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| author | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-02-01 22:21:56 +0000 |
|---|---|---|
| committer | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-02-01 22:21:56 +0000 |
| commit | eeb5d7afca37fbe34ebd17f0740f3360c063a92c (patch) | |
| tree | 97d2b91d6f507098ba928abf637c3511ed8895a9 | |
| parent | 75b6a4f6978e3ae4d0d45a1e8525d7519d8bda3e (diff) | |
Copy noncodeccapability over to the joint variable so that telephone-event will get transmitted in the sent INVITE.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@53103 f38db490-d61c-443f-a65b-d21fe96a405b
| -rw-r--r-- | channels/chan_sip.c | 1 |
1 files changed, 1 insertions, 0 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 38084e663..f7dcda5f8 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -2083,6 +2083,7 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout) if ( res != -1 ) { p->callingpres = ast->cid.cid_pres; p->jointcapability = p->capability; + p->jointnoncodeccapability = p->noncodeccapability; transmit_invite(p, SIP_INVITE, 1, 2); if (p->maxtime) { /* Initialize auto-congest time */ |