Try to add a qualify=XXXX to sip.conf, and try to exec a sip show peers. In spite of phones appears like register, if you use NAT, your firewall can cut communication. Try the next:
Just after phone register call to it, and then wait for a minutes and try to call again. Could you call first time but not in second one? It is due to your firewall. Try to configure wuth next config: [1004] ...... ..... qualify=XXXX ....... ...... In you grandstream configuration try to put time to expire register 1 minute and then try to do the previous test. I'm sorry for my english, but I hope this let you call. Regards, srsergio -----Mensaje original----- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de pesb Enviado el: lunes, 29 de marzo de 2004 20:26 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] Asterisk + GrandStream SIP phones -This is my 'sip.conf' file: ;************************************************************* ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls tos=184 maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=120 ; Default length of incoming/outoing registration disallow=all ; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=alaw [1004] type=friend username=1004 secret= reinvite=no canreinvite=no host=dynamic dtmfmode=inband mailbox=1004 nat=1 disallow=all allow=ulaw allow=alaw [1005] type=friend username=1005 secret= reinvite=no canreinvite=no host=dynamic dtmfmode=inband mailbox=1005 nat=1 disallow=all allow=ulaw allow=alaw ;******************************************* -And this is the basic seting of my two GrandStream SIP phones: ***************[1005]**************** IP Address:192.168.0.105 Subnet Mask:255.255.255.0 SIP Server: 192.168.0.103 Outbound Proxy:<empty> SIP User ID:1005 Authenticate ID:1005 Authenticate Password:123 Name:1005 Preferred Vocoder: choice 1: PCMU choice 2: PCMA choice 3: G723 choice 4: G729 choice 5: G726-32 choice 6: G728 G723 rate: 6.3kbps Silence Suppression:No Send DTMF:in-audio ***************[1004]**************** IP Address:192.168.0.104 Subnet Mask:255.255.255.0 SIP Server: 192.168.0.103 Outbound Proxy:<empty> SIP User ID:1004 Authenticate ID:1004 Authenticate Password:123 Name:1004 Preferred Vocoder: choice 1: PCMU choice 2: PCMA choice 3: G723 choice 4: G729 choice 5: G726-32 choice 6: G728 G723 rate: 6.3kbps Silence Suppression:No Send DTMF:in-audio ****************************** I have 2 SIP GrandStream phones, both phones are correctly registered to the Asterisk server. But, when I try to make a call from registered phone '1005' to registered phone '1004', dialing 1004, Asterisk responds with the 'Status: 404 Not Found' message. How do I have to dial? What else do I need to set? Find attached my traffic captured on ethereal. _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
