Summary:	ASTERISK-07597: asterisk does not recognize correct peer trunk
Reporter:	Stefano Benetti (benez)	Labels:
Date Opened:	2006-08-24 11:37:59	Date Closed:	2006-08-24 12:50:21
Priority:	Major	Regression?	No
Status:	Closed/Complete	Components:	Channels/chan_sip/Registration
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:
Description:	Asterisk SVN-branch-1.2-r40901 I have found this bug: If you register asterisk with more than one trunk of the same provider it does not recognize the correct peer from witch the call is coming. I activated the sip debug function of the peer to be sure that it was not an error of the provider and I seen that every call from one of the trunk had debugged. Asterisk seem to check only the IP address of the provider. This is a big problem because * can not select the correct trunk into the dialplan.   ****** ADDITIONAL INFORMATION ****** my trunks Host                            Username       Refresh State voip.eutelia.it:5060            0302053900         585 Registered voip.eutelia.it:5060            0302056966         585 Registered I have activated the debug option on the second trunk, but wen I call the first one the debug is active also for it and asterisk select the wrong dialplane. Capabilities: us - 0x100 (g729), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Looking for s in 002_incoming (domain 81.174.26.150) list_route: hop: <sip:195.62.225.244;ftag=5946A06C-192C;lr=on> Transmitting (NAT) to 195.62.225.244:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 195.62.225.244;branch=z9hG4bK4081.cd11f78.0;received=195.62.225.244 Via: SIP/2.0/UDP  83.211.172.173:5060;branch=z9hG4bK913561AA4 From: <sip:3332692708@83.211.172.173>;tag=5946A06C-192C To: <sip:0302056966@voip.eutelia.it> Call-ID: 17B53641-32C511DB-BB609FA9-A8D359A8@83.211.172.173 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:s@81.174.26.150> Content-Length: 0
Comments:	By: Serge Vecher (serge-v) 2006-08-24 12:50:09 benez: this is a known bug in Asterisk. Please visit bug ASTERISK-5683 for the latest status and developer's branch addressing this issue. Thanks.