3 Feb
2006
3 Feb
'06
1:38 p.m.
Barry Flanagan wrote:
Bogdan-Andrei Iancu wrote:
Hmm, OK, it did work until I started using nathelper and mediaproxy.
With mediaprixy I get no audio, and Asterisk is retransmitting
It appears that mediaproxy is looking for the "unmunged" username. Below
is the mediaproxy log for this call. It is expecting
from:sipps2@sip.domain.com, whereas asterisk is sending
sipps2_domain.com(a)sip.domain.com
Feb 3 11:25:05 www1 mediaproxy[7461]: command request
1647296070-45779966(a)XXX.XXX.96.225
XXX.XXX.96.225:12047:audio,XXX.XXX.96.225:12049:video XXX.XXX.96.225
sip.domain.com local sip.domain.com local
Nero=20SIPPS=20IP=20Phone=20Version=202.1.3.25
info=from:sipps2@sip.domain.com,to:0863854334@sip.domain.com,fromtag:622fb836,totag:
Feb 3 11:25:05 www1 mediaproxy[7461]: session
1647296070-45779966(a)XXX.XXX.96.225: started. listening on
XXX.XXX.1.16:35194,35196
Here Asterisk is retransmitting:
Retransmitting #1 (NAT) to XXX.XXX.1.16:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
XXX.XXX.1.16;branch=z9hG4bK9e52.4f879791.0;received=XXX.XXX.1.16
Via: SIP/2.0/UDP
XXX.XXX.96.225:12046;branch=z9hG4bKnp1643392953-45a6e6feXXX.XXX.96.225;rport=12020
Record-Route: <sip:XXX.XXX.1.16:5060;nat=yes;ftag=61f4a3ce;lr=on>
From: ""Barry Flanagan""
<sip:sipps2_domain.com@sip.domain.com>;tag=61f4a3ce
To: <sip:0863854334@sip.domain.com>;tag=as53b2855b
Call-ID: 1643422664-49d79852(a)XXX.XXX.96.225
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0863854334@XXX.XXX.1.68>
Content-Type: application/sdp
Content-Length: 201
Any idea?
Thanks.
-Barry Flanagan
if you enable auto from_restore_mode, you do not
need to perform any
restore from script. Just replace the from in the initial INVITE and
this is it - all replies and sequential request would be auto fixed
(restore/replace).
Aha! Thank you very much - that seems to have done the trick!
Regards,
-Barry Flanagan
--
-Barry Flanagan
regards,
bogdan
Barry Flanagan wrote:
Bogdan-Andrei Iancu wrote:
Hi Barry,
have you set auto from restoring? See:
http://openser.org/docs/modules/1.1.x/uac.html#AEN75
Yes, but I am not sure where it is supposed to go.
I have the following in just before relaying to Asterisk:
rewritehostport("XXX.XXX.XXX.XXX:5060");
uac_replace_from("$fn","sip:$au_$ar@$fd");
append_hf("P-hint: GATEWAY\r\n");
t_relay("udp:XXX.XXX.XXX.XXX:5060");
and I put in uac_restore_from(); just after the record_route()
with all the other modparams I have:
modparam("uac","from_restore_mode","auto")
Thanks for the help.
-Barry
regards,
bogdan
Barry Flanagan wrote:
>
> So, the only way around it that I can see is to somehow have
> OpenSER change the username to username_domain so that each will be
> unique.
>
> It looks like uac_from_replace should handle this. I have tried it,
> and I can see that Asterisk does in fact get user_domain@domain in
> the first invite, but thereafter for some reason OpenSER changes it
> to just _@domain for subsequent requests.
>
>
> Regards,
>
> -Barry
>