Jonas Kellens
2010-Oct-07 11:54 UTC
[asterisk-users] 401 Unauthorized with Snom but not with Zoiper softphone
Hello, I'm having difficulty with registering a SIP account in a Snom 320 IP-phone. This is what sip debug tells me : [Oct 7 13:28:42] VERBOSE[20314] chan_sip.c: [Oct 7 13:28:42] <--- SIP read from UDP:public_ip:58697 ---> REGISTER sip:sip.domain.tld SIP/2.0 Via: SIP/2.0/UDP 192.168.114.200:2048;branch=z9hG4bK-vj1xvbdnp4dw;rport From: <sip:test3 at sip.domain.tld>;tag=sd2b3o74zc To: <sip:test3 at sip.domain.tld> Call-ID: 3c28a76e73cf-gp9nioi8zdci CSeq: 12 REGISTER Max-Forwards: 70 Contact: <sip:test3 at 192.168.114.200:2048;line=nagqjxpx>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:8d4403c1-af97-4139-b535-de7d460a9187>";audio;mobility="fix ed";duplex="full";description="snom320";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom320/8.4.18 Allow-Events: dialog X-Real-IP: 192.168.114.200 Supported: path, gruu Expires: 3600 Content-Length: 0 <-------------> [Oct 7 13:28:42] VERBOSE[20314] chan_sip.c: [Oct 7 13:28:42] --- (14 headers 0 lines) --- [Oct 7 13:28:42] VERBOSE[20314] chan_sip.c: [Oct 7 13:28:42] Sending to 192.168.114.200 : 2048 (no NAT) [Oct 7 13:28:42] VERBOSE[20314] chan_sip.c: [Oct 7 13:28:42] <--- Transmitting (NAT) to public_ip:58697 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.114.200:2048;branch=z9hG4bK-vj1xvbdnp4dw;received=public_ip;rport=58697 From: <sip:test3 at sip.domain.tld>;tag=sd2b3o74zc To: <sip:test3 at sip.domain.tld>;tag=as6108a7e2 Call-ID: 3c28a76e73cf-gp9nioi8zdci CSeq: 12 REGISTER Server: Asterisk PBX 1.6.2.10 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="domain.tld", nonce="398aee1e" Content-Length: 0 I would expect the Snom to try a second register, this time with some type of nonce. But there is just 1 REGISTER and 1 Unauthorized and that's it... Other Snom phones with SIP-accounts go very well, but at this location the registration fails. Another remark : when using a Zoiper softphone, the registration goes very well : REGISTER sip:sip.domain.tld;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.114.20:5060;branch=z9hG4bK-d8754z-fab4a5effbf90a05-1---d8754z- Max-Forwards: 70 Contact: <sip:test3 at public_ip:51363;rinstance=b6fd38105c91b9bf;transport=UDP> To: <sip:test3 at sip.domain.tld;transport=UDP> From: <sip:test3 at sip.domain.tld;transport=UDP>;tag=db1a5018 Call-ID: NzBlZDMyN2U0YTEzZDk4Y2M2N2NmNzMxYTk4OWUxYTY. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Zoiper rev.7797 Allow-Events: presence, kpml Content-Length: 0 <-------------> [Oct 7 13:46:52] VERBOSE[20314] chan_sip.c: [Oct 7 13:46:52] --- (13 headers 0 lines) --- [Oct 7 13:46:52] VERBOSE[20314] chan_sip.c: [Oct 7 13:46:52] Sending to 192.168.114.20 : 5060 (no NAT) [Oct 7 13:46:52] VERBOSE[20314] chan_sip.c: [Oct 7 13:46:52] <--- Transmitting (NAT) to public_ip:51363 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.114.20:5060;branch=z9hG4bK-d8754z-fab4a5effbf90a05-1---d8754z-;received=public_ip From: <sip:test3 at sip.domain.tld;transport=UDP>;tag=db1a5018 To: <sip:test3 at sip.domain.tld;transport=UDP>;tag=as2fcfde3c Call-ID: NzBlZDMyN2U0YTEzZDk4Y2M2N2NmNzMxYTk4OWUxYTY. CSeq: 1 REGISTER Server: Asterisk PBX 1.6.2.10 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="domain.tld", nonce="7833b268" Content-Length: 0 REGISTER sip:sip.domain.tld;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.114.20:5060;branch=z9hG4bK-d8754z-fdd59e394f9c23b9-1---d8754z- Max-Forwards: 70 Contact: <sip:test3 at public_ip:51363;rinstance=b6fd38105c91b9bf;transport=UDP> To: <sip:test3 at sip.domain.tld;transport=UDP> From: <sip:test3 at sip.domain.tld;transport=UDP>;tag=db1a5018 Call-ID: NzBlZDMyN2U0YTEzZDk4Y2M2N2NmNzMxYTk4OWUxYTY. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Zoiper rev.7797 Authorization: Digest username="test3",realm="domain.tld",nonce="7833b268",uri="sip:sip.domain.tld;transport=UDP",response="198f6262248fb11fe6cb55408a1cb8ce",algorithm=MD5 Allow-Events: presence, kpml Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.114.20:5060;branch=z9hG4bK-d8754z-fdd59e394f9c23b9-1---d8754z-;received=public_ip From: <sip:test3 at sip.domain.tld;transport=UDP>;tag=db1a5018 To: <sip:test3 at sip.domain.tld;transport=UDP>;tag=as2fcfde3c Call-ID: NzBlZDMyN2U0YTEzZDk4Y2M2N2NmNzMxYTk4OWUxYTY. CSeq: 2 REGISTER Server: Asterisk PBX 1.6.2.10 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 60 Contact: <sip:test3 at public_ip:51363;rinstance=b6fd38105c91b9bf;transport=UDP>;expires=60 Date: Thu, 07 Oct 2010 11:46:52 GMT Content-Length: 0 It's the same account, the same password, but other agent. Can anyone help me with this please ?! I see no difference but there must be !! Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101007/72e35aff/attachment-0001.htm
Daniel Tryba
2010-Oct-07 12:24 UTC
[asterisk-users] 401 Unauthorized with Snom but not with Zoiper softphone
On Thu, Oct 07, 2010 at 01:54:58PM +0200, Jonas Kellens wrote:> It's the same account, the same password, but other agent. > > Can anyone help me with this please ?! I see no difference but there > must be !!The difference is the SNOM is using rport and Zoiper isn't. Is nat for this client set to 'yes' or something else? -- Daniel Tryba
Philipp von Klitzing
2010-Oct-07 14:18 UTC
[asterisk-users] 401 Unauthorized with Snom but not with Zoiper softphone
Hi!> I'm having difficulty with registering a SIP account in a Snom 320 IP- > phone.Do a SIP trace on your SNOM phone, and you will most probably see that the 401 reply of Asterisk does not arrive on the phone. Then check your STUN/ICE settings on the phone in combination with the nat= settings in sip.conf on Asterisk. BTW: Are you happy with firmware 8.4.18? I still stick to 7.3.30 for the 3xx models. Philipp