JR Richardson
2006-Nov-07 18:09 UTC
[asterisk-users] Asterisk and Max TNT Passing calls SIP to PRI, not PRI to SIP Authentication Issue
Hi All, I have a lab setup with two asterisk servers and a MAX TNT in the middle like this: asterisk sip >< sip TNT pri >< pri asterisk The TNT is running 11.0.6 and the asterisk servers are running 1.2.9.1. I can get calls to pass from asterisk sip to tnt to pri to asterisk but not the other way. The call from asterisk to pri to tnt is good, the TNT is passing SIP invite to the SIP Asterisk server. I have tried many variations of using sip options insecure, autocreatepeer, permit/deny, host, user, etc.... but can't seem to get asterisk to accept an unauthenticated call from the TNT using SIP. I keep getting SIP/2.0 407 Proxy Authentication Required. I know others have done this, but with older Asterisk versions, I'm wondering what versions of Asterisk are known to work with the MAX TNT and with what version of the TNT? I'm confident this is an asterisk issue, with insecure=very, I should be able to pass calls to asterisk without trying to authenticate it first. But this is not happening. Here is a debug of a call and a snip from my sip.conf: sip.conf [maxtnt] type=friend host=10.10.14.131 insecure=very dtmfmode=inband callerid="MaxTNT" <maxtnt> context=trunktntin qualify=yes reinvite=no canreinvite=no disallow=all allow=ulaw debug lab1*CLI> <-- SIP read from 10.10.14.131:5060: INVITE sip:2145551212@10.10.14.121:5060;user=phone SIP/2.0 t: <sip:2145551212@10.10.14.121:5060;user=phone> f: "NO CID NAME" <sip:1239@10.10.14.131:5060;user=phone>;tag=5fe9f589-1fb1f65c-830e0a0a Remote-Party-Id: "NO CID NAME" <sip:1239@10.10.14.131:5060;user=phone>;screen=no;id-type=subscriber;party=calling;privacy=off i: 3a8884d9-64-1fb1f65c@10.10.14.131 CSeq: 639089 INVITE v: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d Max-Forwards: 70 m: <sip:1239@10.10.14.131:5060;user=phone> k: replaces c: application/sdp Accept: application/sdp Accept-Encoding: Accept-Language: en User-Agent: Lucent-Universal-Gateway l: 232 v=0 o=t1gw01 531756636 531756636 IN IP4 10.10.14.131 s=Session SDP c=IN IP4 10.10.14.131 t=0 0 m=audio 40198 RTP/AVP 0 96 a=silenceSupp:on a=ecan:b on g168 a=ptime:20 a=rtpmap:96 telephone-event/8000 a=rtpmap:0 PCMU/8000 --- (16 headers 11 lines) --- Using INVITE request as basis request - 3a8884d9-64-1fb1f65c@10.10.14.131 Sending to 10.10.14.131 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 10.10.14.131:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d;received=10.10.14.131 From: "NO CID NAME" <sip:1239@10.10.14.131:5060;user=phone>;tag=5fe9f589-1fb1f65c-830e0a0a To: <sip:2145551212@10.10.14.121:5060;user=phone>;tag=as41f8454e Call-ID: 3a8884d9-64-1fb1f65c@10.10.14.131 CSeq: 639089 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3ea7e98a" Content-Length: 0 --- Scheduling destruction of call '3a8884d9-64-1fb1f65c@10.10.14.131' in 15000 ms Found user '1239' lab1*CLI> <-- SIP read from 10.10.14.131:5060: ACK sip:2145551212@10.10.14.121:5060;user=phone SIP/2.0 t: <sip:2145551212@10.10.14.121:5060;user=phone>;tag=as41f8454e f: "NO CID NAME" <sip:1239@10.10.14.131:5060;user=phone>;tag=5fe9f589-1fb1f65c-830e0a0a i: 3a8884d9-64-1fb1f65c@10.10.14.131 CSeq: 639089 ACK v: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d Max-Forwards: 70 User-Agent: Lucent-Universal-Gateway l: 0 Any guidance will be much appreciated. Thanks. JR -- JR Richardson Engineering for the Masses
JR Richardson
2006-Nov-07 18:10 UTC
[asterisk-users] Re: Asterisk and Max TNT Passing calls SIP to PRI, not PRI to SIP Authentication Issue
Update, I loaded asterisk 1.0.10 and it worked straight away. I can send unauthenticated calls to asterisk. Something in 1.2.9.1 and 1.2.13 are not allowing unauthenticated calls when insecure=very is set in sip.conf, either in the global or peer context. Are there any switches in the Asterisk Makefile to allow this? JR On 11/7/06, JR Richardson <jmr.richardson@gmail.com> wrote:> Hi All, > > I have a lab setup with two asterisk servers and a MAX TNT in the > middle like this: > > asterisk sip >< sip TNT pri >< pri asterisk > > The TNT is running 11.0.6 and the asterisk servers are running > 1.2.9.1. I can get calls to pass from asterisk sip to tnt to pri to > asterisk but not the other way. The call from asterisk to pri to tnt > is good, the TNT is passing SIP invite to the SIP Asterisk server. I > have tried many variations of using sip options insecure, > autocreatepeer, permit/deny, host, user, etc.... but can't seem to get > asterisk to accept an unauthenticated call from the TNT using SIP. I > keep getting SIP/2.0 407 Proxy Authentication Required. I know others > have done this, but with older Asterisk versions, I'm wondering what > versions of Asterisk are known to work with the MAX TNT and with what > version of the TNT? > > I'm confident this is an asterisk issue, with insecure=very, I should > be able to pass calls to asterisk without trying to authenticate it > first. But this is not happening. > > Here is a debug of a call and a snip from my sip.conf: > > sip.conf > > [maxtnt] > type=friend > host=10.10.14.131 > insecure=very > dtmfmode=inband > callerid="MaxTNT" <maxtnt> > context=trunktntin > qualify=yes > reinvite=no > canreinvite=no > disallow=all > allow=ulaw > > debug > > lab1*CLI> > <-- SIP read from 10.10.14.131:5060: > INVITE sip:2145551212@10.10.14.121:5060;user=phone SIP/2.0 > t: <sip:2145551212@10.10.14.121:5060;user=phone> > f: "NO CID NAME" > <sip:1239@10.10.14.131:5060;user=phone>;tag=5fe9f589-1fb1f65c-830e0a0a > Remote-Party-Id: "NO CID NAME" > <sip:1239@10.10.14.131:5060;user=phone>;screen=no;id-type=subscriber;party=calling;privacy=off > i: 3a8884d9-64-1fb1f65c@10.10.14.131 > CSeq: 639089 INVITE > v: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d > Max-Forwards: 70 > m: <sip:1239@10.10.14.131:5060;user=phone> > k: replaces > c: application/sdp > Accept: application/sdp > Accept-Encoding: > Accept-Language: en > User-Agent: Lucent-Universal-Gateway > l: 232 > > v=0 > o=t1gw01 531756636 531756636 IN IP4 10.10.14.131 > s=Session SDP > c=IN IP4 10.10.14.131 > t=0 0 > m=audio 40198 RTP/AVP 0 96 > a=silenceSupp:on > a=ecan:b on g168 > a=ptime:20 > a=rtpmap:96 telephone-event/8000 > a=rtpmap:0 PCMU/8000 > > --- (16 headers 11 lines) --- > Using INVITE request as basis request - 3a8884d9-64-1fb1f65c@10.10.14.131 > Sending to 10.10.14.131 : 5060 (non-NAT) > Reliably Transmitting (no NAT) to 10.10.14.131:5060: > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP > 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d;received=10.10.14.131 > From: "NO CID NAME" > <sip:1239@10.10.14.131:5060;user=phone>;tag=5fe9f589-1fb1f65c-830e0a0a > To: <sip:2145551212@10.10.14.121:5060;user=phone>;tag=as41f8454e > Call-ID: 3a8884d9-64-1fb1f65c@10.10.14.131 > CSeq: 639089 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3ea7e98a" > Content-Length: 0 > > > --- > Scheduling destruction of call '3a8884d9-64-1fb1f65c@10.10.14.131' in 15000 ms > Found user '1239' > lab1*CLI> > <-- SIP read from 10.10.14.131:5060: > ACK sip:2145551212@10.10.14.121:5060;user=phone SIP/2.0 > t: <sip:2145551212@10.10.14.121:5060;user=phone>;tag=as41f8454e > f: "NO CID NAME" > <sip:1239@10.10.14.131:5060;user=phone>;tag=5fe9f589-1fb1f65c-830e0a0a > i: 3a8884d9-64-1fb1f65c@10.10.14.131 > CSeq: 639089 ACK > v: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d > Max-Forwards: 70 > User-Agent: Lucent-Universal-Gateway > l: 0 > > > Any guidance will be much appreciated. > > Thanks. > > JR > > -- > JR Richardson > Engineering for the Masses >-- JR Richardson Engineering for the Masses
Barry Fawthrop
2006-Nov-07 18:54 UTC
[asterisk-users] Asterisk and Max TNT Passing calls SIP to PRI, not PRI to SIP Authentication Issue
what is the sip.conf for 1239 which I'm going to assume is a extension on the TNT Barry JR Richardson wrote:> Hi All, > > I have a lab setup with two asterisk servers and a MAX TNT in the > middle like this: > > asterisk sip >< sip TNT pri >< pri asterisk > > The TNT is running 11.0.6 and the asterisk servers are running > 1.2.9.1. I can get calls to pass from asterisk sip to tnt to pri to > asterisk but not the other way. The call from asterisk to pri to tnt > is good, the TNT is passing SIP invite to the SIP Asterisk server. I > have tried many variations of using sip options insecure, > autocreatepeer, permit/deny, host, user, etc.... but can't seem to get > asterisk to accept an unauthenticated call from the TNT using SIP. I > keep getting SIP/2.0 407 Proxy Authentication Required. I know others > have done this, but with older Asterisk versions, I'm wondering what > versions of Asterisk are known to work with the MAX TNT and with what > version of the TNT? > > I'm confident this is an asterisk issue, with insecure=very, I should > be able to pass calls to asterisk without trying to authenticate it > first. But this is not happening. > > Here is a debug of a call and a snip from my sip.conf: > > sip.conf > > [maxtnt] > type=friend > host=10.10.14.131 > insecure=very > dtmfmode=inband > callerid="MaxTNT" <maxtnt> > context=trunktntin > qualify=yes > reinvite=no > canreinvite=no > disallow=all > allow=ulaw > > debug > > lab1*CLI> > <-- SIP read from 10.10.14.131:5060: > INVITE sip:2145551212@10.10.14.121:5060;user=phone SIP/2.0 > t: <sip:2145551212@10.10.14.121:5060;user=phone> > f: "NO CID NAME" > <sip:1239@10.10.14.131:5060;user=phone>;tag=5fe9f589-1fb1f65c-830e0a0a > Remote-Party-Id: "NO CID NAME" > <sip:1239@10.10.14.131:5060;user=phone>;screen=no;id-type=subscriber;party=calling;privacy=off > > i: 3a8884d9-64-1fb1f65c@10.10.14.131 > CSeq: 639089 INVITE > v: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d > Max-Forwards: 70 > m: <sip:1239@10.10.14.131:5060;user=phone> > k: replaces > c: application/sdp > Accept: application/sdp > Accept-Encoding: > Accept-Language: en > User-Agent: Lucent-Universal-Gateway > l: 232 > > v=0 > o=t1gw01 531756636 531756636 IN IP4 10.10.14.131 > s=Session SDP > c=IN IP4 10.10.14.131 > t=0 0 > m=audio 40198 RTP/AVP 0 96 > a=silenceSupp:on > a=ecan:b on g168 > a=ptime:20 > a=rtpmap:96 telephone-event/8000 > a=rtpmap:0 PCMU/8000 > > --- (16 headers 11 lines) --- > Using INVITE request as basis request - 3a8884d9-64-1fb1f65c@10.10.14.131 > Sending to 10.10.14.131 : 5060 (non-NAT) > Reliably Transmitting (no NAT) to 10.10.14.131:5060: > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP > 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d;received=10.10.14.131 > From: "NO CID NAME" > <sip:1239@10.10.14.131:5060;user=phone>;tag=5fe9f589-1fb1f65c-830e0a0a > To: <sip:2145551212@10.10.14.121:5060;user=phone>;tag=as41f8454e > Call-ID: 3a8884d9-64-1fb1f65c@10.10.14.131 > CSeq: 639089 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", > nonce="3ea7e98a" > Content-Length: 0 > > > --- > Scheduling destruction of call '3a8884d9-64-1fb1f65c@10.10.14.131' in > 15000 ms > Found user '1239' > lab1*CLI> > <-- SIP read from 10.10.14.131:5060: > ACK sip:2145551212@10.10.14.121:5060;user=phone SIP/2.0 > t: <sip:2145551212@10.10.14.121:5060;user=phone>;tag=as41f8454e > f: "NO CID NAME" > <sip:1239@10.10.14.131:5060;user=phone>;tag=5fe9f589-1fb1f65c-830e0a0a > i: 3a8884d9-64-1fb1f65c@10.10.14.131 > CSeq: 639089 ACK > v: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d > Max-Forwards: 70 > User-Agent: Lucent-Universal-Gateway > l: 0 > > > Any guidance will be much appreciated. > > Thanks. > > JR >
Scott Keagy
2006-Nov-07 18:59 UTC
[asterisk-users] Asterisk and Max TNT Passing calls SIP to PRI, not PRI to SIP Authentication Issue
When all else fails I resort to adding this in the sip.conf peer config: Insecure=invite,port It took me a while to figure out they can be used together. Regards, Scott ----- Original Message ----- From: asterisk-users-bounces@lists.digium.com <asterisk-users-bounces@lists.digium.com> To: asterisk-users@lists.digium.com <asterisk-users@lists.digium.com> Sent: Tue Nov 07 15:23:26 2006 Subject: [asterisk-users] Asterisk and Max TNT Passing calls SIP to PRI,not PRI to SIP Authentication Issue Hi All, I have a lab setup with two asterisk servers and a MAX TNT in the middle like this: asterisk sip >< sip TNT pri >< pri asterisk The TNT is running 11.0.6 and the asterisk servers are running 1.2.9.1. I can get calls to pass from asterisk sip to tnt to pri to asterisk but not the other way. The call from asterisk to pri to tnt is good, the TNT is passing SIP invite to the SIP Asterisk server. I have tried many variations of using sip options insecure, autocreatepeer, permit/deny, host, user, etc.... but can't seem to get asterisk to accept an unauthenticated call from the TNT using SIP. I keep getting SIP/2.0 407 Proxy Authentication Required. I know others have done this, but with older Asterisk versions, I'm wondering what versions of Asterisk are known to work with the MAX TNT and with what version of the TNT? I'm confident this is an asterisk issue, with insecure=very, I should be able to pass calls to asterisk without trying to authenticate it first. But this is not happening. Here is a debug of a call and a snip from my sip.conf: sip.conf [maxtnt] type=friend host=10.10.14.131 insecure=very dtmfmode=inband callerid="MaxTNT" <maxtnt> context=trunktntin qualify=yes reinvite=no canreinvite=no disallow=all allow=ulaw debug lab1*CLI> <-- SIP read from 10.10.14.131:5060: INVITE sip:2145551212@10.10.14.121:5060;user=phone SIP/2.0 t: <sip:2145551212@10.10.14.121:5060;user=phone> f: "NO CID NAME" <sip:1239@10.10.14.131:5060;user=phone>;tag=5fe9f589-1fb1f65c-830e0a0a Remote-Party-Id: "NO CID NAME" <sip:1239@10.10.14.131:5060;user=phone>;screen=no;id-type=subscriber;party=calling;privacy=off i: 3a8884d9-64-1fb1f65c@10.10.14.131 CSeq: 639089 INVITE v: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d Max-Forwards: 70 m: <sip:1239@10.10.14.131:5060;user=phone> k: replaces c: application/sdp Accept: application/sdp Accept-Encoding: Accept-Language: en User-Agent: Lucent-Universal-Gateway l: 232 v=0 o=t1gw01 531756636 531756636 IN IP4 10.10.14.131 s=Session SDP c=IN IP4 10.10.14.131 t=0 0 m=audio 40198 RTP/AVP 0 96 a=silenceSupp:on a=ecan:b on g168 a=ptime:20 a=rtpmap:96 telephone-event/8000 a=rtpmap:0 PCMU/8000 --- (16 headers 11 lines) --- Using INVITE request as basis request - 3a8884d9-64-1fb1f65c@10.10.14.131 Sending to 10.10.14.131 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 10.10.14.131:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d;received=10.10.14.131 From: "NO CID NAME" <sip:1239@10.10.14.131:5060;user=phone>;tag=5fe9f589-1fb1f65c-830e0a0a To: <sip:2145551212@10.10.14.121:5060;user=phone>;tag=as41f8454e Call-ID: 3a8884d9-64-1fb1f65c@10.10.14.131 CSeq: 639089 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3ea7e98a" Content-Length: 0 --- Scheduling destruction of call '3a8884d9-64-1fb1f65c@10.10.14.131' in 15000 ms Found user '1239' lab1*CLI> <-- SIP read from 10.10.14.131:5060: ACK sip:2145551212@10.10.14.121:5060;user=phone SIP/2.0 t: <sip:2145551212@10.10.14.121:5060;user=phone>;tag=as41f8454e f: "NO CID NAME" <sip:1239@10.10.14.131:5060;user=phone>;tag=5fe9f589-1fb1f65c-830e0a0a i: 3a8884d9-64-1fb1f65c@10.10.14.131 CSeq: 639089 ACK v: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d Max-Forwards: 70 User-Agent: Lucent-Universal-Gateway l: 0 Any guidance will be much appreciated. Thanks. JR -- JR Richardson Engineering for the Masses _______________________________________________ --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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061107/802f1ce1/attachment.htm