OK, Thanks Daniel. I will upgrade.
Regards.
Luis.
2013/7/29 Daniel-Constantin Mierla
> Hello,
>
> if you run release 3.0.0 then you should update at least to latest in its
> series 3.0.x. But I strongly recommend to go for a newer release, the best
> is 4.0.x, or at least 3.3.x. Anyhow, alwa
Hello,
if you run release 3.0.0 then you should update at least to latest in
its series 3.0.x. But I strongly recommend to go for a newer release,
the best is 4.0.x, or at least 3.3.x. Anyhow, always upgrade in the
releases series, so whenever you run x.y.z, be sure that you have the
highest
Hello,
load debugger module, set its cfgtrace parameter to 1 and then send the
logs messages printed for the entire call (from the initial INVITE to
the end).
Also, use ngrep to get the sip traffic and send that as well. It will
help to see what is executed and what could be done wrong in co
I have checked that I'm , experiencing the same problem when the
redirection to voicemail is originated by the destination UAC via 302
message. Kamailio sends the packet to the destination UAC, even when I set
$du to null.
??¿?¿?¿¿??¿
if ($rU=~"^voicemail.*") {
$du = $null;
rem
OK, Daniel and thanks for your help,
I see that you don't append brach but you are calling route(RELAY) instead
of t_relay() directly. I have tryed with this configuration within failure
route:
if (t_check_status("486|408")) {
#revert_uri();
prefix("voicemail");
remove_hf
Hi,
do an
- exit;
after t_relay().
if (t_check_status("486|408")) {
revert_uri();
prefix("voicemail");
remove_hf("P-App-Name");
append_hf("P-App-Name: voicemail\r\n");
append_hf("P-App-Param: mod=box;usr= $rU;dom=sipproxy.a.com;uid=$
rU;did=sipproxy.a.c
On Thursday 25 July 2013 16:30:21 you wrote:
> if (t_check_status("486|408")) {
>
> revert_uri();
> prefix("voicemail");
> remove_hf("P-App-Name");
> append_hf("P-App-Name: voicemail\r\n");
> append_hf("P-App-Param: mod=box;usr= $rU;dom=sipproxy.a.com
> ;ui
Hello Daniel,
I have tried without append_branch(); and it just don't create the new
branch and it sends back the 486 message to the UAC that originated the
call:
if (t_check_status("486|408")) {
revert_uri();
prefix("voicemail");
remove_hf("P-App-Name");
append_h
On Wednesday 24 July 2013 20:41:04 LAA wrote:
> May be I'm loosing something. I have changed my config as you
> suggested (I thing so...):
>
> if (t_check_status("486|408")) {
...
> $du = $null;
> #$du = "sip:192.168.0.197";
> append_branch();
> t_relay();
Did you
Hello Hero,
Thanks for your help.
May be I'm loosing something. I have changed my config as you
suggested (I thing so...):
if (t_check_status("486|408")) {
revert_uri();
prefix("voicemail");
remove_hf("P-App-Name");
append_hf("P-App-Name: voicemail\r\n");
had the same issue here. you have to manually set $du=$null, else it
doesn't get reset for the failure branch.
On 7/23/13, LAA wrote:
> Hi all,
>
> I'm running Kamailio 3.0.0, with SEMS integration as Media Server for Voice
> mail. I'm trying to get a configuration to forward calls on busy to voi
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