Re: [SR-Users] CANCEL not matching INVITES !

2011-12-02 Thread Daniel-Constantin Mierla 

Hello,

On 12/2/11 5:24 AM, Sammy Govind wrote:

Hello again,

You were right, as soon as I made changes in asterisk SIP profile for 
the Kamailio proxy server and stopped the 401 Auth from Asterisk to 
Kamailio the CANCELS started to work fine.
well, the 401 from asterisk is ok from specs point of view (although 
many phones don't work with many challenges), but this case revealed 
some bugs in asterisk as well as in xlite, both of them had misbehavior.


Cheers,
Daniel



So the SIP flow now is:

- invite from phone to kamailio
- kamailio asks for authentication - 407
- ack
- invite with credentials, kamailio forwards to asterisk
- asterisk starts processing the invite and call can be cancelled now.


Thanks alot

--

Best Regards,
Sammy.

On Thu, Dec 1, 2011 at 12:01 PM, Sammy Govind govoi...@gmail.com 
mailto:govoi...@gmail.com wrote:


Hey Daniel,

I've exactly followed your point, I'll try some stuff on asterisk
server to stop asking for 401 Auth to Kamailio., maybe this will
eliminate the need for another INVITE with authentication params.

But one thing which just makes me curious is that a soft phone
directly coming from a Public IP is always able to successfully
CANCEL the call.

Anyway I'll use some brain of mine on this and let you know what
resolved it, or what I'm missing.

Thanks,
Sammy


On Wed, Nov 30, 2011 at 5:47 PM, Daniel-Constantin Mierla
mico...@gmail.com mailto:mico...@gmail.com wrote:

Hello,

is the SIP trace complete?

What I could find inside is:
- invite from phone to kamailio
- kamailio asks for authentication - 407
- ack
- invite with credentials, kamailio forwards to asterisk
- asterisk asks for authentication - 401
- ack
- there is no new INVITE with credentials for kamailio and
asterisk
- but the phone starts sending CANCELs -- since there is no
active INVITE transaction, kamailio just drops it due to
config rules
- after a while asterisk starts sending like 180 ringing, then
200ok ... really strange

Maybe you haven't captured all the sip traffic. If you want to
use ngrep, do on kamailio server:


ngrep -d any -qt -W byline port 5060

If that's all the traffic, then xlite and asterisk seems to
have some bugs - both were aware of 401 reply (asterisk
generated it, xlite sent the ACK for it) -- so no ongoing call
to CANCEL by xlite, or to answer by Asterisk (the 180, 200
replies).

From kamailio point of view, if there is no INVITE following
the 401 reply to xlite, there is no active invite transaction
to cancel.

Cheers,
Daniel


On 11/30/11 12:02 AM, Daniel-Constantin Mierla wrote:

Hello,

I will look over it soon - since you sent pcap I couldn't
look at it directly from the email. ngrep outputs plain text
which is easy to read from email, the reason I am asking
mainly for ngrep traces since many times I am not around a
computer where is convenient to open pcap file. On the other
hand, if it is a transmission problem (at transport layer),
pcap file is better.

Cheers,
Daniel

On 11/29/11 5:07 AM, Sammy Govind wrote:

Hello again,

Please see the attached wireshark trace, I tried for a
sipgrep trace but couldn't somehow. I hope this will get me
some clue on what I'm doing wrong.

This is a setup with Kamailio in front of Asterisk Servers.
Kamailio is multihomed and MS are on private IPs, all the
calls are routed to MSs and then comeback for further dial-outs.

Please see the Continuous CANCEL requests which aren't
terminating the call.

Thanks,
Sammy.

On Mon, Nov 28, 2011 at 4:41 PM, Sammy Govind
govoi...@gmail.com mailto:govoi...@gmail.com wrote:

Thanks for your reply I will attach the wireshark traces
as soon as I get to my workstation.

BR,
Sammy.


On Mon, Nov 28, 2011 at 3:33 PM, Daniel-Constantin
Mierla mico...@gmail.com mailto:mico...@gmail.com wrote:

Hello,

send the ngrep trace of such call, from the initial
INVITE, you can use:

ngrep -d any -qt -W byline port 5060

The sip trace will help to see what is wrong with
that CANCEL.

Cheers,
Daniel


On 11/28/11 7:19 AM, Sammy Govind wrote:

Anyone please help.

On Sat, Nov 26, 2011 at 10:39 PM, Sammy Govind
govoi...@gmail.com mailto:govoi...@gmail.com wrote:

Hello list,

I'm using Kamailio 3.1.5 in front of asterisk

Re: [SR-Users] CANCEL not matching INVITES !

2011-12-01 Thread Sammy Govind 
Hello again,

You were right, as soon as I made changes in asterisk SIP profile for the
Kamailio proxy server and stopped the 401 Auth from Asterisk to Kamailio
the CANCELS started to work fine.

So the SIP flow now is:

- invite from phone to kamailio
- kamailio asks for authentication - 407
- ack
- invite with credentials, kamailio forwards to asterisk
- asterisk starts processing the invite and call can be cancelled now.


Thanks alot

--

Best Regards,
Sammy.

On Thu, Dec 1, 2011 at 12:01 PM, Sammy Govind govoi...@gmail.com wrote:

 Hey Daniel,

 I've exactly followed your point, I'll try some stuff on asterisk server
 to stop asking for 401 Auth to Kamailio., maybe this will eliminate the
 need for another INVITE with authentication params.

 But one thing which just makes me curious is that a soft phone directly
 coming from a Public IP is always able to successfully CANCEL the call.

 Anyway I'll use some brain of mine on this and let you know what resolved
 it, or what I'm missing.

 Thanks,
 Sammy


 On Wed, Nov 30, 2011 at 5:47 PM, Daniel-Constantin Mierla 
 mico...@gmail.com wrote:

  Hello,

 is the SIP trace complete?

 What I could find inside is:
 - invite from phone to kamailio
 - kamailio asks for authentication - 407
 - ack
 - invite with credentials, kamailio forwards to asterisk
 - asterisk asks for authentication - 401
 - ack
 - there is no new INVITE with credentials for kamailio and asterisk
 - but the phone starts sending CANCELs -- since there is no active INVITE
 transaction, kamailio just drops it due to config rules
 - after a while asterisk starts sending like 180 ringing, then 200ok ...
 really strange

 Maybe you haven't captured all the sip traffic. If you want to use ngrep,
 do on kamailio server:


 ngrep -d any -qt -W byline port 5060

 If that's all the traffic, then xlite and asterisk seems to have some
 bugs - both were aware of 401 reply (asterisk generated it, xlite sent the
 ACK for it) -- so no ongoing call to CANCEL by xlite, or to answer by
 Asterisk (the 180, 200 replies).

 From kamailio point of view, if there is no INVITE following the 401
 reply to xlite, there is no active invite transaction to cancel.

 Cheers,
 Daniel


 On 11/30/11 12:02 AM, Daniel-Constantin Mierla wrote:

 Hello,

 I will look over it soon - since you sent pcap I couldn't look at it
 directly from the email. ngrep outputs plain text which is easy to read
 from email, the reason I am asking mainly for ngrep traces since many times
 I am not around a computer where is convenient to open pcap file. On the
 other hand, if it is a transmission problem (at transport layer), pcap file
 is better.

 Cheers,
 Daniel

 On 11/29/11 5:07 AM, Sammy Govind wrote:

 Hello again,

  Please see the attached wireshark trace, I tried for a sipgrep trace
 but couldn't somehow. I hope this will get me some clue on what I'm doing
 wrong.

  This is a setup with Kamailio in front of Asterisk Servers. Kamailio is
 multihomed and MS are on private IPs, all the calls are routed to MSs and
 then comeback for further dial-outs.

  Please see the Continuous CANCEL requests which aren't terminating the
 call.

  Thanks,
 Sammy.

  On Mon, Nov 28, 2011 at 4:41 PM, Sammy Govind govoi...@gmail.comwrote:

 Thanks for your reply I will attach the wireshark traces as soon as I
 get to my workstation.

  BR,
 Sammy.


 On Mon, Nov 28, 2011 at 3:33 PM, Daniel-Constantin Mierla 
 mico...@gmail.com wrote:

  Hello,

 send the ngrep trace of such call, from the initial INVITE, you can use:

 ngrep -d any -qt -W byline port 5060

 The sip trace will help to see what is wrong with that CANCEL.

 Cheers,
 Daniel


 On 11/28/11 7:19 AM, Sammy Govind wrote:

  Anyone please help.

 On Sat, Nov 26, 2011 at 10:39 PM, Sammy Govind govoi...@gmail.comwrote:

 Hello list,

  I'm using Kamailio 3.1.5 in front of asterisk servers. Kamailio
 handles all the SIP registrations. Calls from SIP phones are forwarded to
 asterisks and then dialled out to Kamailio.

  root@SBCserver:~# kamailio -V
 version: kamailio 3.1.5 (x86_64/linux) 76fff5
 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS,
 USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP,
 PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT,
 USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST,
 HAVE_RESOLV_RES
 ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
 MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4MB
 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
 id: 76fff5
 compiled on 08:21:33 Oct 27 2011 with gcc 4.6.1
 root@SBCserver:~#


  Problem:
 When call is initiated from a softphone and is in ringing phase,
 CANCEL just don't work. I've done some initial debugging and
 the following piece of code in main route is failing.

  # CANCEL processing
 if (is_method(CANCEL))
 {
  xlog(L_NOTICE,$rm from $fu (IP:$si:$sp) ---CAPTURED IN
 MAIN---\n);
  if (t_check_trans()){
 t_relay();

Re: [SR-Users] CANCEL not matching INVITES !

2011-11-30 Thread Daniel-Constantin Mierla 

Hello,

is the SIP trace complete?

What I could find inside is:
- invite from phone to kamailio
- kamailio asks for authentication - 407
- ack
- invite with credentials, kamailio forwards to asterisk
- asterisk asks for authentication - 401
- ack
- there is no new INVITE with credentials for kamailio and asterisk
- but the phone starts sending CANCELs -- since there is no active 
INVITE transaction, kamailio just drops it due to config rules
- after a while asterisk starts sending like 180 ringing, then 200ok ... 
really strange


Maybe you haven't captured all the sip traffic. If you want to use 
ngrep, do on kamailio server:


ngrep -d any -qt -W byline port 5060

If that's all the traffic, then xlite and asterisk seems to have some 
bugs - both were aware of 401 reply (asterisk generated it, xlite sent 
the ACK for it) -- so no ongoing call to CANCEL by xlite, or to answer 
by Asterisk (the 180, 200 replies).


From kamailio point of view, if there is no INVITE following the 401 
reply to xlite, there is no active invite transaction to cancel.


Cheers,
Daniel

On 11/30/11 12:02 AM, Daniel-Constantin Mierla wrote:

Hello,

I will look over it soon - since you sent pcap I couldn't look at it 
directly from the email. ngrep outputs plain text which is easy to 
read from email, the reason I am asking mainly for ngrep traces since 
many times I am not around a computer where is convenient to open pcap 
file. On the other hand, if it is a transmission problem (at transport 
layer), pcap file is better.


Cheers,
Daniel

On 11/29/11 5:07 AM, Sammy Govind wrote:

Hello again,

Please see the attached wireshark trace, I tried for a sipgrep trace 
but couldn't somehow. I hope this will get me some clue on what I'm 
doing wrong.


This is a setup with Kamailio in front of Asterisk Servers. Kamailio 
is multihomed and MS are on private IPs, all the calls are routed to 
MSs and then comeback for further dial-outs.


Please see the Continuous CANCEL requests which aren't terminating 
the call.


Thanks,
Sammy.

On Mon, Nov 28, 2011 at 4:41 PM, Sammy Govind govoi...@gmail.com 
mailto:govoi...@gmail.com wrote:


Thanks for your reply I will attach the wireshark traces as soon
as I get to my workstation.

BR,
Sammy.


On Mon, Nov 28, 2011 at 3:33 PM, Daniel-Constantin Mierla
mico...@gmail.com mailto:mico...@gmail.com wrote:

Hello,

send the ngrep trace of such call, from the initial INVITE,
you can use:

ngrep -d any -qt -W byline port 5060

The sip trace will help to see what is wrong with that CANCEL.

Cheers,
Daniel


On 11/28/11 7:19 AM, Sammy Govind wrote:

Anyone please help.

On Sat, Nov 26, 2011 at 10:39 PM, Sammy Govind
govoi...@gmail.com mailto:govoi...@gmail.com wrote:

Hello list,

I'm using Kamailio 3.1.5 in front of asterisk servers.
Kamailio handles all the SIP registrations. Calls from
SIP phones are forwarded to asterisks and then dialled
out to Kamailio.

root@SBCserver:~# kamailio -V
version: kamailio 3.1.5 (x86_64/linux) 76fff5
flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS,
TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST,
DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT,
USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR,
USE_DST_BLACKLIST, HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144,
MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535,
PKG_SIZE 4MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt,
select.
id: 76fff5
compiled on 08:21:33 Oct 27 2011 with gcc 4.6.1
root@SBCserver:~#


Problem:
When call is initiated from a softphone and is in
ringing phase, CANCEL just don't work. I've done some
initial debugging and the following piece of code in
main route is failing.

# CANCEL processing
if (is_method(CANCEL))
{
 xlog(L_NOTICE,$rm from $fu (IP:$si:$sp)
---CAPTURED IN MAIN---\n);
 if (t_check_trans()){
t_relay();
xlog(L_NOTICE,$rm from $fu (IP:$si:$sp)
---CHECK TRANS TRUE---\n);
 }
 xlog(L_NOTICE,$rm from $fu (IP:$si:$sp) ---CHECK
TRANS FALSE---\n);
 exit;
}

Also the CANCEL fails the has_totag() condition !

The same Call CANCEL scenario works fine for any client
on Public IP !

Hope to get some pointers for the solution.

Regards,
Sammy.




___
SIP Express Router (SER) and

Re: [SR-Users] CANCEL not matching INVITES !

2011-11-29 Thread Daniel-Constantin Mierla 

Hello,

I will look over it soon - since you sent pcap I couldn't look at it 
directly from the email. ngrep outputs plain text which is easy to read 
from email, the reason I am asking mainly for ngrep traces since many 
times I am not around a computer where is convenient to open pcap file. 
On the other hand, if it is a transmission problem (at transport layer), 
pcap file is better.


Cheers,
Daniel

On 11/29/11 5:07 AM, Sammy Govind wrote:

Hello again,

Please see the attached wireshark trace, I tried for a sipgrep trace 
but couldn't somehow. I hope this will get me some clue on what I'm 
doing wrong.


This is a setup with Kamailio in front of Asterisk Servers. Kamailio 
is multihomed and MS are on private IPs, all the calls are routed to 
MSs and then comeback for further dial-outs.


Please see the Continuous CANCEL requests which aren't terminating the 
call.


Thanks,
Sammy.

On Mon, Nov 28, 2011 at 4:41 PM, Sammy Govind govoi...@gmail.com 
mailto:govoi...@gmail.com wrote:


Thanks for your reply I will attach the wireshark traces as soon
as I get to my workstation.

BR,
Sammy.


On Mon, Nov 28, 2011 at 3:33 PM, Daniel-Constantin Mierla
mico...@gmail.com mailto:mico...@gmail.com wrote:

Hello,

send the ngrep trace of such call, from the initial INVITE,
you can use:

ngrep -d any -qt -W byline port 5060

The sip trace will help to see what is wrong with that CANCEL.

Cheers,
Daniel


On 11/28/11 7:19 AM, Sammy Govind wrote:

Anyone please help.

On Sat, Nov 26, 2011 at 10:39 PM, Sammy Govind
govoi...@gmail.com mailto:govoi...@gmail.com wrote:

Hello list,

I'm using Kamailio 3.1.5 in front of asterisk servers.
Kamailio handles all the SIP registrations. Calls from
SIP phones are forwarded to asterisks and then dialled
out to Kamailio.

root@SBCserver:~# kamailio -V
version: kamailio 3.1.5 (x86_64/linux) 76fff5
flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS,
USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK,
SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX,
FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER,
USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144,
MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535,
PKG_SIZE 4MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt,
select.
id: 76fff5
compiled on 08:21:33 Oct 27 2011 with gcc 4.6.1
root@SBCserver:~#


Problem:
When call is initiated from a softphone and is in ringing
phase, CANCEL just don't work. I've done some initial
debugging and the following piece of code in main route
is failing.

# CANCEL processing
if (is_method(CANCEL))
{
 xlog(L_NOTICE,$rm from $fu (IP:$si:$sp)
---CAPTURED IN MAIN---\n);
 if (t_check_trans()){
t_relay();
xlog(L_NOTICE,$rm from $fu (IP:$si:$sp)
---CHECK TRANS TRUE---\n);
 }
 xlog(L_NOTICE,$rm from $fu (IP:$si:$sp) ---CHECK
TRANS FALSE---\n);
 exit;
}

Also the CANCEL fails the has_totag() condition !

The same Call CANCEL scenario works fine for any client
on Public IP !

Hope to get some pointers for the solution.

Regards,
Sammy.




___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org  mailto:sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


-- 
Daniel-Constantin Mierla --http://www.asipto.com

Kamailio Advanced Training, Dec 5-8, Berlin:http://asipto.com/u/kat
http://linkedin.com/in/miconda  -- http://twitter.com/miconda





___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


--
Daniel-Constantin Mierla -- http://www.asipto.com
Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat
http://linkedin.com/in/miconda -- http://twitter.com/miconda

___
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sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] CANCEL not matching INVITES !

2011-11-28 Thread Daniel-Constantin Mierla 

Hello,

send the ngrep trace of such call, from the initial INVITE, you can use:

ngrep -d any -qt -W byline port 5060

The sip trace will help to see what is wrong with that CANCEL.

Cheers,
Daniel

On 11/28/11 7:19 AM, Sammy Govind wrote:

Anyone please help.

On Sat, Nov 26, 2011 at 10:39 PM, Sammy Govind govoi...@gmail.com 
mailto:govoi...@gmail.com wrote:


Hello list,

I'm using Kamailio 3.1.5 in front of asterisk servers. Kamailio
handles all the SIP registrations. Calls from SIP phones are
forwarded to asterisks and then dialled out to Kamailio.

root@SBCserver:~# kamailio -V
version: kamailio 3.1.5 (x86_64/linux) 76fff5
flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS,
USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM,
SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX,
FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER,
USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN
16, MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: 76fff5
compiled on 08:21:33 Oct 27 2011 with gcc 4.6.1
root@SBCserver:~#


Problem:
When call is initiated from a softphone and is in ringing phase,
CANCEL just don't work. I've done some initial debugging and
the following piece of code in main route is failing.

# CANCEL processing
if (is_method(CANCEL))
{
 xlog(L_NOTICE,$rm from $fu (IP:$si:$sp) ---CAPTURED IN
MAIN---\n);
 if (t_check_trans()){
t_relay();
xlog(L_NOTICE,$rm from $fu (IP:$si:$sp) ---CHECK TRANS
TRUE---\n);
 }
 xlog(L_NOTICE,$rm from $fu (IP:$si:$sp) ---CHECK TRANS
FALSE---\n);
 exit;
}

Also the CANCEL fails the has_totag() condition !

The same Call CANCEL scenario works fine for any client on Public IP !

Hope to get some pointers for the solution.

Regards,
Sammy.




___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


--
Daniel-Constantin Mierla -- http://www.asipto.com
Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat
http://linkedin.com/in/miconda -- http://twitter.com/miconda

___
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sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] CANCEL not matching INVITES !

2011-11-28 Thread Sammy Govind 
Thanks for your reply I will attach the wireshark traces as soon as I get
to my workstation.

BR,
Sammy.

On Mon, Nov 28, 2011 at 3:33 PM, Daniel-Constantin Mierla mico...@gmail.com
 wrote:

  Hello,

 send the ngrep trace of such call, from the initial INVITE, you can use:

 ngrep -d any -qt -W byline port 5060

 The sip trace will help to see what is wrong with that CANCEL.

 Cheers,
 Daniel


 On 11/28/11 7:19 AM, Sammy Govind wrote:

 Anyone please help.

 On Sat, Nov 26, 2011 at 10:39 PM, Sammy Govind govoi...@gmail.com wrote:

 Hello list,

  I'm using Kamailio 3.1.5 in front of asterisk servers. Kamailio handles
 all the SIP registrations. Calls from SIP phones are forwarded to asterisks
 and then dialled out to Kamailio.

  root@SBCserver:~# kamailio -V
 version: kamailio 3.1.5 (x86_64/linux) 76fff5
 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS,
 DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
 DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE,
 USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
 ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
 MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4MB
 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
 id: 76fff5
 compiled on 08:21:33 Oct 27 2011 with gcc 4.6.1
 root@SBCserver:~#


  Problem:
 When call is initiated from a softphone and is in ringing phase, CANCEL
 just don't work. I've done some initial debugging and the following piece
 of code in main route is failing.

  # CANCEL processing
 if (is_method(CANCEL))
 {
  xlog(L_NOTICE,$rm from $fu (IP:$si:$sp) ---CAPTURED IN
 MAIN---\n);
  if (t_check_trans()){
 t_relay();
 xlog(L_NOTICE,$rm from $fu (IP:$si:$sp) ---CHECK TRANS
 TRUE---\n);
  }
  xlog(L_NOTICE,$rm from $fu (IP:$si:$sp) ---CHECK TRANS
 FALSE---\n);
  exit;
 }

  Also the CANCEL fails the has_totag() condition !

  The same Call CANCEL scenario works fine for any client on Public IP !

  Hope to get some pointers for the solution.

  Regards,
 Sammy.




 ___
 SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing 
 listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


 --
 Daniel-Constantin Mierla -- http://www.asipto.com
 Kamailio Advanced Training, Dec 5-8, Berlin: 
 http://asipto.com/u/kathttp://linkedin.com/in/miconda -- 
 http://twitter.com/miconda


___
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sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] CANCEL not matching INVITES !

2011-11-28 Thread Sammy Govind 
Hello again,

Please see the attached wireshark trace, I tried for a sipgrep trace but
couldn't somehow. I hope this will get me some clue on what I'm doing wrong.

This is a setup with Kamailio in front of Asterisk Servers. Kamailio is
multihomed and MS are on private IPs, all the calls are routed to MSs and
then comeback for further dial-outs.

Please see the Continuous CANCEL requests which aren't terminating the call.

Thanks,
Sammy.

On Mon, Nov 28, 2011 at 4:41 PM, Sammy Govind govoi...@gmail.com wrote:

 Thanks for your reply I will attach the wireshark traces as soon as I get
 to my workstation.

 BR,
 Sammy.


 On Mon, Nov 28, 2011 at 3:33 PM, Daniel-Constantin Mierla 
 mico...@gmail.com wrote:

  Hello,

 send the ngrep trace of such call, from the initial INVITE, you can use:

 ngrep -d any -qt -W byline port 5060

 The sip trace will help to see what is wrong with that CANCEL.

 Cheers,
 Daniel


 On 11/28/11 7:19 AM, Sammy Govind wrote:

 Anyone please help.

 On Sat, Nov 26, 2011 at 10:39 PM, Sammy Govind govoi...@gmail.comwrote:

 Hello list,

  I'm using Kamailio 3.1.5 in front of asterisk servers. Kamailio
 handles all the SIP registrations. Calls from SIP phones are forwarded to
 asterisks and then dialled out to Kamailio.

  root@SBCserver:~# kamailio -V
 version: kamailio 3.1.5 (x86_64/linux) 76fff5
 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS,
 DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
 DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE,
 USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
 ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
 MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4MB
 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
 id: 76fff5
 compiled on 08:21:33 Oct 27 2011 with gcc 4.6.1
 root@SBCserver:~#


  Problem:
 When call is initiated from a softphone and is in ringing phase, CANCEL
 just don't work. I've done some initial debugging and the following piece
 of code in main route is failing.

  # CANCEL processing
 if (is_method(CANCEL))
 {
  xlog(L_NOTICE,$rm from $fu (IP:$si:$sp) ---CAPTURED IN
 MAIN---\n);
  if (t_check_trans()){
 t_relay();
 xlog(L_NOTICE,$rm from $fu (IP:$si:$sp) ---CHECK TRANS
 TRUE---\n);
  }
  xlog(L_NOTICE,$rm from $fu (IP:$si:$sp) ---CHECK TRANS
 FALSE---\n);
  exit;
 }

  Also the CANCEL fails the has_totag() condition !

  The same Call CANCEL scenario works fine for any client on Public IP !

  Hope to get some pointers for the solution.

  Regards,
 Sammy.




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 --
 Daniel-Constantin Mierla -- http://www.asipto.com
 Kamailio Advanced Training, Dec 5-8, Berlin: 
 http://asipto.com/u/kathttp://linkedin.com/in/miconda -- 
 http://twitter.com/miconda





siptrace.pcap
Description: Binary data
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Re: [SR-Users] CANCEL not matching INVITES !

2011-11-27 Thread Sammy Govind 
Anyone please help.

On Sat, Nov 26, 2011 at 10:39 PM, Sammy Govind govoi...@gmail.com wrote:

 Hello list,

 I'm using Kamailio 3.1.5 in front of asterisk servers. Kamailio handles
 all the SIP registrations. Calls from SIP phones are forwarded to asterisks
 and then dialled out to Kamailio.

 root@SBCserver:~# kamailio -V
 version: kamailio 3.1.5 (x86_64/linux) 76fff5
 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS,
 DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
 DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE,
 USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
 ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
 MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4MB
 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
 id: 76fff5
 compiled on 08:21:33 Oct 27 2011 with gcc 4.6.1
 root@SBCserver:~#


 Problem:
 When call is initiated from a softphone and is in ringing phase, CANCEL
 just don't work. I've done some initial debugging and the following piece
 of code in main route is failing.

 # CANCEL processing
 if (is_method(CANCEL))
 {
  xlog(L_NOTICE,$rm from $fu (IP:$si:$sp) ---CAPTURED IN MAIN---\n);
  if (t_check_trans()){
 t_relay();
 xlog(L_NOTICE,$rm from $fu (IP:$si:$sp) ---CHECK TRANS
 TRUE---\n);
  }
  xlog(L_NOTICE,$rm from $fu (IP:$si:$sp) ---CHECK TRANS
 FALSE---\n);
  exit;
 }

 Also the CANCEL fails the has_totag() condition !

 The same Call CANCEL scenario works fine for any client on Public IP !

 Hope to get some pointers for the solution.

 Regards,
 Sammy.

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[SR-Users] CANCEL not matching INVITES !

2011-11-26 Thread Sammy Govind 
Hello list,

I'm using Kamailio 3.1.5 in front of asterisk servers. Kamailio handles all
the SIP registrations. Calls from SIP phones are forwarded to asterisks and
then dialled out to Kamailio.

root@SBCserver:~# kamailio -V
version: kamailio 3.1.5 (x86_64/linux) 76fff5
flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS,
DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE,
USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: 76fff5
compiled on 08:21:33 Oct 27 2011 with gcc 4.6.1
root@SBCserver:~#


Problem:
When call is initiated from a softphone and is in ringing phase, CANCEL
just don't work. I've done some initial debugging and the following piece
of code in main route is failing.

# CANCEL processing
if (is_method(CANCEL))
{
 xlog(L_NOTICE,$rm from $fu (IP:$si:$sp) ---CAPTURED IN MAIN---\n);
 if (t_check_trans()){
t_relay();
xlog(L_NOTICE,$rm from $fu (IP:$si:$sp) ---CHECK TRANS
TRUE---\n);
 }
 xlog(L_NOTICE,$rm from $fu (IP:$si:$sp) ---CHECK TRANS FALSE---\n);
 exit;
}

Also the CANCEL fails the has_totag() condition !

The same Call CANCEL scenario works fine for any client on Public IP !

Hope to get some pointers for the solution.

Regards,
Sammy.
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