search for: ast_sip_ouraddrfor

I have been trying out several stun servers with Asterisk 1.6.0.9 and 1.6.1.0 and I keep getting the following message: [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]: chan_s...

Hi, I have the following problem. I have 5 phones behind the same nat (canreinvite=yes). it works fine to receive calls and to make calls. sound quality is good, so everything works fine. The poblem is that the phone behind nat cant call each other. It works if canreinvite=no. But i want to do this. Does anyone have an idea? Regards, cjk.

...sing SIP RTP CoS mark 5 [Mar 19 18:22:56] DEBUG[17418]: chan_sip.c:4683 do_setnat: Setting NAT on RTP to Off [Mar 19 18:22:57] DEBUG[17418]: acl.c:715 ast_ouraddrfor: For destination '192.168.1.6', our source address is '192.168.1.25'. [Mar 19 18:22:57] DEBUG[17418]: chan_sip.c:3243 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.1.25:5060 [Mar 19 18:22:57] DEBUG[17418]: chan_sip.c:6557 sip_new: *** Our native formats are 0x100 (g729) [Mar 19 18:22:57] DEBUG[17418]: chan_sip.c:6558 sip_new: *** Joint capabilities are 0x100 (g729) [Mar 19 18:22:57] DEBUG[17418]: chan_sip.c:6559...

..., but releases it */ struct sip_pvt *p; int newmsgs, oldmsgs; + char *s; /* Check for messages */ ast_app_messagecount(peer->mailbox, &newmsgs, &oldmsgs); @@ -9735,6 +9736,10 @@ /* Recalculate our side, and recalculate Call ID */ if (ast_sip_ouraddrfor(&p->sa.sin_addr,&p->ourip)) memcpy(&p->ourip, &__ourip, sizeof(p->ourip)); + strcpy(p -> username, peer -> mailbox); /* Username = Mailbox name */ + s = strchr(p -> username, '@'); /* Remove the context part */ +...

Dear list, I'm trying to get Asterisk to work dual-stack on Linux and I'm left with a question. Imagine that a user (on the road) connects to Asterisk from various places. Many of them probably don't have IPv6 support yet. However, his house and office do have IPv6 connectivity. I would like to make sure that whenever IPv6 is available, the connection will be made over IPv6, but

...ooking for Call ID: hDVA1Kyx-1327766611250 at lucidesktop.lan (Checking From) --From tag grUqFtoE --To-tag [Jan 28 23:03:32] DEBUG[1654]: acl.c:728 ast_ouraddrfor: For destination '192.168.2.159', our source address is '192.168.2.172'. [Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:3482 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.2.172:5060 [Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:7694 sip_alloc: Allocating new SIP dialog for hDVA1Kyx-1327766611250 at lucidesktop.lan - INVITE (No RTP) [Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:24907 handle_incoming: **** Received INVITE (5) -...