Just want to ask for help here. I registered a alpha-numeric account on sip.conf and works fine in and out but when I tried to set up a numeric account there seems to be a problem.
– Executing [18773656070@vvu_api:3] Dial(“SIP/8888888888-09c07a00”, “SIP/18773656070@66.251.216.50|60”) in new stack
– Called 18773656070@66.251.216.50
[Jan 24 17:40:25] WARNING[478]: chan_sip.c:8403 check_auth: username mismatch, have <8888888888>, digest has <>
[Jan 24 17:40:25] NOTICE[478]: chan_sip.c:13828 handle_request_invite: Failed to authenticate user “8888888888” sip:8888888888@66.251.216.50;tag=as7ac95b37
[Jan 24 17:40:26] WARNING[478]: chan_sip.c:8403 check_auth: username mismatch, have <8888888888>, digest has <>
[Jan 24 17:40:26] NOTICE[478]: chan_sip.c:13828 handle_request_invite: Failed to authenticate user “8888888888” sip:8888888888@66.251.216.50;tag=as63ac9919
Now I got busy or can send out call to my gateway.
Here is the scenario:
Scenario A:
sip account name is testaccount
testaccount -> dial 18773656070 -> SIP/18773656070@66.251.216.50 works fine
but this one not
sip account name is 8888888888
8888888888 > dial 18773656070 -> SIP/18773656070@66.251.216.50
does not work and sends authentication mismatch
I think I have the same problem; I am new to Asterisk, so this was baffling, but I don’t think it is my fault. I see nobody has responded to rowell yet.
I have a new setup of AsteriskNOW Beta 6. I am trying to create the dialplan for Linksys SPA942 phones on 12 stations. So far I cannot get my first station working right. (I wonder if I would have the same problem with each other phone?) Sometimes I can send and receive calls, but other times I have problems. The Asterisk console gives a similar error message. My extension is set as 504, my username is 504, my password is 504. The console says username mismatch, “digest has <540>”
I don’t know what the digest is, but I am sure that I never registered the number 540 during AsteriskNOW setup. Is there a way to change the digest manually?
I am not pretty sure where this problem is originating. I never had encountered this problem on earlier versions of asterisk 1.4.x until I upgraded to the latest version.
If somebody could here could enlighten us how we are going to fix this issue, that would be nice. But based on some other forums that I have read already there seems to be no clear instruction on how to deal with this issue.
But to further clear out the problem I’ve been encountering. This just actually happens when I sends call going to our sip gateway (outgoing). But if it’s just like 101 calling 102 there would be no issue.
