30 Jan
2006
30 Jan
'06
9:19 p.m.
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).
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
>
30 Jan
30 Jan
10:02 p.m.
New subject: [Users] Using uac_replace_from to munge username between ser and Asterisk?
Bogdan-Andrei Iancu wrote:
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
regards,
bogdan
Barry Flanagan wrote:
--
-Barry Flanagan
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
>
3 Feb
3 Feb
1:38 p.m.
New subject: [Users] Using uac_replace_from to munge username between ser and Asterisk?
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
>
10:50 p.m.
New subject: [Users] Using uac_replace_from to munge username between ser and Asterisk?
Hi Barry,
indeed, messing with the FROM URI may brake things that rely on it...
not sure about mediaproxy, but nathelper uses only the from-tag, which
is not replaced....I would say it should work..have you tried?
regards,
bogdan
Barry Flanagan wrote:
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
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
>>
4 Feb
4 Feb
12:16 a.m.
New subject: [Users] Using uac_replace_from to munge username between ser and Asterisk?
Bogdan-Andrei Iancu wrote:
Hi Barry,
indeed, messing with the FROM URI may brake things that rely on it...
not sure about mediaproxy, but nathelper uses only the from-tag, which
is not replaced....I would say it should work..have you tried?
Yes, nathelper on its own appears to be ok.
Is nathelper on its own generally sufficient for a public SIP server
where most clients are behind NAT?
Thanks.
-Barry
regards,
bogdan
Barry Flanagan wrote:
--
-Barry Flanagan
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
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
>>>
>
>
12:26 a.m.
New subject: [Users] Using uac_replace_from to munge username between ser and Asterisk?
Barry Flanagan wrote:
Bogdan-Andrei Iancu wrote:
yes, nathelper + rtpproxy can do the job.
regards,
bogdan
Hi Barry,
indeed, messing with the FROM URI may brake things that rely on it...
not sure about mediaproxy, but nathelper uses only the from-tag,
which is not replaced....I would say it should work..have you tried?
Yes, nathelper on its own appears to be ok.
Is nathelper on its own generally sufficient for a public SIP server
where most clients are behind NAT?
6870
days inactive
6874
days old
5 comments
2 participants
participants (2)
-
Barry Flanagan -
Bogdan-Andrei Iancu