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Summary:ASTERISK-01863: Outgoing SIP Proxy authentication failes due to missing credentials
Reporter:kurti (kurti)Labels:
Date Opened:2004-06-22 02:14:50Date Closed:2011-06-07 14:00:50
Priority:BlockerRegression?No
Status:Closed/CompleteComponents:Core/General
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Description:Registering on sip-proxy works fine, but dialout does not work. I added

       ast_log(LOG_WARNING, "p->authname: %s p->realm: %s p->peersecret: %s\n", p->authname,p->realm,p->peersecret);

to "build_reply_digest" to see what is used.

This is the registration which works fine:
Jun 22 08:49:50 WARNING[1101798320]: chan_sip.c:5822 build_reply_digest: p->authname: 533XXXX p->realm: sipgate.de p->peersecret: XXXXXX

This is the dial which failes:
   -- Executing Dial("SIP/533XXXX-e458", "SIP/0XXXXXXXX@sipgate.de|120") in new stack
   -- Called 0XXXXXXXX@sipgate.de
Jun 22 08:49:57 WARNING[1101798320]: chan_sip.c:5822 build_reply_digest: p->authname:  p->realm: 81.48.73.83 p->peersecret:
Jun 22 08:49:57 WARNING[1101798320]: chan_sip.c:5822 build_reply_digest: p->authname:  p->realm: 81.48.73.83 p->peersecret:
Jun 22 08:49:57 NOTICE[1101798320]: chan_sip.c:6234 handle_response: Failed to authenticate on INVITE to '"Kurt Huwig" <sip:533XXXX@81.48.73.83>;tag=as333aXXXX'
 == No one is available to answer at this time

I did a tcpdump on the line and I got this:

Proxy-Authorization: Digest username="", realm="81.48.73.83", algorithm="MD5", uri="sip:...."

the rest looks the same as with direct dialing.

I even modified the code to 'strcpy' the right username and password into struct sip_pvt just before creating the hash, but it does not work either.

Calls to toll-free phone numbers work fine, but I guess this is due to no needed authentication.
Comments:By: kurti (kurti) 2004-06-22 03:32:19

Ok, I forgot something:

1. Version is CVS from 21st June 20:00
2. Debian Testing
By: kurti (kurti) 2004-06-22 05:39:48

/me stupid!

Please close this bug. I used

 exten => _9.,1,Dial(SIP/${EXTEN:1}@sipgate.de,120)

instead of

 exten => _9.,1,Dial(SIP/${EXTEN:1}@sipgate-out,120)

Now I understand 'Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]'...