote: >>>> Hi, >>>> >>>> I'm trying to setup a SIP trunk between 2 Asterisk >> servers on the same LAN (no NAT, no firewalls). >>>> >>>> With IAX2 all's fine but I'm unable to setup SIP. >> I must be missing something obvious. >>>> >>>> I followed the simple example at >>>> http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/. >>>> >>>> so Asterisk server 1 (192.168.250.111) sip.conf >> contains: >>>> >>>> [interboxsip] >>>> type=peer >>>> host=192.168.250.112 >>>> context=mycontext >>>> >>>> Asterisk server 2 (192.168.250.112) sip.conf >> contains: >>>> >>>> [interboxsip] >>>> type=peer >>>> host=192.168.250.111 >>>> context=mycontext >>>> >>>> I dialed from a SIP extension (4053) in server 2 >> (192.168.250.112) to 3666 in server 1 (192.168.250.111) via >> the interboxsip SIP trunk. >>>> >>>> The call fails and according to the SIP messages >> it seems to be an authentication problem. >>>> >>>> What am I missing? >>>> >>>> SIP messages on 192.168.250.112 (Asterisk server 2 >> - transmitting call): >>>> >>>> -- Executing >> [3...@from-internal:2] Dial("SIP/4053-00006dea", >> "SIP/interboxsip/3666|300|rt") in new stack >>>> Audio is at 192.168.250.112 port 15850 >>>> Adding codec 0x4 (ulaw) to SDP >>>> Adding codec 0x8 (alaw) to SDP >>>> Adding non-codec 0x1 (telephone-event) to SDP >>>> Reliably Transmitting (no NAT) to >> 192.168.250.111:5060: >>>> INVITE sip:[email protected] SIP/2.0 >>>> Via: SIP/2.0/UDP >> 192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport >>>> From: >> "device"<sip:[email protected]>;tag=as4d17a185 >>>> To:<sip:[email protected]> >>>> Contact:<sip:[email protected]> >>>> Call-ID: >> [email protected] >>>> CSeq: 102 INVITE >>>> User-Agent: Asterisk PBX >>>> Max-Forwards: 70 >>>> Date: Wed, 12 May 2010 09:13:06 GMT >>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, >> SUBSCRIBE, NOTIFY, INFO >>>> Supported: replaces >>>> Content-Type: application/sdp >>>> Content-Length: 270 >>>> >>>> v=0 >>>> o=root 20611 20611 IN IP4 192.168.250.112 >>>> s=session >>>> c=IN IP4 192.168.250.112 >>>> t=0 0 >>>> m=audio 15850 RTP/AVP 0 8 101 >>>> a=rtpmap:0 PCMU/8000 >>>> a=rtpmap:8 PCMA/8000 >>>> a=rtpmap:101 telephone-event/8000 >>>> a=fmtp:101 0-16 >>>> a=silenceSupp:off - - - - >>>> a=ptime:20 >>>> a=sendrecv >>>> >>>> --- >>>> -- Called >> interboxsip/3666 >>>> >>>> <--- SIP read from 192.168.250.111:5060 >> ---> >>>> SIP/2.0 407 Proxy Authentication Required >>>> Via: SIP/2.0/UDP >> 192.168.250.112:5060;branch=z9hG4bK3c951a1d;received=192.168.250.112;rport=5060 >>>> From: >> "device"<sip:[email protected]>;tag=as4d17a185 >>>> >> To:<sip:[email protected]>;tag=as00842b82 >>>> Call-ID: >> [email protected] >>>> CSeq: 102 INVITE >>>> User-Agent: Asterisk PBX >>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, >> SUBSCRIBE, NOTIFY >>>> Proxy-Authenticate: Digest algorithm=MD5, >> realm="asterisk", nonce="2545a5dd" >>>> Content-Length: 0 >>>> >>>> >>>> <-------------> >>>> >>>> --- (10 headers 0 lines) --- >>>> Transmitting (no NAT) to 192.168.250.111:5060: >>>> ACK sip:[email protected] SIP/2.0 >>>> Via: SIP/2.0/UDP >> 192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport >>>> From: >> "device"<sip:[email protected]>;tag=as4d17a185 >>>> >> To:<sip:[email protected]>;tag=as00842b82 >>>> Contact:<sip:[email protected]> >>>> Call-ID: >> [email protected] >>>> CSeq: 102 ACK >>>> User-Agent: Asterisk PBX >>>> Max-Forwards: 70 >>>> Content-Length: 0 >>>> >>>> >>>> --- >>>> -- >> SIP/interboxsip-00006deb is circuit-busy >>>> >>>> >>>> SIP messages on 192.168.250.111 (Asterisk server 1 >> - receiving end): >>>> >>>> <-- SIP read from 192.168.250.112:5060: >>>> INVITE sip:[email protected] SIP/2.0 >>>> Via: SIP/2.0/UDP >> 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport >>>> From: >> "device"<sip:[email protected]>;tag=as18a568d6 >>>> To:<sip:[email protected]> >>>> Contact:<sip:[email protected]> >>>> Call-ID: >> [email protected] >>>> CSeq: 102 INVITE >>>> User-Agent: Asterisk PBX >>>> Max-Forwards: 70 >>>> Date: Wed, 12 May 2010 09:20:26 GMT >>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, >> SUBSCRIBE, NOTIFY, INFO >>>> upported: replaces >>>> Content-Type: application/sdp >>>> Content-Length: 270 >>>> >>>> v=0 >>>> o=root 20611 20611 IN IP4 192.168.250.112 >>>> s=session >>>> c=IN IP4 192.168.250.112 >>>> t=0 0 >>>> m=audio 14648 RTP/AVP 0 8 101 >>>> a=rtpmap:0 PCMU/8000 >>>> a=rtpmap:8 PCMA/8000 >>>> a=rtpmap:101 telephone-event/8000 >>>> a=fmtp:101 0-16 >>>> a=silenceSupp:off - - - - >>>> a=ptime:20 >>>> a=sendrecv >>>> >>>> --- (14 headers 13 lines) --- >>>> Using INVITE request as basis request - >> [email protected] >>>> Sending to 192.168.250.112 : 5060 (NAT) >>>> Reliably Transmitting (NAT) to >> 192.168.250.112:5060: >>>> SIP/2.0 407 Proxy Authentication Required >>>> Via: SIP/2.0/UDP >> 192.168.250.112:5060;branch=z9hG4bK504ddbc7;received=192.168.250.112;rport=5060 >>>> From: >> "device"<sip:[email protected]>;tag=as18a568d6 >>>> >> To:<sip:[email protected]>;tag=as57a19dac >>>> Call-ID: >> [email protected] >>>> CSeq: 102 INVITE >>>> User-Agent: Asterisk PBX >>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, >> SUBSCRIBE, NOTIFY >>>> Proxy-Authenticate: Digest algorithm=MD5, >> realm="asterisk", nonce="1327c5b6" >>>> Content-Length: 0 >>>> >>>> >>>> --- >>>> Scheduling destruction of call >> '[email protected]' in 15000 >> ms >>>> Found user '4053' >>>> >>>> <-- SIP read from 192.168.250.112:5060: >>>> ACK sip:[email protected] SIP/2.0 >>>> Via: SIP/2.0/UDP >> 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport >>>> From: >> "device"<sip:[email protected]>;tag=as18a568d6 >>>> >> To:<sip:[email protected]>;tag=as57a19dac >>>> Contact:<sip:[email protected]> >>>> Call-ID: >> [email protected] >>>> CSeq: 102 ACK >>>> User-Agent: Asterisk PBX >>>> Max-Forwards: 70 >>>> Content-Length: 0 > > > Hi, > > I tried your suggestion (then I even added the insecure param) but I still > get the error: > > SIP/2.0 407 Proxy Authentication Required > > > on server 2: > > [interboxsip] > type=friend > insecure=invite > host=192.168.250.111 > context=mycontext > disallow=all > allow=ulaw > allow=alaw > allow=g729 > allow=gsm > > on server 1: > > [interboxsip] > type=friend > insecure=invite > host=192.168.250.112 > context=mycontext > disallow=all > allow=ulaw > allow=alaw > allow=g729 > allow=gsm > > to call from one server to the other: > > exten => 3666,1,Dial(SIP/interboxsip/${EXTEN},20,rt) > exten => 3666,n,HangUp() > > This should be simple but it puzzles me why it's not working. > > Thanks, > > Vieri > > > > > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? 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