Thread: [OpenSIPStack] help need , i am not able to send the register request
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From: prasad k. <pra...@gm...> - 2008-05-20 10:54:58
|
Hi all,ope...@li...
i am using eXosip as sip stack , i am able to initilaze the osip and exosip
and the problem with eXosip _send_register ,where iam not able to fix up the
bug can any bodyhelp me
#include<eXosip2/eXosip.h>
#include<sys/socket.h>
#include<netinet/in.h>
#include<arpa/inet.h>
main() {
int i;
int id;
osip_message_t* reg = NULL;
TRACE_INITIALIZE(6,stdout);
i = eXosip_init();
if(i !=0)
{
printf("Not Initilazed\n");
return -1;
}
else
{
printf("\nInitilazed.....\n");
}
i = eXosip_listen_addr(IPPROTO_UDP,"10.232.19.204
",10234,AF_INET,0);
if(i != 0)
{
eXosip_quit();
printf("Could not initilaize transport layer\n");
return -1;
}
else
{
printf("\nTransport layer initilazed
......................\n");
printf("%d\n", i);
}
eXosip_lock();
id = eXosip_register_build_initial_register("sip:1094@10.232.19.204<sip%3A1094@10.232.19.204>
","10.232.19.204:5060",NULL,1800,®);
printf("%d\n", id);
if(id < 0)
{
eXosip_unlock();
printf("\n unable to build register request\n");
return -1;
}
else
{
printf("Register request built sucessfully\n");
}
i = eXosip_register_build_register(id,1800,reg);
osip_message_set_supported(reg,"100rel");
osip_message_set_supported(reg,"path");
i=eXosip_register_send_register(id,reg);
return -1;
}
Error message i am getting is
Initilazed.....
Transport layer initilazed ......................
0
1
Register request built sucessfully
*** glibc detected *** ./init: double free or corruption (out): 0x097f4cf8
***
======= Backtrace: =========
/lib/libc.so.6[0x166efd]
/lib/libc.so.6(cfree+0x90)[0x16a550]
/usr/lib/libosipparser2.so.3(osip_message_free+0x5c)[0x401f6d]
/usr/lib/libeXosip2.so.4(eXosip_register_send_register+0x112)[0x3acd97]
./init[0x8048772]
/lib/libc.so.6(__libc_start_main+0xdc)[0x116f2c]
./init[0x8048581]
======= Memory map: ========
00101000-00238000 r-xp 00000000 03:01 391121 /lib/libc-2.5.so
00238000-0023a000 r-xp 00137000 03:01 391121 /lib/libc-2.5.so
0023a000-0023b000 rwxp 00139000 03:01 391121 /lib/libc-2.5.so
0023b000-0023e000 rwxp 0023b000 00:00 0
0023e000-00269000 r-xp 00000000 03:01 516459
/usr/lib/libosipparser2.so.2.2.0
00269000-0026a000 rwxp 0002b000 03:01 516459
/usr/lib/libosipparser2.so.2.2.0
002f3000-002fa000 r-xp 00000000 03:01 391124 /lib/librt-2.5.so
002fa000-002fb000 r-xp 00006000 03:01 391124 /lib/librt-2.5.so
002fb000-002fc000 rwxp 00007000 03:01 391124 /lib/librt-2.5
With regards
Prasad
9900782382
|
|
From: Whit T. <de...@wh...> - 2008-05-20 16:45:14
Attachments:
b2bulogsnip..txt
|
Hey list members,
I have a couple of questions on a simple configuration with OpenSBC that I'm
having trouble proxying the media stream. I've read some other posts on this
topic, but nothing has help so far. Here is the setup:
Office SIP Phone <192.168.0.100>
|
[192.168.0.0] Internal Office #1 network
----------------------
| Office #1 LAN Router |
----------------------
[25.x.x.x] External IP address of office #1
|
|
INTERNET
|
|
[44.x.x.x] External IP of Office #2
----------------------
| Office #2 LAN Router |
----------------------
[192.168.1.0] Internal Office#2 network
|
|
OpenSBC <192.168.1.100>
|
|
Asterisk<192.168.1.101>
The phone in Office #1 is registering to the Asterisk box in Office #2 using
the UpperRegistration of OpenSBC which is in 'Full Mode'.
I am able to register through OpenSBC with the Asterisk box from Office #1,
however I am not getting any media to get proxied via OpenSBC. I've selected
the 'Proxy-All-Media' but nothing seems to work.
I must be missing something simple in the configuration. Where should I
focus my attention? In the B2BUA routes? Proxy-Relay-Routes?
While debugging the SIP messages on the Asterisk output I get the following
SIP message while launching a call:
//---------------------------
// from Asterisk CLI
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.100:5060;iid=7375;branch=z9hG4bK74bda256ea24dd1184caf507b23fcaa4-9
4d4d1526a6e1a341bd360c9d23def44;uas-addr=192.168.1.101;received=192.168.1.10
0;rport=5060
Via: SIP/2.0/UDP
192.168.0.100:61798;branch=z9hG4bK-d87543-77271a19797af374-1--d87543-;rport=
61798;received=25.X.X.X
Record-Route: <sip:192.168.1.100:5060;route-shift=192.168.1.100:5060;lr>
From: "7101" <sip:71...@in...>;tag=114af83a
To: "5555" <sip:55...@in...>;tag=as262aed0b
Call-ID: ZTMxZTgzMzFiNDQ0Njk1ODFjNWYxNGJlZDhlMmFhMzY.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:5555@192.168.1.101>
Content-Type: application/sdp
Content-Length: 268
v=0
o=root 22965 22965 IN IP4 192.168.1.101
s=session
c=IN IP4 192.168.1.101
t=0 0
m=audio 14662 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
//-------------------------------
internalsip.com is an internal mapped DNS to the asterisk box:
[sip:internalsip.com] sip:192.168.1.101:5060
Debugging Asterisk RTP it shows that its trying to send RTP to 192.168.0.100
which is the NAT'd address of the office phone in Office #1 which doesn't
make sense to me since obviously it can't reach that network.
//-----------------
// from Asterisk CLI
Sent RTP packet to 192.168.0.100:45976 (type 00, seq 016083, ts 056160,
len 000160)
Sent RTP packet to 192.168.0.100:45976 (type 00, seq 016084, ts 056320,
len 000160)
Sent RTP packet to 192.168.0.100:45976 (type 00, seq 016085, ts 056480,
len 000160)
//------------------
I feel that I'm 98% there... Does anyone have an idea on where to focus my
efforts?
I've attached a b2bualog snip as well in case this helps.
Best Regards,
Whit
|
|
From: Joegen E. B. <joe...@gm...> - 2008-05-20 23:30:34
|
Whit, Have you tried B2BUpperReg mode instead of full mode? The logs indicate that your invite is relayed (proxy) instead of using the B2BUA. Relay only call does not spawn the RTP Proxy. Joegen Whit Thiele wrote: > > > > > Hey list members, > > > > I have a couple of questions on a simple configuration with OpenSBC that I'm > having trouble proxying the media stream. I've read some other posts on this > topic, but nothing has help so far. Here is the setup: > > > > Office SIP Phone <192.168.0.100> > > | > > [192.168.0.0] Internal Office #1 network > > ---------------------- > > | Office #1 LAN Router | > > ---------------------- > > [25.x.x.x] External IP address of office #1 > > | > > | > > INTERNET > > | > > | > > > > [44.x.x.x] External IP of Office #2 > > ---------------------- > > | Office #2 LAN Router | > > ---------------------- > > [192.168.1.0] Internal Office#2 network > > | > > | > > > > OpenSBC <192.168.1.100> > > | > > | > > Asterisk<192.168.1.101> > > > > > > > > The phone in Office #1 is registering to the Asterisk box in Office #2 using > the UpperRegistration of OpenSBC which is in 'Full Mode'. > > > > > > I am able to register through OpenSBC with the Asterisk box from Office #1, > however I am not getting any media to get proxied via OpenSBC. I've selected > the 'Proxy-All-Media' but nothing seems to work. > > > > I must be missing something simple in the configuration. Where should I > focus my attention? In the B2BUA routes? Proxy-Relay-Routes? > > > > While debugging the SIP messages on the Asterisk output I get the following > SIP message while launching a call: > > > > > > //--------------------------- > > // from Asterisk CLI > > > > SIP/2.0 200 OK > > Via: SIP/2.0/UDP > 192.168.1.100:5060;iid=7375;branch=z9hG4bK74bda256ea24dd1184caf507b23fcaa4-9 > 4d4d1526a6e1a341bd360c9d23def44;uas-addr=192.168.1.101;received=192.168.1.10 > 0;rport=5060 > > Via: SIP/2.0/UDP > 192.168.0.100:61798;branch=z9hG4bK-d87543-77271a19797af374-1--d87543-;rport= > 61798;received=25.X.X.X > > Record-Route: <sip:192.168.1.100:5060;route-shift=192.168.1.100:5060;lr> > > From: "7101" <sip:71...@in...>;tag=114af83a > > To: "5555" <sip:55...@in...>;tag=as262aed0b > > Call-ID: ZTMxZTgzMzFiNDQ0Njk1ODFjNWYxNGJlZDhlMmFhMzY. > > CSeq: 2 INVITE > > User-Agent: Asterisk PBX > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > > Supported: replaces > > Contact: <sip:5555@192.168.1.101> > > Content-Type: application/sdp > > Content-Length: 268 > > > > v=0 > > o=root 22965 22965 IN IP4 192.168.1.101 > > s=session > > c=IN IP4 192.168.1.101 > > t=0 0 > > m=audio 14662 RTP/AVP 0 8 101 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=silenceSupp:off - - - - > > a=ptime:20 > > a=sendrecv > > //------------------------------- > > > > internalsip.com is an internal mapped DNS to the asterisk box: > > [sip:internalsip.com] sip:192.168.1.101:5060 > > > > > > Debugging Asterisk RTP it shows that its trying to send RTP to 192.168.0.100 > which is the NAT'd address of the office phone in Office #1 which doesn't > make sense to me since obviously it can't reach that network. > > > > > > //----------------- > > // from Asterisk CLI > > > > Sent RTP packet to 192.168.0.100:45976 (type 00, seq 016083, ts 056160, > len 000160) > > Sent RTP packet to 192.168.0.100:45976 (type 00, seq 016084, ts 056320, > len 000160) > > Sent RTP packet to 192.168.0.100:45976 (type 00, seq 016085, ts 056480, > len 000160) > > //------------------ > > > > > > I feel that I'm 98% there... Does anyone have an idea on where to focus my > efforts? > > > > I've attached a b2bualog snip as well in case this helps. > > > > Best Regards, > > > > Whit > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2008. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > ------------------------------------------------------------------------ > > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > ------------------------------------------------------------------------ > > No virus found in this incoming message. > Checked by AVG. > Version: 7.5.524 / Virus Database: 269.23.21/1454 - Release Date: 5/19/2008 7:44 AM |
|
From: Craig G. <cra...@gm...> - 2008-05-20 23:45:18
|
Hi Whit,
Are the offices VPN or just double natted? If natted then your OpenSBC is
in the wrong place - it needs to have two interfaces - one on the internal
LAN and the other with a public facing address.
Alternately you could omit OpenSBC and port map an external IP address to
your asterisk box and set the 'externip' in sip.conf to your external port
mapped address and add the appropriate value to 'localnet'
Probably easier to create an inter office VPN IMHO.
Craig
-----Original Message-----
From: ope...@li...
[mailto:ope...@li...] On Behalf Of Whit
Thiele
Sent: Wednesday, 21 May 2008 1:45 AM
To: ope...@li...
Subject: [OpenSIPStack] OpenSBC Media Proxy Problem
Hey list members,
I have a couple of questions on a simple configuration with OpenSBC that I'm
having trouble proxying the media stream. I've read some other posts on this
topic, but nothing has help so far. Here is the setup:
Office SIP Phone <192.168.0.100>
|
[192.168.0.0] Internal Office #1 network
----------------------
| Office #1 LAN Router |
----------------------
[25.x.x.x] External IP address of office #1
|
|
INTERNET
|
|
[44.x.x.x] External IP of Office #2
----------------------
| Office #2 LAN Router |
----------------------
[192.168.1.0] Internal Office#2 network
|
|
OpenSBC <192.168.1.100>
|
|
Asterisk<192.168.1.101>
The phone in Office #1 is registering to the Asterisk box in Office #2 using
the UpperRegistration of OpenSBC which is in 'Full Mode'.
I am able to register through OpenSBC with the Asterisk box from Office #1,
however I am not getting any media to get proxied via OpenSBC. I've selected
the 'Proxy-All-Media' but nothing seems to work.
I must be missing something simple in the configuration. Where should I
focus my attention? In the B2BUA routes? Proxy-Relay-Routes?
While debugging the SIP messages on the Asterisk output I get the following
SIP message while launching a call:
//---------------------------
// from Asterisk CLI
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.100:5060;iid=7375;branch=z9hG4bK74bda256ea24dd1184caf507b23fcaa4-9
4d4d1526a6e1a341bd360c9d23def44;uas-addr=192.168.1.101;received=192.168.1.10
0;rport=5060
Via: SIP/2.0/UDP
192.168.0.100:61798;branch=z9hG4bK-d87543-77271a19797af374-1--d87543-;rport=
61798;received=25.X.X.X
Record-Route: <sip:192.168.1.100:5060;route-shift=192.168.1.100:5060;lr>
From: "7101" <sip:71...@in...>;tag=114af83a
To: "5555" <sip:55...@in...>;tag=as262aed0b
Call-ID: ZTMxZTgzMzFiNDQ0Njk1ODFjNWYxNGJlZDhlMmFhMzY.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:5555@192.168.1.101>
Content-Type: application/sdp
Content-Length: 268
v=0
o=root 22965 22965 IN IP4 192.168.1.101
s=session
c=IN IP4 192.168.1.101
t=0 0
m=audio 14662 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
//-------------------------------
internalsip.com is an internal mapped DNS to the asterisk box:
[sip:internalsip.com] sip:192.168.1.101:5060
Debugging Asterisk RTP it shows that its trying to send RTP to 192.168.0.100
which is the NAT'd address of the office phone in Office #1 which doesn't
make sense to me since obviously it can't reach that network.
//-----------------
// from Asterisk CLI
Sent RTP packet to 192.168.0.100:45976 (type 00, seq 016083, ts 056160,
len 000160)
Sent RTP packet to 192.168.0.100:45976 (type 00, seq 016084, ts 056320,
len 000160)
Sent RTP packet to 192.168.0.100:45976 (type 00, seq 016085, ts 056480,
len 000160)
//------------------
I feel that I'm 98% there... Does anyone have an idea on where to focus my
efforts?
I've attached a b2bualog snip as well in case this helps.
Best Regards,
Whit
|
|
From: Joegen E. B. <joe...@gm...> - 2008-05-20 23:36:06
|
Prasad,
I think you might have confused the osip project with OpenSIPStack?
These are two different projects :-). This is the mailing list for
OpenSIPStack, not osip. However, looking at your code, is the call
eXosip_register_send_register() synchronous? If it is not, then the app
might exit and destroy resources while the osip transaction layer is
still performing its tasks. This is just a hunch. Try putting a
sleep() right after the function and see if that changes anything.
Lastly, please find the correct list for osip and post your questions
there. I am sure you will have better chances in getting the correct
answers there.
Joegen
prasad keshav wrote:
> Hi all,ope...@li...
>
>
> i am using eXosip as sip stack , i am able to initilaze the osip and exosip
> and the problem with eXosip _send_register ,where iam not able to fix up the
> bug can any bodyhelp me
>
> #include<eXosip2/eXosip.h>
> #include<sys/socket.h>
> #include<netinet/in.h>
> #include<arpa/inet.h>
> main() {
> int i;
> int id;
> osip_message_t* reg = NULL;
> TRACE_INITIALIZE(6,stdout);
> i = eXosip_init();
> if(i !=0)
> {
> printf("Not Initilazed\n");
> return -1;
> }
> else
> {
> printf("\nInitilazed.....\n");
> }
> i = eXosip_listen_addr(IPPROTO_UDP,"10.232.19.204
> ",10234,AF_INET,0);
> if(i != 0)
> {
> eXosip_quit();
> printf("Could not initilaize transport layer\n");
> return -1;
> }
> else
> {
> printf("\nTransport layer initilazed
> ......................\n");
> printf("%d\n", i);
> }
> eXosip_lock();
> id = eXosip_register_build_initial_register("sip:1094@10.232.19.204<sip%3A1094@10.232.19.204>
> ","10.232.19.204:5060",NULL,1800,®);
> printf("%d\n", id);
> if(id < 0)
> {
> eXosip_unlock();
> printf("\n unable to build register request\n");
> return -1;
> }
> else
> {
> printf("Register request built sucessfully\n");
> }
> i = eXosip_register_build_register(id,1800,reg);
> osip_message_set_supported(reg,"100rel");
> osip_message_set_supported(reg,"path");
> i=eXosip_register_send_register(id,reg);
> return -1;
> }
>
> Error message i am getting is
>
>
> Initilazed.....
>
> Transport layer initilazed ......................
> 0
> 1
> Register request built sucessfully
> *** glibc detected *** ./init: double free or corruption (out): 0x097f4cf8
> ***
> ======= Backtrace: =========
> /lib/libc.so.6[0x166efd]
> /lib/libc.so.6(cfree+0x90)[0x16a550]
> /usr/lib/libosipparser2.so.3(osip_message_free+0x5c)[0x401f6d]
> /usr/lib/libeXosip2.so.4(eXosip_register_send_register+0x112)[0x3acd97]
> ./init[0x8048772]
> /lib/libc.so.6(__libc_start_main+0xdc)[0x116f2c]
> ./init[0x8048581]
> ======= Memory map: ========
> 00101000-00238000 r-xp 00000000 03:01 391121 /lib/libc-2.5.so
> 00238000-0023a000 r-xp 00137000 03:01 391121 /lib/libc-2.5.so
> 0023a000-0023b000 rwxp 00139000 03:01 391121 /lib/libc-2.5.so
> 0023b000-0023e000 rwxp 0023b000 00:00 0
> 0023e000-00269000 r-xp 00000000 03:01 516459
> /usr/lib/libosipparser2.so.2.2.0
> 00269000-0026a000 rwxp 0002b000 03:01 516459
> /usr/lib/libosipparser2.so.2.2.0
> 002f3000-002fa000 r-xp 00000000 03:01 391124 /lib/librt-2.5.so
> 002fa000-002fb000 r-xp 00006000 03:01 391124 /lib/librt-2.5.so
> 002fb000-002fc000 rwxp 00007000 03:01 391124 /lib/librt-2.5
> With regards
> Prasad
> 9900782382
> -------------------------------------------------------------------------
> This SF.net email is sponsored by: Microsoft
> Defy all challenges. Microsoft(R) Visual Studio 2008.
> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/
> _______________________________________________
> opensipstack-devel mailing list
> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>
>
|
|
From: Whit T. <de...@wh...> - 2008-05-21 13:00:15
|
Hey, Thanks for the idea! I solved my problem however by starting from scratch (even redownloading from cvs and deleting the ini file) then I set OpenSBC to B2BUAUpperReg mode along with the Always Proxy Media selection. I am going to document these steps more clearly and perhaps they will make it onto the site. A quick question though about linking this all together. In the Manual, the quickstart sample shows how to set up SJphone with the domain opensbcrocks.org which is an internal DNS mapping in OpenSBC. Where in ATLSIP would you set this domain when communicating with OpenSBC? Whit -----Original Message----- From: ope...@li... [mailto:ope...@li...] On Behalf Of Craig Guy Sent: Tuesday, May 20, 2008 7:45 PM To: ope...@li... Subject: Re: [OpenSIPStack] OpenSBC Media Proxy Problem Hi Whit, Are the offices VPN or just double natted? If natted then your OpenSBC is in the wrong place - it needs to have two interfaces - one on the internal LAN and the other with a public facing address. Alternately you could omit OpenSBC and port map an external IP address to your asterisk box and set the 'externip' in sip.conf to your external port mapped address and add the appropriate value to 'localnet' Probably easier to create an inter office VPN IMHO. Craig -----Original Message----- From: ope...@li... [mailto:ope...@li...] On Behalf Of Whit Thiele Sent: Wednesday, 21 May 2008 1:45 AM To: ope...@li... Subject: [OpenSIPStack] OpenSBC Media Proxy Problem Hey list members, I have a couple of questions on a simple configuration with OpenSBC that I'm having trouble proxying the media stream. I've read some other posts on this topic, but nothing has help so far. Here is the setup: Office SIP Phone <192.168.0.100> | [192.168.0.0] Internal Office #1 network ---------------------- | Office #1 LAN Router | ---------------------- [25.x.x.x] External IP address of office #1 | | INTERNET | | [44.x.x.x] External IP of Office #2 ---------------------- | Office #2 LAN Router | ---------------------- [192.168.1.0] Internal Office#2 network | | OpenSBC <192.168.1.100> | | Asterisk<192.168.1.101> The phone in Office #1 is registering to the Asterisk box in Office #2 using the UpperRegistration of OpenSBC which is in 'Full Mode'. I am able to register through OpenSBC with the Asterisk box from Office #1, however I am not getting any media to get proxied via OpenSBC. I've selected the 'Proxy-All-Media' but nothing seems to work. I must be missing something simple in the configuration. Where should I focus my attention? In the B2BUA routes? Proxy-Relay-Routes? While debugging the SIP messages on the Asterisk output I get the following SIP message while launching a call: //--------------------------- // from Asterisk CLI SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.100:5060;iid=7375;branch=z9hG4bK74bda256ea24dd1184caf507b23fcaa4-9 4d4d1526a6e1a341bd360c9d23def44;uas-addr=192.168.1.101;received=192.168.1.10 0;rport=5060 Via: SIP/2.0/UDP 192.168.0.100:61798;branch=z9hG4bK-d87543-77271a19797af374-1--d87543-;rport= 61798;received=25.X.X.X Record-Route: <sip:192.168.1.100:5060;route-shift=192.168.1.100:5060;lr> From: "7101" <sip:71...@in...>;tag=114af83a To: "5555" <sip:55...@in...>;tag=as262aed0b Call-ID: ZTMxZTgzMzFiNDQ0Njk1ODFjNWYxNGJlZDhlMmFhMzY. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:5555@192.168.1.101> Content-Type: application/sdp Content-Length: 268 v=0 o=root 22965 22965 IN IP4 192.168.1.101 s=session c=IN IP4 192.168.1.101 t=0 0 m=audio 14662 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv //------------------------------- internalsip.com is an internal mapped DNS to the asterisk box: [sip:internalsip.com] sip:192.168.1.101:5060 Debugging Asterisk RTP it shows that its trying to send RTP to 192.168.0.100 which is the NAT'd address of the office phone in Office #1 which doesn't make sense to me since obviously it can't reach that network. //----------------- // from Asterisk CLI Sent RTP packet to 192.168.0.100:45976 (type 00, seq 016083, ts 056160, len 000160) Sent RTP packet to 192.168.0.100:45976 (type 00, seq 016084, ts 056320, len 000160) Sent RTP packet to 192.168.0.100:45976 (type 00, seq 016085, ts 056480, len 000160) //------------------ I feel that I'm 98% there... Does anyone have an idea on where to focus my efforts? I've attached a b2bualog snip as well in case this helps. Best Regards, Whit ------------------------------------------------------------------------- This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ _______________________________________________ opensipstack-devel mailing list ope...@li... https://lists.sourceforge.net/lists/listinfo/opensipstack-devel No virus found in this incoming message. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.21/1458 - Release Date: 5/21/2008 7:21 AM |
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