Peter Hoppe
2011-Jul-25 17:05 UTC
[asterisk-users] Registration problems, Linksys SPA 3102 on Asterisk 1.4.20
Sorry, I am resending this, I tried earlier, but I couldn't see it appear on the archives - apologogies if it appears double! -------------------------------------------------- My Sipura 3000 ATA died on me this morning. I had a Linksys SPA 3102 available which I would like to use as a replacement. Unfortunately, the SPA3102 is not able to register with the asterisk server - I am always getting a> SIP/2.0 401 Unauthorizedresponse from the asterisk server upon which the SPA3102 unit reports on its web interface that the registration FAILED. * I checked (and double checked) whether the given credentials stored in the SPA unit match the required ones as defined in the server's sip.conf file, and they do match. * I also upgraded the SPA's firmware from the older 3.6 version to 5.1.10 (GW) * During my research in different forums I found some posts that hinted on using TCP instead of UDP for SIP transport, so I set> PSTN Line -> SIP Settings -> SIP Transportto> TCPbut no success. I still get the 401 error response from the server. I wonder how to get the SPA unit to successfully register with my asterisk server. 1. Are there any settings (either on the SPA unit or the asterisk server) which I have overlooked? 2. Is there some sort of compatibility issue between the SPA3102 and asterisk 1.4.20? Below I posted some more details. Thank you so much for your consideration, help is very much appreciated! Peter Hoppe 1. sip.conf ============> [general] > ; --------------------------------------------------------------------------------- > ; 1.1 General setup > ; > bindaddr = 192.168.0.1 > port = 5060 > tos = none > > ; --------------------------------------------------------------------------------- > ; 1.2 Jitter buffer configuration > ; > > ; --------------------------------------------------------------------------------- > ; 1.3 Codecs setup > ; > disallow = all > allow = alaw > > ; --------------------------------------------------------------------------------- > ; 1.4 Other options > ; > context = default > defaultexpirey = 160 > dtmfmode = info > maxexpirey = 180 > nat = never > qualify = no > record-in = On-Demand > record-out = On-Demand > type = friend > > ; --------------------------------------------------------------------------------- > ; 2 Devices for their respective contexts > ; > [spaphone] > accountcode = spaphone > callerid = spaphone > canreinvite = yes > context = pstn > dtmfmode = info > host = dynamic > mailbox > port = 5060 > qualify = yes > secret = abcde > type = friend > username = spaphone2. Asterisk version: ============> Asterisk 1.4.20-1 RPM by vc-rpms at voipconsulting.nl, Copyright (C) 1999 - 2008 Digium, Inc. and others. > Created by Mark Spencer <markster at digium.com> > Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. > This is free software, with components licensed under the GNU General Public > License version 2 and other licenses; you are welcome to redistribute it under > certain conditions. Type 'core show license' for details. > ========================================================================> == Parsing '/etc/asterisk/asterisk.conf': Parsing /etc/asterisk/asterisk.conf > Found > Connected to Asterisk 1.4.20-1 RPM by vc-rpms at voipconsulting.nl currently running on asterisk2 (pid = 2336) > Verbosity is at least 5 > Core debug is at least 13. spa-3102 details: ============> Product Name: SPA-3102 > Software Version: 5.1.10(GW) > Hardware Version: 1.4.5(a) > LAN IP address: 192.168.0.10 > LAN subnet mask: 255.255.255.0 > > PSTN Line -> SIP settings > SIP Transport: UDP > SIP Port: 5060 > PSTN Line -> Proxy and Registration > Proxy: 192.168.0.1 > PSTN Line -> Subscriber information > Display name: spaphone > User ID: spaphone > Password: abcde4. SIP debug output on asterisk console: ============> REGISTER sip:192.168.0.1 SIP/2.0> Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-f05c05f9 > From: spaphone <sip:spaphone at 192.168.0.1>;tag=c2f457c6347c4f23o1 > To: spaphone <sip:spaphone at 192.168.0.1> > Call-ID: f264209-bccc3039 at 127.0.0.1 > CSeq: 38885 REGISTER > Max-Forwards: 70 > Contact: spaphone <sip:spaphone at 127.0.0.1:5060>;expires=3600 > User-Agent: Linksys/SPA3102-5.1.10(GW) > Content-Length: 0 > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER > Supported: x-sipura, replaces > > > <-------------> > --- (12 headers 0 lines) --- > Using latest REGISTER request as basis request > Sending to 127.0.0.1 : 5060 (no NAT) > asterisk2*CLI> > <--- Transmitting (no NAT) to 127.0.0.1:5060 ---> > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-f05c05f9;received=192.168.0.10 > From: spaphone <sip:spaphone at 192.168.0.1>;tag=c2f457c6347c4f23o1 > To: spaphone <sip:spaphone at 192.168.0.1> > Call-ID: f264209-bccc3039 at 127.0.0.1 > CSeq: 38885 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:spaphone at 192.168.0.1> > Content-Length: 0 > > > <------------> > asterisk2*CLI> > <--- Transmitting (no NAT) to 127.0.0.1:5060 ---> > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-f05c05f9;received=192.168.0.10 > From: spaphone <sip:spaphone at 192.168.0.1>;tag=c2f457c6347c4f23o1 > To: spaphone <sip:spaphone at 192.168.0.1>;tag=as6140f831 > Call-ID: f264209-bccc3039 at 127.0.0.1 > CSeq: 38885 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1aa1d724" > Content-Length: 0 > > > <------------> > Scheduling destruction of SIP dialog 'f264209-bccc3039 at 127.0.0.1' in 32000 ms (Method: REGISTER) > asterisk2*CLI> > <--- SIP read from 192.168.0.10:5060 ---> > REGISTER sip:192.168.0.1 SIP/2.0 > Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-f05c05f9 > From: spaphone <sip:spaphone at 192.168.0.1>;tag=c2f457c6347c4f23o1 > To: spaphone <sip:spaphone at 192.168.0.1> > Call-ID: f264209-bccc3039 at 127.0.0.1 > CSeq: 38885 REGISTER > Max-Forwards: 70 > Contact: spaphone <sip:spaphone at 127.0.0.1:5060>;expires=3600 > User-Agent: Linksys/SPA3102-5.1.10(GW) > Content-Length: 0 > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER > Supported: x-sipura, replaces > > > <-------------> > --- (12 headers 0 lines) --- > Using latest REGISTER request as basis request > Sending to 127.0.0.1 : 5060 (no NAT) > asterisk2*CLI> > <--- Transmitting (no NAT) to 127.0.0.1:5060 ---> > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-f05c05f9;received=192.168.0.10 > From: spaphone <sip:spaphone at 192.168.0.1>;tag=c2f457c6347c4f23o1 > To: spaphone <sip:spaphone at 192.168.0.1> > Call-ID: f264209-bccc3039 at 127.0.0.1 > CSeq: 38885 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:spaphone at 192.168.0.1> > Content-Length: 0 > > > <------------> > asterisk2*CLI> > <--- Transmitting (no NAT) to 127.0.0.1:5060 ---> > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-f05c05f9;received=192.168.0.10 > From: spaphone <sip:spaphone at 192.168.0.1>;tag=c2f457c6347c4f23o1 > To: spaphone <sip:spaphone at 192.168.0.1>;tag=as6140f831 > Call-ID: f264209-bccc3039 at 127.0.0.1 > CSeq: 38885 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1aa1d724" > Content-Length: 0 > > > <------------> > Scheduling destruction of SIP dialog 'f264209-bccc3039 at 127.0.0.1' in 32000 ms (Method: REGISTER) > asterisk2*CLI> exit > <--- SIP read from 192.168.0.10:5060 ---> > REGISTER sip:192.168.0.1 SIP/2.0 > Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-f05c05f9 > From: spaphone <sip:spaphone at 192.168.0.1>;tag=c2f457c6347c4f23o1 > To: spaphone <sip:spaphone at 192.168.0.1> > Call-ID: f264209-bccc3039 at 127.0.0.1 > CSeq: 38885 REGISTER > Max-Forwards: 70 > Contact: spaphone <sip:spaphone at 127.0.0.1:5060>;expires=3600 > User-Agent: Linksys/SPA3102-5.1.10(GW) > Content-Length: 0 > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER > Supported: x-sipura, replaces > > > <-------------> > --- (12 headers 0 lines) --- > Using latest REGISTER request as basis request > Sending to 127.0.0.1 : 5060 (no NAT) > asterisk2*CLI> exit > <--- Transmitting (no NAT) to 127.0.0.1:5060 ---> > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-f05c05f9;received=192.168.0.10 > From: spaphone <sip:spaphone at 192.168.0.1>;tag=c2f457c6347c4f23o1 > To: spaphone <sip:spaphone at 192.168.0.1> > Call-ID: f264209-bccc3039 at 127.0.0.1 > CSeq: 38885 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:spaphone at 192.168.0.1> > Content-Length: 0 > > > <------------> > asterisk2*CLI> exit > <--- Transmitting (no NAT) to 127.0.0.1:5060 ---> > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-f05c05f9;received=192.168.0.10 > From: spaphone <sip:spaphone at 192.168.0.1>;tag=c2f457c6347c4f23o1 > To: spaphone <sip:spaphone at 192.168.0.1>;tag=as6140f831 > Call-ID: f264209-bccc3039 at 127.0.0.1 > CSeq: 38885 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1aa1d724" > Content-Length: 0 > > > <------------> > Scheduling destruction of SIP dialog 'f264209-bccc3039 at 127.0.0.1' in 32000 ms (Method: REGISTER) > asterisk2*CLI> sip no debug > SIP Debugging Disabled > asterisk2*CLI>
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