Re: [OpenSIPS-Users] Voicemail doesn't work with Mediaproxy
Hi Duane,
In opensips 1.5.x, the usage of append_branch is slightly different than
before. Actually its usage from failure_route was aligned to the usage
from request route.
See: http://www.opensips.org/Resources/DocsMigration14to15
http://www.opensips.org/Resources/DocsMigration14to15#toc4
Regards,
Bogdan
Duane Larson wrote:
I am running OpenSIPS version 1.5.1
I am sure I found the append_branch command from someone's code I
found online.
I did have voicemail working to where there was two-way audio. I
thought I had solved everything, but then I came back from the weekend
holiday, tried to make a call between two natted devices and noticed
that I had broken that. I forget what I did to make voicemail work,
but I will try and find out again. I will do like you say and make
sure I am placing use_media_proxy in all the proper routes. I will
post once I figure it out. I hate it when people on this mailing list
figure it out and all the post is Fixed it without mentioning what
they did.
On Thu, Sep 10, 2009 at 5:35 PM, Thomas Gelf tho...@gelf.net
mailto:tho...@gelf.net wrote:
osiris123d wrote:
I am trying to use fix_nated_sdp(3, ip of mediaproxy0), but
I am sure I
am placing it and using it in the wrong spot in my script.
Don't do that. If you are using fix_nated_sdp together with
mediaproxy,
you will for sure mess up something. Calling use_media_proxy in your
INVITE route, your reply route and also for your in-dialog ReINVITES
will automagically fix your SDP.
failure_route[1] {
...
append_branch();
What OpenSIPS version are you using? In recent versions append_branch
is not needed here.
Best regards,
Thomas Gelf
--
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Re: [OpenSIPS-Users] Voicemail doesn't work with Mediaproxy
Thanks for the extra info.
Bogdan-Andrei Iancu wrote:
Hi Duane,
In opensips 1.5.x, the usage of append_branch is slightly different than
before. Actually its usage from failure_route was aligned to the usage
from request route.
See: http://www.opensips.org/Resources/DocsMigration14to15
http://www.opensips.org/Resources/DocsMigration14to15#toc4
Regards,
Bogdan
Duane Larson wrote:
I am running OpenSIPS version 1.5.1
I am sure I found the append_branch command from someone's code I
found online.
I did have voicemail working to where there was two-way audio. I
thought I had solved everything, but then I came back from the weekend
holiday, tried to make a call between two natted devices and noticed
that I had broken that. I forget what I did to make voicemail work,
but I will try and find out again. I will do like you say and make
sure I am placing use_media_proxy in all the proper routes. I will
post once I figure it out. I hate it when people on this mailing list
figure it out and all the post is Fixed it without mentioning what
they did.
On Thu, Sep 10, 2009 at 5:35 PM, Thomas Gelf tho...@gelf.net
mailto:tho...@gelf.net wrote:
osiris123d wrote:
I am trying to use fix_nated_sdp(3, ip of mediaproxy0), but
I am sure I
am placing it and using it in the wrong spot in my script.
Don't do that. If you are using fix_nated_sdp together with
mediaproxy,
you will for sure mess up something. Calling use_media_proxy in
your
INVITE route, your reply route and also for your in-dialog ReINVITES
will automagically fix your SDP.
failure_route[1] {
...
append_branch();
What OpenSIPS version are you using? In recent versions append_branch
is not needed here.
Best regards,
Thomas Gelf
--
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web: http://thomas.gelf.net/
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Re: [OpenSIPS-Users] Voicemail doesn't work with Mediaproxy
Fixed the issue. You were right. usemediaproxy wasn't getting called
because in the onreply route the first IF statement was never true because
bflag6 or bflag7 were never being set. So here is what I did to fix the
issue.
failure_route[1] {
##--
##-- If cancelled, exit.
##--
if (t_was_cancelled()) {
exit;
};
##--
##-- If busy send to the e-mail server, prefix the b
##-- character to indicate busy.
##--
if (t_check_status(486)) {
revert_uri();
append_hf(P-App-Name: voicemail\r\nP-App-Param:
Email-Address=blah.blahmail.com;Mode=both;usr=$tU;d...@$td\r\n);
rewritehostport(VoicemailIP);
# append_branch(); -DELETED this
setbflag(6);
route(1);
exit;
};
##--
##-- If timeout (408) or unavailable temporarily (480),
##-- prefix the uri with the ucharacter to indicate
##-- unanswered and send to the e-mail
##-- sever
##--
if (t_check_status(408) || t_check_status(480)) {
revert_uri();
append_hf(P-App-Name: voicemail\r\nP-App-Param:
Email-Address=blah.blahmail.com;Mode=both;usr=$tU;d...@$td\r\n);
rewritehostport(VoicemailIP);
#append_branch(); -Deleted this
setbflag(6);
route(1);
exit;
};
}
Thanks for the hint. This made me go back and re-read up on OpenSIPS
routing and really understand the routing logic. Thanks again.
osiris123d wrote:
I have been trying to fix this for the last 4 days. I have searched the
mailing list with the following keywords (mediaproxy rewritehostport
one-way audio, etc) and I find some posts that explain that my 200 ok
messages need say that the connection information needs to point to my
mediaproxy. I am trying to use fix_nated_sdp(3, ip of mediaproxy0),
but I am sure I am placing it and using it in the wrong spot in my script.
So here is the problem
When a user calls someone and they are busy or not there the caller will
be directed to voicemail by using the rewritehostport(ip of voicemail)
command. The caller is nated behind a router. The caller doesn't hear
the voicemail voice prompts, but the caller IS able to leave a voicemail
and be heard. So there is one-way audio going on where the caller can't
hear, but voicemail can. When I do a sniff I see that this is because the
caller's RTP is going directly to the Voicemail Servers Public IP address
and the Voicemails RTP is going to the Mediaproxy.
Here is the code in question
failure_route[1] {
##--
##-- If cancelled, exit.
##--
if (t_was_cancelled()) {
exit;
};
##--
##-- If busy send to the e-mail server, prefix the b
##-- character to indicate busy.
##--
if (t_check_status(486)) {
revert_uri();
append_hf(P-App-Name: voicemail\r\nP-App-Param:
Email-Address=blah.blahmail.com;Mode=both;usr=$tU;d...@$td\r\n);
rewritehostport(VoicemailIP);
append_branch();
route(1);
exit;
};
##--
##-- If timeout (408) or unavailable temporarily (480),
##-- prefix the uri with the ucharacter to indicate
##-- unanswered and send to the e-mail
##-- sever
##--
if (t_check_status(408) || t_check_status(480)) {
revert_uri();
append_hf(P-App-Name: voicemail\r\nP-App-Param:
Email-Address=blah.blahmail.com;Mode=both;usr=$tU;d...@$td\r\n);
rewritehostport(VoicemailIP);
append_branch();
route(1);
exit;
};
}
onreply_route[1] {
#
#-- On-replay block routing --
#
if ((isbflagset(6) || isbflagset(7))
(status=~(180)|(183)|2[0-9][0-9])) {
xlog(L_INFO, - Inside OnReply Route 1 with bflag
being 6 or 7 and a bunch other stuff);
if (search(^Content-Type:[ ]*application/sdp)) {
xlog(L_INFO, - Inside OnReply Route 1 with
content type equaling sdp);
append_hf(P-hint: onreply_route|usemediaproxy
\r\n);
use_media_proxy();
};
};
if (client_nat_test(1)) {
xlog(L_INFO, - Inside OnReply Route 1 and client
nat equaled 1);
append_hf(P-hint: Onreply-route - fixcontact \r\n);
fix_contact();
};
exit;
}
So how do I solve this issue?
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Sent from the OpenSIPS - Users mailing
[OpenSIPS-Users] Voicemail doesn't work with Mediaproxy
I have been trying to fix this for the last 4 days. I have searched the
mailing list with the following keywords (mediaproxy rewritehostport one-way
audio, etc) and I find some posts that explain that my 200 ok messages
need say that the connection information needs to point to my mediaproxy.
I am trying to use fix_nated_sdp(3, ip of mediaproxy0), but I am sure I
am placing it and using it in the wrong spot in my script.
So here is the problem
When a user calls someone and they are busy or not there the caller will be
directed to voicemail by using the rewritehostport(ip of voicemail)
command. The caller is nated behind a router. The caller doesn't hear the
voicemail voice prompts, but the caller IS able to leave a voicemail and be
heard. So there is one-way audio going on where the caller can't hear, but
voicemail can. When I do a sniff I see that this is because the caller's
RTP is going directly to the Voicemail Servers Public IP address and the
Voicemails RTP is going to the Mediaproxy.
Here is the code in question
failure_route[1] {
##--
##-- If cancelled, exit.
##--
if (t_was_cancelled()) {
exit;
};
##--
##-- If busy send to the e-mail server, prefix the b
##-- character to indicate busy.
##--
if (t_check_status(486)) {
revert_uri();
append_hf(P-App-Name: voicemail\r\nP-App-Param:
Email-Address=blah.blahmail.com;Mode=both;usr=$tU;d...@$td\r\n);
rewritehostport(VoicemailIP);
append_branch();
route(1);
exit;
};
##--
##-- If timeout (408) or unavailable temporarily (480),
##-- prefix the uri with the ucharacter to indicate
##-- unanswered and send to the e-mail
##-- sever
##--
if (t_check_status(408) || t_check_status(480)) {
revert_uri();
append_hf(P-App-Name: voicemail\r\nP-App-Param:
Email-Address=blah.blahmail.com;Mode=both;usr=$tU;d...@$td\r\n);
rewritehostport(VoicemailIP);
append_branch();
route(1);
exit;
};
}
onreply_route[1] {
#
#-- On-replay block routing --
#
if ((isbflagset(6) || isbflagset(7))
(status=~(180)|(183)|2[0-9][0-9])) {
xlog(L_INFO, - Inside OnReply Route 1 with bflag
being 6 or 7 and a bunch other stuff);
if (search(^Content-Type:[ ]*application/sdp)) {
xlog(L_INFO, - Inside OnReply Route 1 with
content type equaling sdp);
append_hf(P-hint: onreply_route|usemediaproxy
\r\n);
use_media_proxy();
};
};
if (client_nat_test(1)) {
xlog(L_INFO, - Inside OnReply Route 1 and client nat
equaled 1);
append_hf(P-hint: Onreply-route - fixcontact \r\n);
fix_contact();
};
exit;
}
So how do I solve this issue?
--
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Re: [OpenSIPS-Users] Voicemail doesn't work with Mediaproxy
I am running OpenSIPS version 1.5.1
I am sure I found the append_branch command from someone's code I found
online.
I did have voicemail working to where there was two-way audio. I thought I
had solved everything, but then I came back from the weekend holiday, tried
to make a call between two natted devices and noticed that I had broken
that. I forget what I did to make voicemail work, but I will try and find
out again. I will do like you say and make sure I am placing
use_media_proxy in all the proper routes. I will post once I figure it
out. I hate it when people on this mailing list figure it out and all the
post is Fixed it without mentioning what they did.
On Thu, Sep 10, 2009 at 5:35 PM, Thomas Gelf tho...@gelf.net wrote:
osiris123d wrote:
I am trying to use fix_nated_sdp(3, ip of mediaproxy0), but I am sure
I
am placing it and using it in the wrong spot in my script.
Don't do that. If you are using fix_nated_sdp together with mediaproxy,
you will for sure mess up something. Calling use_media_proxy in your
INVITE route, your reply route and also for your in-dialog ReINVITES
will automagically fix your SDP.
failure_route[1] {
...
append_branch();
What OpenSIPS version are you using? In recent versions append_branch
is not needed here.
Best regards,
Thomas Gelf
--
mail: tho...@gelf.net
web: http://thomas.gelf.net/
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