Juraj Varchol
2005-Nov-16 11:51 UTC
[Asterisk-Users] SER authenitification failure on ASTERISK
Hey guys, I'm quite new in this SER/Asterisk environment and I'm having some problems with authentification. Now i'm running SER and Asterisk on 2 boxes. It is working according to the scheme SIP user <------> SER Server <--------> Asterisk <----->IAX users/PSTN/... SIP calls are managed by SER Server and the rest is forwarded to Asterisk for further processing. Also acording to dialing plan on Asterisk, calls to SIP users are fwd to SER Server where are redirected to their actual location. Outgoing calls to SER Server are set up: **extensions.conf** exten => _77.,1,Dial(SIP/${EXTEN:2}@sip_proxy-out,10) exten => _77.,2,Hangup **sip.conf** - where only for outgoing calls is set up context [sip_proxy-out] type=peer username= asterisk_user-name_on_SER_Server secret=asterisk_passwd_on_SER_Server allow=all host=SER_Server_IP or URL ----> this way it does work even sometimes UA on SER shows as incoming call from "asterisk_user-name_on_SER_Server" instead of callerID from Asterisk ... but that would not be such a problem ... also is important to mention that "asterisk_user-name_on_SER_Server" is registred on SER Server. (* if there is better way how to deal with it i would b very glad to at least get pointed the right direction ... ;)) !!!!THE PROBLEM comes the other way when SER users want to call extension that is managed by ASTERISK. I get error message in both cases that r described at the end. I'm looking for some solution that would not include registering all SER users on Asterisk or just omit authentification for all users coming from SER_Server Thanx in advance, Juraj **error message** Nov 16 19:13:53 NOTICE[9339]: chan_sip.c:7355 handle_request: Failed to authenticate user <sip:SER_USER@SER_SERVER_IP:5060> ;tag=93af6721-13c4-437b76e2-5f0d59d-2fee *****both options i tried but no luck************** my setup on SER Server is **ser.cfg** .... if (uri=~"^sip:[4-9]*@(IP_SER_SERVER|(proxy\.)?URL_SER_SERVER)") { rewritehostport("IP_ASTERISK_SERVER:5060"); forward(uri:host, uri:port); break; }; ... **extensions.cong** --- i tried to redirect incoming call right away to normal dialing plan... [inpbx] ... exten => 456,2,Dial(IAX2/User_1,15) exten => 456,3,Voicemail(u44@test) ... exten => 456,6,Hangup ... exten => 567,2,Dial(IAX2/User_2,15) .... exten => 567,6,Hangup ... OR modification 2 ... when we want to reach Asterisk Server must by dialed prefix (for example 44_) + extensions by SIP user. **extension.conf** [default] exten => _44.,1,Dial(SIP/${EXTEN:2}@sip_proxy,15,rm) ;here i strip it from the extension exten => _44.,2,Hangup **sip.conf** [sip_proxy] type=friend context=inpbx ;insecure=very host=URL_SER_SERVER ;mailbox=user@test -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051116/05968c0e/attachment.htm
Rafael R. GV
2005-Nov-16 12:10 UTC
[Asterisk-Users] SER authenitification failure on ASTERISK
try insecure=very (needed if we want to allow incoming SER calls to bypass authentication) and type=friend in [sip_proxy] rafael On 11/16/05, Juraj Varchol <varchol@gmail.com> wrote:> > Hey guys, > > I'm quite new in this SER/Asterisk environment and I'm having some > > problems with authentification. Now i'm running SER and Asterisk on 2 > > boxes. It is working according to the scheme > > SIP user <------> SER Server <--------> Asterisk <----->IAX > users/PSTN/... > > SIP calls are managed by SER Server and the rest is forwarded to > > Asterisk for further processing. Also acording to dialing plan on > > Asterisk, calls to SIP users are fwd to SER Server where are > > redirected to their actual location. > > Outgoing calls to SER Server are set up: > > **extensions.conf** > > exten => _77.,1,Dial(SIP/${EXTEN:2}@sip_proxy-out,10) > > exten => _77.,2,Hangup > > **sip.conf** - where only for outgoing calls is set up context > > [sip_proxy-out] > > type=peer > > username= asterisk_user-name_on_SER_Server > > secret=asterisk_passwd_on_SER_Server > > allow=all > > host=SER_Server_IP or URL > > ----> this way it does work even sometimes UA on SER shows as incoming > > call from "asterisk_user-name_on_SER_Server" instead of callerID from > > Asterisk ... but that would not be such a problem > > ... also is important to mention that > > "asterisk_user-name_on_SER_Server" is registred on SER Server. (* if > > there is better way how to deal with it i would b very glad to at > > least get pointed the right direction ... ;)) > > !!!!THE PROBLEM comes the other way when SER users want to call > > extension that is managed by ASTERISK. > > I get error message in both cases that r described at the end. I'm > > looking for some solution that would not include registering all SER > > users on Asterisk or just omit authentification for all users coming > > from SER_Server > > Thanx in advance, > > Juraj > > **error message** > > Nov 16 19:13:53 NOTICE[9339]: chan_sip.c:7355 handle_request: Failed > > to authenticate user <sip:SER_USER@SER_SERVER_IP:5060> > > ;tag=93af6721-13c4-437b76e2-5f0d59d-2fee > > *****both options i tried but no luck ************** > > my setup on SER Server is > > **ser.cfg** > > .... > > if (uri=~"^sip:[4-9]*@(IP_SER_SERVER|(proxy\.)?URL_SER_SERVER)") { > > rewritehostport("IP_ASTERISK_SERVER:5060"); > > forward(uri:host, uri:port); > > break; > > }; > > ... > > **extensions.cong** --- i tried to redirect incoming call right away > > to normal dialing plan... > > [inpbx] > > ... > > exten => 456,2,Dial(IAX2/User_1,15) > > exten => 456,3,Voicemail(u44@test) > > ... > > exten => 456,6,Hangup > > ... > > exten => 567,2,Dial(IAX2/User_2,15) > > .... > > exten => 567,6,Hangup > > ... > > OR modification 2 ... when we want to reach Asterisk Server must by > > dialed prefix (for example 44_) + extensions by SIP user. > > **extension.conf** > > [default] > > exten => _44.,1,Dial(SIP/${EXTEN:2}@sip_proxy,15,rm) ;here i strip > > it from the extension > > exten => _44.,2,Hangup > > **sip.conf** > > [sip_proxy] > > type=friend > > context=inpbx > > ;insecure=very > > host=URL_SER_SERVER > ;mailbox=user@test > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com <http://Easynews.com>-- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >-- rrgv -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051116/bec7c1fb/attachment.htm