Displaying 7 results from an estimated 7 matches for "ourip".
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yourip
2005 Sep 03
0
MWI - message waiting indication
...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 */
+ if (s != NULL)
+ *s =...
2003 Jun 10
1
SIP sdp o= and c= fields
Hello,
If I understand it correctly, when sending INVITE, o= and c= sdp fields are
built using p->ourip
IP address. At this point RTP packets will be coming to the default asterisk
IP address.
For the machine with multiple interfaces this could be not the right one
(not what we want).
Could it be configured (in rtp.conf or in sip.conf per context) ?
Thank you.
Alex Zarubin
-------------- next pa...
2003 Oct 27
0
Asterisk behind nat with hole, hardcoding solution
...ated below (the original line is
commented out):
static void build_contact(struct sip_pvt *p)
{
/* Construct Contact: header */
if (ourport != 5060)
snprintf(p->our_contact, sizeof(p->our_contact),
"<sip:%s@%s:%d>", p->exten, inet_ntoa(p->ourip), ourport);
else
// snprintf(p->our_contact, sizeof(p->our_contact),
"<sip:%s@%s>", p->exten, inet_ntoa(p->ourip));
snprintf(p->our_contact, sizeof(p->our_contact),
"<sip:%s@213.84.4.39>", p->exten, inet_ntoa(...
2010 Nov 05
2
How to append custom option to Contact: header on outgoing SIP INVITE msgs?
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
2006 Mar 08
2
REGISTER headers changed
Can someone help me with upgrading to the lastest version. I am using the
same sip.conf file, but the headers have changed and registration fails.
Has something change in the conf file that would cause this?
Notice 1.2.5 has no Authoization at all...
Regards,
Jason
Version 1.0.9
---------------------------
REGISTER sip:voip-ca35323.ocn.ne.jp SIP/2.0
Via: SIP/2.0/UDP
2003 Dec 10
0
Native Bridging and Polycom 600 Solved
...p *vrtp)
{
struct sip_request req;
if (p->canreinvite == REINVITE_UPDATE)
reqprep(&req, p, "UPDATE", 0);
else {
// BEGIN POLYCOM CHANGE
p->branch++;
snprintf(p->via, sizeof(p->via), "SIP/2.0/UDP
%s:%d;branch=z9hG4bK%08x", inet_ntoa(p->ourip), ourport, p->branch);
// END POLYCOM CHANGE
reqprep(&req, p, "INVITE", 0);
}
... the rest of the method follows.
Does anyone with any detailed knowledge of other SIP phones know if this will
cause something bad to happen? And, if any Asterisk developers are rea...
2007 Jul 12
0
No subject
...port= fields in sip replies.
In a nutshell, these fields are used to tell a sip entity
the address and port its request came from, and are extremely
useful in the presence of NATs, especially with symmetric NATs
where STUN is totally ineffective.
This patch stores the address and port in the 'ourip' field of
the dialog descriptor, so they can be reused in subsequent transactions.
As it is, it works well for things like REGISTER requiring authentication,
because the second REGISTER request (with auth credentials) will carry
the correct address. Maybe it can also be useful, in case of an ad...