diff options
| author | mmichelson <mmichelson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2010-04-09 14:37:50 +0000 |
|---|---|---|
| committer | mmichelson <mmichelson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2010-04-09 14:37:50 +0000 |
| commit | 6c57cdc6ac82a6a6700ebdb788d690471d8fc49d (patch) | |
| tree | a4229d9951584abfc017622e1a74d2d358d9edf5 /CHANGES | |
| parent | c39bfddbfd1fcb7bb65c61e9592d024cc16d87d0 (diff) | |
func_srv and explicit specification of a remote IP for SIP.
From Review Board:
There are two interrelated changes here.
First, there is the introduction of func_srv. This adds two new read-only
dialplan functions, SRVQUERY and SRVRESULT. They work very similarly to the
ENUMQUERY and ENUMRESULT functions, except that this allows one to query SRV
records instead. In order to facilitate this work, I added a couple of new API
calls to srv.h. ast_srv_get_record_count tells the number of records returned
by an SRV lookup. This number is calculated at the time of the SRV lookup.
ast_srv_get_nth_record allows one to get a numbered SRV record.
Second, there is the modification to chan_sip that allows one to specify a
hostname or IP address (along with a port) to send an outgoing INVITE to when
dialing a SIP peer. This goes hand-in-hand with func_srv. You can query SRV
records and then use the host and port from the results to dial via a specific
host instead of what is configured in sip.conf.
Review: https://reviewboard.asterisk.org/r/608
SWP-1200
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@256485 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'CHANGES')
| -rw-r--r-- | CHANGES | 7 |
1 files changed, 7 insertions, 0 deletions
@@ -56,6 +56,9 @@ SIP Changes * Added 'use_q850_reason' configuration option for generating and parsing if available Reason: Q.850;cause=<cause code> header. It is implemented in some gateways for better passing PRI/SS7 cause codes via SIP. + * When dialing SIP peers, a new component may be added to the end of the dialstring + to indicate that a specific remote IP address or host should be used when dialing + the particular peer. The dialstring format is SIP/peer/exten/host_or_IP. IAX2 Changes ----------- @@ -146,6 +149,10 @@ Applications Dialplan Functions ------------------ + * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate + over SRV records associated with a specific service. From the CLI, type + 'core show function SRVQUERY' and 'core show function SRVRESULT' for more + details on how these may be used. * PITCH_SHIFT dialplan function added. This function can be used to modify the pitch of a channel's tx and rx audio streams. * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits |