488 Not acceptable here

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488 Not acceptable here

troxlinux
Hi list , I am making some tests with a server opensips and  adds him
the rtpproxy for the nat, the problem is that when adding the nat and
to call to an extension that  don't answer it doesn't jump me to the
asterisk voicemail and it shows me an error 488

I explain that in the same server opensips I have installed asterisk
, in the asterisk cli when the call is not answered he throws me this
error:


WARNING[3178]: chan_sip.c:5201 process_sdp: Unable to lookup host in
c= line, 'IN IP4 192.168.10.3192.168.10.3'

the sdp writes it twice , as I can avoid this?

## log sip##


#
U +0.019539 192.168.10.30:5064 -> 192.168.10.3:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.3:5060;branch=0
From: sip:pinger@192.168.10.3;tag=cd0baa81
To: sip:192.168.10.30:5064;tag=a8c59398c8984470
Call-ID: 9528c331-0c6e3641-f@192.168.10.3
CSeq: 1 OPTIONS
User-Agent: Grandstream GXP2020 1.1.6.16
Contact: <sip:201@192.168.10.30:5064;transport=udp>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


#
U +2.000872 192.168.10.3:5060 -> 192.168.10.3:5070
INVITE sip:u201@192.168.10.3:5070 SIP/2.0
Record-Route: <sip:192.168.10.3;lr=on;ftag=42d5a8fbdbb60640o0>
Via: SIP/2.0/UDP 192.168.10.3;branch=z9hG4bKb977.26ab18f7.1
Via: SIP/2.0/UDP
192.168.10.19:5060;rport=5060;received=192.168.10.19;branch=z9hG4bK-a8ea22ed
From: <sip:200@192.168.10.3>;tag=42d5a8fbdbb60640o0
To: "Opensips-14x" <sip:201@192.168.10.3>
Call-ID: f6dccfd7-7f5fad14@192.168.10.19
CSeq: 102 INVITE
Max-Forwards: 69
Contact: <sip:200@192.168.10.19:5060;nat=yes;nat=yes>
Expires: 240
User-Agent: Linksys/SPA942-6.1.3(a)
Content-Length: 263
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
P-hint: inbound->inbound
P-hint: Route[20]: Rtpproxy
P-hint: Route[20]: Rtpproxy

v=0
o=- 811136 811136 IN IP4 192.168.10.19
s=-
c=IN IP4 192.168.10.3192.168.10.3
t=0 0
m=audio 3500435006 RTP/AVP 18 101
a=rtpmap:18 G729a/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
a=nortpproxy:yes
a=nortpproxy:yes

#
U +0.000123 192.168.10.3:5060 -> 192.168.10.30:5064
CANCEL sip:201@192.168.10.30:5064;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.10.3;branch=z9hG4bKb977.26ab18f7.0
From: <sip:200@192.168.10.3>;tag=42d5a8fbdbb60640o0
Call-ID: f6dccfd7-7f5fad14@192.168.10.19
To: "Opensips-14x" <sip:201@192.168.10.3>
CSeq: 102 CANCEL
Max-Forwards: 70
User-Agent: OpenSIPS (1.4.5-notls (i386/linux))
Content-Length: 0


#
U +0.001572 192.168.10.3:5070 -> 192.168.10.3:5060
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP
192.168.10.3;branch=z9hG4bKb977.26ab18f7.1;received=192.168.10.3
Via: SIP/2.0/UDP
192.168.10.19:5060;rport=5060;received=192.168.10.19;branch=z9hG4bK-a8ea22ed
From: <sip:200@192.168.10.3>;tag=42d5a8fbdbb60640o0
To: "Opensips-14x" <sip:201@192.168.10.3>;tag=as50300fb2
Call-ID: f6dccfd7-7f5fad14@192.168.10.19
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


#
U +0.000244 192.168.10.3:5060 -> 192.168.10.3:5070
ACK sip:u201@192.168.10.3:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.3;branch=z9hG4bKb977.26ab18f7.1
From: <sip:200@192.168.10.3>;tag=42d5a8fbdbb60640o0
Call-ID: f6dccfd7-7f5fad14@192.168.10.19
To: "Opensips-14x" <sip:201@192.168.10.3>;tag=as50300fb2
CSeq: 102 ACK
Max-Forwards: 70
User-Agent: OpenSIPS (1.4.5-notls (i386/linux))
Content-Length: 0


regardss

--
rickygm

http://gnuforever.homelinux.com

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Re: 488 Not acceptable here

Romanov Vladimir
This is bug. I solve this problem by upgrading to 1.5.0

-----------------
Vladimir Romanov
Yota Lab | http://www.yota.ru
CTO
+7 (960) 239-0853


-----Original Message-----
From: [hidden email] [mailto:[hidden email]] On Behalf Of troxlinux
Sent: Saturday, April 11, 2009 9:17 AM
To: [hidden email]
Subject: [OpenSIPS-Users] 488 Not acceptable here

Hi list , I am making some tests with a server opensips and  adds him
the rtpproxy for the nat, the problem is that when adding the nat and
to call to an extension that  don't answer it doesn't jump me to the
asterisk voicemail and it shows me an error 488

I explain that in the same server opensips I have installed asterisk
, in the asterisk cli when the call is not answered he throws me this
error:


WARNING[3178]: chan_sip.c:5201 process_sdp: Unable to lookup host in
c= line, 'IN IP4 192.168.10.3192.168.10.3'

the sdp writes it twice , as I can avoid this?

## log sip##


#
U +0.019539 192.168.10.30:5064 -> 192.168.10.3:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.3:5060;branch=0
From: sip:pinger@192.168.10.3;tag=cd0baa81
To: sip:192.168.10.30:5064;tag=a8c59398c8984470
Call-ID: 9528c331-0c6e3641-f@192.168.10.3
CSeq: 1 OPTIONS
User-Agent: Grandstream GXP2020 1.1.6.16
Contact: <sip:201@192.168.10.30:5064;transport=udp>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


#
U +2.000872 192.168.10.3:5060 -> 192.168.10.3:5070
INVITE sip:u201@192.168.10.3:5070 SIP/2.0
Record-Route: <sip:192.168.10.3;lr=on;ftag=42d5a8fbdbb60640o0>
Via: SIP/2.0/UDP 192.168.10.3;branch=z9hG4bKb977.26ab18f7.1
Via: SIP/2.0/UDP
192.168.10.19:5060;rport=5060;received=192.168.10.19;branch=z9hG4bK-a8ea22ed
From: <sip:200@192.168.10.3>;tag=42d5a8fbdbb60640o0
To: "Opensips-14x" <sip:201@192.168.10.3>
Call-ID: f6dccfd7-7f5fad14@192.168.10.19
CSeq: 102 INVITE
Max-Forwards: 69
Contact: <sip:200@192.168.10.19:5060;nat=yes;nat=yes>
Expires: 240
User-Agent: Linksys/SPA942-6.1.3(a)
Content-Length: 263
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
P-hint: inbound->inbound
P-hint: Route[20]: Rtpproxy
P-hint: Route[20]: Rtpproxy

v=0
o=- 811136 811136 IN IP4 192.168.10.19
s=-
c=IN IP4 192.168.10.3192.168.10.3
t=0 0
m=audio 3500435006 RTP/AVP 18 101
a=rtpmap:18 G729a/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
a=nortpproxy:yes
a=nortpproxy:yes

#
U +0.000123 192.168.10.3:5060 -> 192.168.10.30:5064
CANCEL sip:201@192.168.10.30:5064;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.10.3;branch=z9hG4bKb977.26ab18f7.0
From: <sip:200@192.168.10.3>;tag=42d5a8fbdbb60640o0
Call-ID: f6dccfd7-7f5fad14@192.168.10.19
To: "Opensips-14x" <sip:201@192.168.10.3>
CSeq: 102 CANCEL
Max-Forwards: 70
User-Agent: OpenSIPS (1.4.5-notls (i386/linux))
Content-Length: 0


#
U +0.001572 192.168.10.3:5070 -> 192.168.10.3:5060
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP
192.168.10.3;branch=z9hG4bKb977.26ab18f7.1;received=192.168.10.3
Via: SIP/2.0/UDP
192.168.10.19:5060;rport=5060;received=192.168.10.19;branch=z9hG4bK-a8ea22ed
From: <sip:200@192.168.10.3>;tag=42d5a8fbdbb60640o0
To: "Opensips-14x" <sip:201@192.168.10.3>;tag=as50300fb2
Call-ID: f6dccfd7-7f5fad14@192.168.10.19
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


#
U +0.000244 192.168.10.3:5060 -> 192.168.10.3:5070
ACK sip:u201@192.168.10.3:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.3;branch=z9hG4bKb977.26ab18f7.1
From: <sip:200@192.168.10.3>;tag=42d5a8fbdbb60640o0
Call-ID: f6dccfd7-7f5fad14@192.168.10.19
To: "Opensips-14x" <sip:201@192.168.10.3>;tag=as50300fb2
CSeq: 102 ACK
Max-Forwards: 70
User-Agent: OpenSIPS (1.4.5-notls (i386/linux))
Content-Length: 0


regardss

--
rickygm

http://gnuforever.homelinux.com

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Re: 488 Not acceptable here

troxlinux
Hi Romanov , I upgrade my opensips with the last version 1.5 but i
have the same problem

in asterisk cli,  I see that the opensips writes the address ip twice


<--- SIP read from 192.168.10.3:5060 --->
INVITE sip:u200@192.168.10.3:5070 SIP/2.0
Record-Route: <sip:192.168.10.3;lr=on;ftag=26f16c5a110a5bd9>
Via: SIP/2.0/UDP 192.168.10.3;branch=z9hG4bK3bba.4fa31861.1
Via: SIP/2.0/UDP
192.168.10.30:5064;rport=5064;received=192.168.10.30;branch=z9hG4bKdbd6939c968835d1
From: "Opensips-14x" <sip:201@192.168.10.3>;tag=26f16c5a110a5bd9
To: <sip:200@192.168.10.3>
Contact: <sip:201@192.168.10.30:5064;transport=udp>
Supported: replaces, timer, path
Call-ID: e46fec9f04600434@192.168.10.30
CSeq: 47478 INVITE
User-Agent: Grandstream GXP2020 1.1.6.44
Max-Forwards: 69
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 302
P-hint: inbound->inbound

v=0
o=201 8000 8001 IN IP4 192.168.10.3192.168.10.3
s=SIP Call
c=IN IP4 192.168.10.3192.168.10.3
t=0 0
m=audio 3500635006 RTP/AVP 18 0 101
a=sendrecv
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
a=nortpproxy:yes
a=nortpproxy:yes



any idea , how solve this problem ?

regardss

2009/4/11 Romanov Vladimir <[hidden email]>:
> This is bug. I solve this problem by upgrading to 1.5.0
>
> -----------------

--
rickygm

http://gnuforever.homelinux.com

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Re: 488 Not acceptable here

Bogdan-Andrei Iancu
Hi,

But the INVITE seams to go twice through some NAT logic as I see the
"a=nortpproxy:yes" header twice. Can you upload somewhere the entire SIP
trace (inbound and outbound part) ?

Regards,
Bogdan

troxlinux wrote:

> Hi Romanov , I upgrade my opensips with the last version 1.5 but i
> have the same problem
>
> in asterisk cli,  I see that the opensips writes the address ip twice
>
>
> <--- SIP read from 192.168.10.3:5060 --->
> INVITE sip:u200@192.168.10.3:5070 SIP/2.0
> Record-Route: <sip:192.168.10.3;lr=on;ftag=26f16c5a110a5bd9>
> Via: SIP/2.0/UDP 192.168.10.3;branch=z9hG4bK3bba.4fa31861.1
> Via: SIP/2.0/UDP
> 192.168.10.30:5064;rport=5064;received=192.168.10.30;branch=z9hG4bKdbd6939c968835d1
> From: "Opensips-14x" <sip:201@192.168.10.3>;tag=26f16c5a110a5bd9
> To: <sip:200@192.168.10.3>
> Contact: <sip:201@192.168.10.30:5064;transport=udp>
> Supported: replaces, timer, path
> Call-ID: e46fec9f04600434@192.168.10.30
> CSeq: 47478 INVITE
> User-Agent: Grandstream GXP2020 1.1.6.44
> Max-Forwards: 69
> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
> Content-Type: application/sdp
> Content-Length: 302
> P-hint: inbound->inbound
>
> v=0
> o=201 8000 8001 IN IP4 192.168.10.3192.168.10.3
> s=SIP Call
> c=IN IP4 192.168.10.3192.168.10.3
> t=0 0
> m=audio 3500635006 RTP/AVP 18 0 101
> a=sendrecv
> a=rtpmap:18 G729/8000
> a=rtpmap:0 PCMU/8000
> a=ptime:20
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-11
> a=nortpproxy:yes
> a=nortpproxy:yes
>
>
>
> any idea , how solve this problem ?
>
> regardss
>
> 2009/4/11 Romanov Vladimir <[hidden email]>:
>  
>> This is bug. I solve this problem by upgrading to 1.5.0
>>
>> -----------------
>>    
>
>  
> ------------------------------------------------------------------------
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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Re: 488 Not acceptable here

troxlinux
2009/4/13 Bogdan-Andrei Iancu <[hidden email]>:
> Hi,

Hi Bogdan

>
> But the INVITE seams to go twice through some NAT logic as I see the
> "a=nortpproxy:yes" header twice. Can you upload somewhere the entire SIP
> trace (inbound and outbound part) ?
>

do I see that the opensips writes me twice the address ip in the sdp,
like I can I remove this?

                route[10] {
                #from an internal domain -> inbound
        #Native SIP destinations are handled using the location table
        append_hf("P-hint: inbound->inbound \r\n");
        if (uri=~"^sip:9[0-9]*@") {
        if (is_user_in("credentials", "local")){
        route(4);
        exit;
        } else {
                sl_send_reply("403", "No tienes permiso para llamadas locales");
                exit;
                        };
                                };

                if (!lookup("location")) {
                xlog("L_INFO","$C(rx)404 User Not Found $C(xx)\n");
                if (does_uri_exist()) {
                revert_uri();
                prefix("u");
                rewritehostport("192.168.10.3:5070");
                route(1);
                } else {
                sl_send_reply("404", "Not Found");
        exit;
        };
                sl_send_reply("404", "Not Found");
                exit;
                };

                route(1);
        }

        route[11] {
# from an internal domain -> outbound
# Simply route the call outbound using DNS search
        append_hf("P-hint: inbound->outbound \r\n");
        route(1);
        }
        route[12] {
# From an external domain -> inbound
# Verify aliases, if found replace R-URI.
        lookup("aliases");
        if (!lookup("location")) {
        xlog("L_INFO", "$C(rx)404 Lo siento usuario no encontrado $C(xx)\n");
        sl_send_reply("404", "Not Found");
        exit;
        };
        route(1);
        }
        route[13] {
#From an external domain outbound
#we are not accepting these calls
        append_hf("P-hint: outbound->inbound \r\n");
        sl_send_reply("403", "Forbidden");
        exit;
        }


        route[4] {
        rewritehostport("192.168.10.3:5070");
        route(1);



I attach  the sip log

regardss

--
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Re: 488 Not acceptable here

Bogdan-Andrei Iancu
Hi,

Looking at the SIP trace, I can see you do serial forking - you do set
the RTProxy for the first branch (sending to 192.168.10.19:5060) and
probably you do it again in failure route when you create the second branch.

You cannot call force_rtp_proxy() twice for the same message - this is
why you get the double IPs.

Regards,
Bogdan


troxlinux wrote:

> 2009/4/13 Bogdan-Andrei Iancu <[hidden email]>:
>  
>> Hi,
>>    
>
> Hi Bogdan
>
>  
>> But the INVITE seams to go twice through some NAT logic as I see the
>> "a=nortpproxy:yes" header twice. Can you upload somewhere the entire SIP
>> trace (inbound and outbound part) ?
>>
>>    
>
> do I see that the opensips writes me twice the address ip in the sdp,
> like I can I remove this?
>
>                 route[10] {
>                 #from an internal domain -> inbound
>         #Native SIP destinations are handled using the location table
>         append_hf("P-hint: inbound->inbound \r\n");
>         if (uri=~"^sip:9[0-9]*@") {
>         if (is_user_in("credentials", "local")){
>         route(4);
>         exit;
>         } else {
>                 sl_send_reply("403", "No tienes permiso para llamadas locales");
>                 exit;
>                         };
>                                 };
>
>                 if (!lookup("location")) {
>                 xlog("L_INFO","$C(rx)404 User Not Found $C(xx)\n");
>                 if (does_uri_exist()) {
>                 revert_uri();
>                 prefix("u");
>                 rewritehostport("192.168.10.3:5070");
>                 route(1);
>                 } else {
>                 sl_send_reply("404", "Not Found");
>         exit;
>         };
>                 sl_send_reply("404", "Not Found");
>                 exit;
>                 };
>
>                 route(1);
>         }
>
>         route[11] {
> # from an internal domain -> outbound
> # Simply route the call outbound using DNS search
>         append_hf("P-hint: inbound->outbound \r\n");
>         route(1);
>         }
>         route[12] {
> # From an external domain -> inbound
> # Verify aliases, if found replace R-URI.
>         lookup("aliases");
>         if (!lookup("location")) {
>         xlog("L_INFO", "$C(rx)404 Lo siento usuario no encontrado $C(xx)\n");
>         sl_send_reply("404", "Not Found");
>         exit;
>         };
>         route(1);
>         }
>         route[13] {
> #From an external domain outbound
> #we are not accepting these calls
>         append_hf("P-hint: outbound->inbound \r\n");
>         sl_send_reply("403", "Forbidden");
>         exit;
>         }
>
>
>         route[4] {
>         rewritehostport("192.168.10.3:5070");
>         route(1);
>
>
>
> I attach  the sip log
>
> regardss
>
>  


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Re: 488 Not acceptable here

Brett Nemeroff
Bogdan,
I know this may be beside the point here, but it seems like it should
be pretty easy to make the rtpproxy module not allow this, or to at
least spit out an error saying that it's been set already.  Point
being, the result of this scripting error isn't obvious what was done
wrong.

Just a thought.. I've done this by mistake before as well...
-Brett


On Wed, Apr 15, 2009 at 5:38 AM, Bogdan-Andrei Iancu
<[hidden email]> wrote:

> Hi,
>
> Looking at the SIP trace, I can see you do serial forking - you do set
> the RTProxy for the first branch (sending to 192.168.10.19:5060) and
> probably you do it again in failure route when you create the second branch.
>
> You cannot call force_rtp_proxy() twice for the same message - this is
> why you get the double IPs.
>
> Regards,
> Bogdan
>
>
> troxlinux wrote:
>> 2009/4/13 Bogdan-Andrei Iancu <[hidden email]>:
>>
>>> Hi,
>>>
>>
>> Hi Bogdan
>>
>>
>>> But the INVITE seams to go twice through some NAT logic as I see the
>>> "a=nortpproxy:yes" header twice. Can you upload somewhere the entire SIP
>>> trace (inbound and outbound part) ?
>>>
>>>
>>
>> do I see that the opensips writes me twice the address ip in the sdp,
>> like I can I remove this?
>>
>>                 route[10] {
>>                 #from an internal domain -> inbound
>>         #Native SIP destinations are handled using the location table
>>         append_hf("P-hint: inbound->inbound \r\n");
>>         if (uri=~"^sip:9[0-9]*@") {
>>         if (is_user_in("credentials", "local")){
>>         route(4);
>>         exit;
>>         } else {
>>                 sl_send_reply("403", "No tienes permiso para llamadas locales");
>>                 exit;
>>                         };
>>                                 };
>>
>>                 if (!lookup("location")) {
>>                 xlog("L_INFO","$C(rx)404 User Not Found $C(xx)\n");
>>                 if (does_uri_exist()) {
>>                 revert_uri();
>>                 prefix("u");
>>                 rewritehostport("192.168.10.3:5070");
>>                 route(1);
>>                 } else {
>>                 sl_send_reply("404", "Not Found");
>>         exit;
>>         };
>>                 sl_send_reply("404", "Not Found");
>>                 exit;
>>                 };
>>
>>                 route(1);
>>         }
>>
>>         route[11] {
>> # from an internal domain -> outbound
>> # Simply route the call outbound using DNS search
>>         append_hf("P-hint: inbound->outbound \r\n");
>>         route(1);
>>         }
>>         route[12] {
>> # From an external domain -> inbound
>> # Verify aliases, if found replace R-URI.
>>         lookup("aliases");
>>         if (!lookup("location")) {
>>         xlog("L_INFO", "$C(rx)404 Lo siento usuario no encontrado $C(xx)\n");
>>         sl_send_reply("404", "Not Found");
>>         exit;
>>         };
>>         route(1);
>>         }
>>         route[13] {
>> #From an external domain outbound
>> #we are not accepting these calls
>>         append_hf("P-hint: outbound->inbound \r\n");
>>         sl_send_reply("403", "Forbidden");
>>         exit;
>>         }
>>
>>
>>         route[4] {
>>         rewritehostport("192.168.10.3:5070");
>>         route(1);
>>
>>
>>
>> I attach  the sip log
>>
>> regardss
>>
>>
>
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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Re: 488 Not acceptable here

Bogdan-Andrei Iancu
Indeed, this will be an idea - can you open a bug or feature request on
this?

Thanks and regards,
Bogdan

Brett Nemeroff wrote:

> Bogdan,
> I know this may be beside the point here, but it seems like it should
> be pretty easy to make the rtpproxy module not allow this, or to at
> least spit out an error saying that it's been set already.  Point
> being, the result of this scripting error isn't obvious what was done
> wrong.
>
> Just a thought.. I've done this by mistake before as well...
> -Brett
>
>
> On Wed, Apr 15, 2009 at 5:38 AM, Bogdan-Andrei Iancu
> <[hidden email]> wrote:
>  
>> Hi,
>>
>> Looking at the SIP trace, I can see you do serial forking - you do set
>> the RTProxy for the first branch (sending to 192.168.10.19:5060) and
>> probably you do it again in failure route when you create the second branch.
>>
>> You cannot call force_rtp_proxy() twice for the same message - this is
>> why you get the double IPs.
>>
>> Regards,
>> Bogdan
>>
>>
>> troxlinux wrote:
>>    
>>> 2009/4/13 Bogdan-Andrei Iancu <[hidden email]>:
>>>
>>>      
>>>> Hi,
>>>>
>>>>        
>>> Hi Bogdan
>>>
>>>
>>>      
>>>> But the INVITE seams to go twice through some NAT logic as I see the
>>>> "a=nortpproxy:yes" header twice. Can you upload somewhere the entire SIP
>>>> trace (inbound and outbound part) ?
>>>>
>>>>
>>>>        
>>> do I see that the opensips writes me twice the address ip in the sdp,
>>> like I can I remove this?
>>>
>>>                 route[10] {
>>>                 #from an internal domain -> inbound
>>>         #Native SIP destinations are handled using the location table
>>>         append_hf("P-hint: inbound->inbound \r\n");
>>>         if (uri=~"^sip:9[0-9]*@") {
>>>         if (is_user_in("credentials", "local")){
>>>         route(4);
>>>         exit;
>>>         } else {
>>>                 sl_send_reply("403", "No tienes permiso para llamadas locales");
>>>                 exit;
>>>                         };
>>>                                 };
>>>
>>>                 if (!lookup("location")) {
>>>                 xlog("L_INFO","$C(rx)404 User Not Found $C(xx)\n");
>>>                 if (does_uri_exist()) {
>>>                 revert_uri();
>>>                 prefix("u");
>>>                 rewritehostport("192.168.10.3:5070");
>>>                 route(1);
>>>                 } else {
>>>                 sl_send_reply("404", "Not Found");
>>>         exit;
>>>         };
>>>                 sl_send_reply("404", "Not Found");
>>>                 exit;
>>>                 };
>>>
>>>                 route(1);
>>>         }
>>>
>>>         route[11] {
>>> # from an internal domain -> outbound
>>> # Simply route the call outbound using DNS search
>>>         append_hf("P-hint: inbound->outbound \r\n");
>>>         route(1);
>>>         }
>>>         route[12] {
>>> # From an external domain -> inbound
>>> # Verify aliases, if found replace R-URI.
>>>         lookup("aliases");
>>>         if (!lookup("location")) {
>>>         xlog("L_INFO", "$C(rx)404 Lo siento usuario no encontrado $C(xx)\n");
>>>         sl_send_reply("404", "Not Found");
>>>         exit;
>>>         };
>>>         route(1);
>>>         }
>>>         route[13] {
>>> #From an external domain outbound
>>> #we are not accepting these calls
>>>         append_hf("P-hint: outbound->inbound \r\n");
>>>         sl_send_reply("403", "Forbidden");
>>>         exit;
>>>         }
>>>
>>>
>>>         route[4] {
>>>         rewritehostport("192.168.10.3:5070");
>>>         route(1);
>>>
>>>
>>>
>>> I attach  the sip log
>>>
>>> regardss
>>>
>>>
>>>      
>> _______________________________________________
>> Users mailing list
>> [hidden email]
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>    
>
>  


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