Summary: | ASTERISK-00022: Registration-requests originated from chan_sip don't use the source-IP specified as "bindaddr". | ||
Reporter: | oliver (oliver) | Labels: | |
Date Opened: | 2003-07-28 08:00:24 | Date Closed: | 2008-01-15 14:31:57.000-0600 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | Core/General |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
Description: | I'm experiencing a problem with chan_sip on a multi-homed machine. The machine has 1 interface to the rest of the world and 1 interface on a local network. The local network has public IP-addresses, though, and the IP-addresses of both interfaces are reachable from the outside world, but by default, outgoing traffic from that machine to the outside world will have the IP-address of the interface the default-route points to as it's source, which is the one with the outside world behind it, obviously. I've set the bind-address in "sip.conf" to the IP-address of the interface on the local network, because I want to force it to only use that interface's IP-address. This works great for binding only to that IP-address and I can even make outgoing calls, but when registering to a remote SIP-provider, chan_sip seems to use the IP-address of the wrong interface (the one the default-route points to) as the source in it's registration-requests, so as soon as a call comes in from that SIP-provider, it's sent to that IP-address and fails, as chan_sip isn't listening on it. If I remove the "bindaddr" to make chan_sip listen on all IP-addresses of the machine, calls come in as expected. BTW: The remote SIP-provider is FWD in this case. ****** ADDITIONAL INFORMATION ****** Asterisk CVS-07/27/03-13:44:14. Linux 2.4.21, RH9 base-system. More info available on request. | ||
Comments: | By: oliver (oliver) 2003-07-28 08:17:36 More info: It seems to be using the wrong IP-address in the "Via:" and "Contact:"-headers during registration. I suggest changing it to use the "bindaddr" there, if specified. edited on: 07-28-03 08:06 By: Mark Spencer (markster) 2003-08-14 23:31:12 Done, please confirm By: oliver (oliver) 2003-08-15 06:40:12 Hey Mark, Seems to be OK now. Thnx! You can close this ticket. By: Digium Subversion (svnbot) 2008-01-15 14:31:57.000-0600 Repository: asterisk Revision: 1338 U trunk/apps/app_agi.c U trunk/apps/app_enumlookup.c U trunk/apps/app_festival.c U trunk/apps/app_queue.c U trunk/apps/app_voicemail2.c U trunk/channels/chan_sip.c U trunk/channels/chan_zap.c U trunk/codecs/ilbc/FrameClassify.c U trunk/codecs/ilbc/LPCdecode.c U trunk/codecs/ilbc/LPCencode.c U trunk/codecs/ilbc/StateConstructW.c U trunk/codecs/ilbc/StateSearchW.c U trunk/codecs/ilbc/anaFilter.c U trunk/codecs/ilbc/createCB.c U trunk/codecs/ilbc/doCPLC.c U trunk/codecs/ilbc/enhancer.c U trunk/codecs/ilbc/filter.c U trunk/codecs/ilbc/gainquant.c U trunk/codecs/ilbc/getCBvec.c U trunk/codecs/ilbc/helpfun.c U trunk/codecs/ilbc/hpInput.c U trunk/codecs/ilbc/hpOutput.c U trunk/codecs/ilbc/iCBConstruct.c U trunk/codecs/ilbc/iCBSearch.c U trunk/codecs/ilbc/iLBC_decode.c U trunk/codecs/ilbc/iLBC_encode.c U trunk/codecs/ilbc/lsf.c U trunk/codecs/ilbc/packing.c U trunk/codecs/ilbc/syntFilter.c U trunk/pbx/pbx_gtkconsole.c ------------------------------------------------------------------------ r1338 | markster | 2008-01-15 14:31:57 -0600 (Tue, 15 Jan 2008) | 8 lines Asterisk: ASTERISK-68 - Should eliminate probs on VMWI ASTERISK-37 - Dynamic add survives reload ASTERISK-69 - Make festival honor its arguments ASTERISK-89 - Make events on FXO interfaces more logical ASTERISK-22 - Prefer "bindaddr" to logical address for registrations ??? - Record crashes AGI ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=1338 |