search for: sip_transport_udp

...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 sip_new: *** Our capabilit...

...To-tag [Feb 12 15:13:59] DEBUG[25824] acl.c: For destination '83.186.238.111', our source address is '172.26.19.13'. [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Target address 83.186.238.111:5060 is not local, substituting externaddr [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 212.105.99.108:5060 [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Allocating new SIP dialog for 857d4ed8 at 83.186.238.111<mailto:857d4ed8 at 83.186.238.111> - MESSAGE (No RTP) [Feb 12 15:13:59] VERBOSE[25824] chan_sip.c: No matching peer for '83.186.238.111' from '83.18...

Hi list, My need is to append a site specific parameter to the Contact: header on all INVITEs exiting * via a SIP trunk. I'd like it to look something like this: Contact: <bob:3125551212 at 10.10.10.10;SITE-ID=us.here> where SITE-ID=us.here is set in a config file that * parses on startup. Or in a Dial() command option? Or I don't care exactly how. :-) It is possible to

...x-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) - Command in SIP INVITE [Jan...

...63 times>}, data = "REGISTER\000sip:10.3.8.1\000SIP/2.0\000\000Route: <sip:10.3.8.1:5060;lr>\000\000CSeq: 2 REGISTER\000\000Via: SIP/2.0/UDP 10.3.8.104:5060;branch=z9hG4bK5a11a1de-24d8-de11-8af3-00248cdec25c;rport\000\000User-Agent: Ekiga/3.2.0\000\000From"..., socket = {type = SIP_TRANSPORT_UDP, fd = -1, port = 50195, tcptls_session = 0x0}, next = {next = 0x0}} sin = {sin_len = 16 '\020', sin_family = 2 '\002', sin_port = 50195, sin_addr = {s_addr = 1745355530}, sin_zero = "\000\000\000\000\000\000\000"} res = 526 len = 16 __PRETTY...

I'm attempting to find where my extra long DTMF Tones are coming from. I'm dialing from my sip handset through my proxy to my Asterisk box which is my PSTN Gateway. I'm pressing 4 to select a menu and everything is fine. [Feb 12 16:58:18] DTMF[29762] channel.c: DTMF begin '4' received on SIP/trunk-0a02dee0 [Feb 12 16:58:18] DTMF[29762] channel.c: DTMF begin passthrough