Re: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius
Try just User-Name attribute in aaa_radius
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Tung Tran
Sent: Tuesday, July 27, 2010 8:58 PM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius
Hello
Thanks for your reponse
I still have a question for you that do I need to set the whole set like this:
User-Name = 1...@x.x.x.x
Calling-Station-Id = 100@ x.x.x.x
Called-Station-Id = sip: x.x.x.x
Digest-User-Name = 100
Digest-Realm = x.x.x.x
Digest-Nonce = 4c4f0b702b314d48a3fcc1aa986ee74362ddff63
Digest-Uri = sip: x.x.x.x
Digest-Method = REGISTER
Digest-Response = ab287b28ee499cc733c27a0c198e066c
Service-Type = Sip-Session
Sip-Uri-User = 100
cisco-avpair = call-id=554f9e2e09304b13
NAS-Port = 5060
NAS-IP-Address x.x.x.x
Or just an User-Name attribute in aaa_radius module?
Above Radius message is what I am doing with Openser 1.2, but I don’t know how
to do the same with Opensips 1.6.2 version.
There is quite changes between 02 versions.
Thanks
T.T
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Denis Putyato
Sent: Tuesday, July 27, 2010 8:11 PM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius
Try to add “Sets” param. with User-Name attribute for acc_radius module in your
opensips.cfg
For example,
modparam(aaa_radius,sets,set1 = (User-Name=$var(usr))”)
where $var(usr) is some PV of your callerid
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Tung Tran
Sent: Tuesday, July 27, 2010 4:55 PM
To: 'OpenSIPS users mailling list'
Subject: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius
Hello,
Just wanna recall if someone can help me out.
Thanks
T.T
From: Tung Tran [mailto:tr.t...@gmail.com]
Sent: Tuesday, July 27, 2010 9:33 AM
To: 'OpenSIPS users mailling list'
Subject: Error when setting OpenSips with Radius
Hi all
I am building the Opensips 1.6.2 to run with external Billing via Radius (using
radiusclient-ng), but I get this error when trying to register. No radius
message is sent to Radius server yet
Jul 27 08:12:03 SVR60 /usr/local/sbin/opensips[19907]: rc_avpair_new: unknown
attribute 0
Jul 27 08:12:03 SVR60 /usr/local/sbin/opensips[19907]:
ERROR:aaa_radius:rad_avp_add: failure
Jul 27 08:12:03 SVR60 /usr/local/sbin/opensips[19907]:
ERROR:auth_aaa:aaa_authorize_sterman: unable to add User-Name attribute
Here is my opensips.cfg
loadmodule auth.so
loadmodule auth_db.so
loadmodule aaa_radius.so
loadmodule auth_aaa.so
modparam(auth_aaa, aaa_url,
radius:/usr/local/etc/radiusclient-ng/radiusclient.conf)
..
if(!aaa_www_authorize())
{
xlog(L_INFO, *-*-* False in Radius_www_authorize ,
challenging M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci .. \n);
www_challenge(, 1);
exit;
}
…
I am very appreciated if someone can point me a hint where is the problem
Thanks in advance
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius
Its works, Thanks so much
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Denis Putyato
Sent: Wednesday, July 28, 2010 1:06 PM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius
Try just User-Name attribute in aaa_radius
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Tung Tran
Sent: Tuesday, July 27, 2010 8:58 PM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius
Hello
Thanks for your reponse
I still have a question for you that do I need to set the whole set like this:
User-Name = 1...@x.x.x.x
Calling-Station-Id = 100@ x.x.x.x
Called-Station-Id = sip: x.x.x.x
Digest-User-Name = 100
Digest-Realm = x.x.x.x
Digest-Nonce = 4c4f0b702b314d48a3fcc1aa986ee74362ddff63
Digest-Uri = sip: x.x.x.x
Digest-Method = REGISTER
Digest-Response = ab287b28ee499cc733c27a0c198e066c
Service-Type = Sip-Session
Sip-Uri-User = 100
cisco-avpair = call-id=554f9e2e09304b13
NAS-Port = 5060
NAS-IP-Address x.x.x.x
Or just an User-Name attribute in aaa_radius module?
Above Radius message is what I am doing with Openser 1.2, but I don’t know how
to do the same with Opensips 1.6.2 version.
There is quite changes between 02 versions.
Thanks
T.T
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Denis Putyato
Sent: Tuesday, July 27, 2010 8:11 PM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius
Try to add “Sets” param. with User-Name attribute for acc_radius module in your
opensips.cfg
For example,
modparam(aaa_radius,sets,set1 = (User-Name=$var(usr))”)
where $var(usr) is some PV of your callerid
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Tung Tran
Sent: Tuesday, July 27, 2010 4:55 PM
To: 'OpenSIPS users mailling list'
Subject: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius
Hello,
Just wanna recall if someone can help me out.
Thanks
T.T
From: Tung Tran [mailto:tr.t...@gmail.com]
Sent: Tuesday, July 27, 2010 9:33 AM
To: 'OpenSIPS users mailling list'
Subject: Error when setting OpenSips with Radius
Hi all
I am building the Opensips 1.6.2 to run with external Billing via Radius (using
radiusclient-ng), but I get this error when trying to register. No radius
message is sent to Radius server yet
Jul 27 08:12:03 SVR60 /usr/local/sbin/opensips[19907]: rc_avpair_new: unknown
attribute 0
Jul 27 08:12:03 SVR60 /usr/local/sbin/opensips[19907]:
ERROR:aaa_radius:rad_avp_add: failure
Jul 27 08:12:03 SVR60 /usr/local/sbin/opensips[19907]:
ERROR:auth_aaa:aaa_authorize_sterman: unable to add User-Name attribute
Here is my opensips.cfg
loadmodule auth.so
loadmodule auth_db.so
loadmodule aaa_radius.so
loadmodule auth_aaa.so
modparam(auth_aaa, aaa_url,
radius:/usr/local/etc/radiusclient-ng/radiusclient.conf)
..
if(!aaa_www_authorize())
{
xlog(L_INFO, *-*-* False in Radius_www_authorize ,
challenging M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci .. \n);
www_challenge(, 1);
exit;
}
…
I am very appreciated if someone can point me a hint where is the problem
Thanks in advance
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius
Just another question, do you know how can I insert the “Calling-station-id”
and “Called-Station-id” in authorize message for INVITE?
Like this ( what I am doing with openser right now)
User-Name = 6528822...@x.x.x.x
Calling-Station-Id = 6528822724@ x.x.x.x
Called-Station-Id = sip:0018323822177@ x.x.x.x
Digest-User-Name = 6528822724
Digest-Realm = x.x.x.x
Digest-Nonce = 4a6763dbd5352b1bf9b8f0873f7bcf781068e516
Digest-Uri = sip:0018323822177@ x.x.x.x
Digest-Method = INVITE
Digest-Response = 0bb87b4cc20f3892c4d743a35cd1fb01
Service-Type = Sip-Session
Sip-Uri-User = 6528822724
cisco-avpair = call-id=b3b259f5-f6fe-1810-86e0-001a803f2...@192.168.1.2
NAS-Port = 5060
NAS-IP-Address = 10.84.0.21
Thanks again.
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Denis Putyato
Sent: Wednesday, July 28, 2010 1:06 PM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius
Try just User-Name attribute in aaa_radius
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Tung Tran
Sent: Tuesday, July 27, 2010 8:58 PM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius
Hello
Thanks for your reponse
I still have a question for you that do I need to set the whole set like this:
User-Name = 1...@x.x.x.x
Calling-Station-Id = 100@ x.x.x.x
Called-Station-Id = sip: x.x.x.x
Digest-User-Name = 100
Digest-Realm = x.x.x.x
Digest-Nonce = 4c4f0b702b314d48a3fcc1aa986ee74362ddff63
Digest-Uri = sip: x.x.x.x
Digest-Method = REGISTER
Digest-Response = ab287b28ee499cc733c27a0c198e066c
Service-Type = Sip-Session
Sip-Uri-User = 100
cisco-avpair = call-id=554f9e2e09304b13
NAS-Port = 5060
NAS-IP-Address x.x.x.x
Or just an User-Name attribute in aaa_radius module?
Above Radius message is what I am doing with Openser 1.2, but I don’t know how
to do the same with Opensips 1.6.2 version.
There is quite changes between 02 versions.
Thanks
T.T
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Denis Putyato
Sent: Tuesday, July 27, 2010 8:11 PM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius
Try to add “Sets” param. with User-Name attribute for acc_radius module in your
opensips.cfg
For example,
modparam(aaa_radius,sets,set1 = (User-Name=$var(usr))”)
where $var(usr) is some PV of your callerid
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Tung Tran
Sent: Tuesday, July 27, 2010 4:55 PM
To: 'OpenSIPS users mailling list'
Subject: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius
Hello,
Just wanna recall if someone can help me out.
Thanks
T.T
From: Tung Tran [mailto:tr.t...@gmail.com]
Sent: Tuesday, July 27, 2010 9:33 AM
To: 'OpenSIPS users mailling list'
Subject: Error when setting OpenSips with Radius
Hi all
I am building the Opensips 1.6.2 to run with external Billing via Radius (using
radiusclient-ng), but I get this error when trying to register. No radius
message is sent to Radius server yet
Jul 27 08:12:03 SVR60 /usr/local/sbin/opensips[19907]: rc_avpair_new: unknown
attribute 0
Jul 27 08:12:03 SVR60 /usr/local/sbin/opensips[19907]:
ERROR:aaa_radius:rad_avp_add: failure
Jul 27 08:12:03 SVR60 /usr/local/sbin/opensips[19907]:
ERROR:auth_aaa:aaa_authorize_sterman: unable to add User-Name attribute
Here is my opensips.cfg
loadmodule auth.so
loadmodule auth_db.so
loadmodule aaa_radius.so
loadmodule auth_aaa.so
modparam(auth_aaa, aaa_url,
radius:/usr/local/etc/radiusclient-ng/radiusclient.conf)
..
if(!aaa_www_authorize())
{
xlog(L_INFO, *-*-* False in Radius_www_authorize ,
challenging M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci .. \n);
www_challenge(, 1);
exit;
}
…
I am very appreciated if someone can point me a hint where is the problem
Thanks in advance
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius
Try add these attributs in aaa_radius
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Tung Tran
Sent: Wednesday, July 28, 2010 10:52 AM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius
Just another question, do you know how can I insert the “Calling-station-id”
and “Called-Station-id” in authorize message for INVITE?
Like this ( what I am doing with openser right now)
User-Name = 6528822...@x.x.x.x
Calling-Station-Id = 6528822724@ x.x.x.x
Called-Station-Id = sip:0018323822177@ x.x.x.x
Digest-User-Name = 6528822724
Digest-Realm = x.x.x.x
Digest-Nonce = 4a6763dbd5352b1bf9b8f0873f7bcf781068e516
Digest-Uri = sip:0018323822177@ x.x.x.x
Digest-Method = INVITE
Digest-Response = 0bb87b4cc20f3892c4d743a35cd1fb01
Service-Type = Sip-Session
Sip-Uri-User = 6528822724
cisco-avpair = call-id=b3b259f5-f6fe-1810-86e0-001a803f2...@192.168.1.2
NAS-Port = 5060
NAS-IP-Address = 10.84.0.21
Thanks again.
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Denis Putyato
Sent: Wednesday, July 28, 2010 1:06 PM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius
Try just User-Name attribute in aaa_radius
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Tung Tran
Sent: Tuesday, July 27, 2010 8:58 PM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius
Hello
Thanks for your reponse
I still have a question for you that do I need to set the whole set like this:
User-Name = 1...@x.x.x.x
Calling-Station-Id = 100@ x.x.x.x
Called-Station-Id = sip: x.x.x.x
Digest-User-Name = 100
Digest-Realm = x.x.x.x
Digest-Nonce = 4c4f0b702b314d48a3fcc1aa986ee74362ddff63
Digest-Uri = sip: x.x.x.x
Digest-Method = REGISTER
Digest-Response = ab287b28ee499cc733c27a0c198e066c
Service-Type = Sip-Session
Sip-Uri-User = 100
cisco-avpair = call-id=554f9e2e09304b13
NAS-Port = 5060
NAS-IP-Address x.x.x.x
Or just an User-Name attribute in aaa_radius module?
Above Radius message is what I am doing with Openser 1.2, but I don’t know how
to do the same with Opensips 1.6.2 version.
There is quite changes between 02 versions.
Thanks
T.T
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Denis Putyato
Sent: Tuesday, July 27, 2010 8:11 PM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius
Try to add “Sets” param. with User-Name attribute for acc_radius module in your
opensips.cfg
For example,
modparam(aaa_radius,sets,set1 = (User-Name=$var(usr))”)
where $var(usr) is some PV of your callerid
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Tung Tran
Sent: Tuesday, July 27, 2010 4:55 PM
To: 'OpenSIPS users mailling list'
Subject: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius
Hello,
Just wanna recall if someone can help me out.
Thanks
T.T
From: Tung Tran [mailto:tr.t...@gmail.com]
Sent: Tuesday, July 27, 2010 9:33 AM
To: 'OpenSIPS users mailling list'
Subject: Error when setting OpenSips with Radius
Hi all
I am building the Opensips 1.6.2 to run with external Billing via Radius (using
radiusclient-ng), but I get this error when trying to register. No radius
message is sent to Radius server yet
Jul 27 08:12:03 SVR60 /usr/local/sbin/opensips[19907]: rc_avpair_new: unknown
attribute 0
Jul 27 08:12:03 SVR60 /usr/local/sbin/opensips[19907]:
ERROR:aaa_radius:rad_avp_add: failure
Jul 27 08:12:03 SVR60 /usr/local/sbin/opensips[19907]:
ERROR:auth_aaa:aaa_authorize_sterman: unable to add User-Name attribute
Here is my opensips.cfg
loadmodule auth.so
loadmodule auth_db.so
loadmodule aaa_radius.so
loadmodule auth_aaa.so
modparam(auth_aaa, aaa_url,
radius:/usr/local/etc/radiusclient-ng/radiusclient.conf)
..
if(!aaa_www_authorize())
{
xlog(L_INFO, *-*-* False in Radius_www_authorize ,
challenging M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci .. \n);
www_challenge(, 1);
exit;
}
…
I am very appreciated if someone can point me a hint where is the problem
Thanks in advance
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] TCP CONNECT ERROR;
Can anyone please help me... On Wed, Jul 28, 2010 at 11:16 AM, Premalatha Kuppan premala...@ngintech.com wrote: Sorry forgot to attach the cfg file..attached now. On Wed, Jul 28, 2010 at 11:15 AM, Premalatha Kuppan premala...@ngintech.com wrote: Still call fails to TC P enabled clients. Please find my config file attached. I dono where iam doing mistake :(..please help Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: REGISTER MESSAGE Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: REGISTER sips:PUBLIC IP http://204.12.57.221:5061SIP/2. From: sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181sip%3a588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181;tag=14097453-6b37-4f1d-bb14-43 Authorization: Digest username=408111, realm=PUBLIC IP, nonce=4c4fbe8500013046f7331a315fdc29e6f14851cbedd9, uri=sips:PUBLIC IP:5061 http://204.12.57.221:5061, response=64375192e261bc5f3213bc6f56 ontent-Length: 0 Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: contact expire=null ; header expire =300; contact uri=sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@209.242.149.98:57086;transport=tls Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: This is register, parsing values - 588aa6e4-2af1-4fc0-ad67-6e5c16681e771000; sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181sip%3a588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181; Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: NORMAL-REGISTRATION Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: CLIENT BEHIND NAT Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: contact before update sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@209.242.149.98:47043;transport=tls Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: via before update SIP/2.0/TLS 10.11.11.181:57086 ;rport;branch=z9hG4bKPj48764d96-c980-4584-aba4-2af5c6273165 Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: REGISTER MESSAGE Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: REGISTER sips: 204.12.57.221:5061 SIP/2. From: sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181sip%3a588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181;tag=9428c49e-e3bb-4241-b3da-8e ontent-Length: 0a6e4-2af1-4fc0-ad67-6e5c16681e771...@209.242.149.98:47043 ;transport=tls Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: REGISTER MESSAGE Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: REGISTER sips:PUBLIC IP:5061 http://204.12.57.221:5061 SIP/2. From: sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181sip%3a588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181;tag=12a8089f-2955-480f-affb-6f ontent-Length: 0a6e4-2af1-4fc0-ad67-6e5c16681e771...@209.242.149.98:57086 ;transport=tls Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: REGISTER MESSAGE Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: REGISTER sips: PUBLIC:IP http://204.12.57.221:5061 SIP/2. From: sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181sip%3a588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181;tag=9428c49e-e3bb-4241-b3da-8e Authorization: Digest username=408111, realm=PUBLIC IP, nonce=4c4fbe8500025275d997ab9c99b5ab5e0eac036575d0, uri=sips:PUBLIC IP http://204.12.57.221:5061, response=a81e038b357fb069199bb83f32 ontent-Length: 0 Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: contact expire=null ; header expire =0; contact uri=sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@209.242.149.98:47043;transport=tls Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: This is de-registration Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: De-register DONE - 588aa6e4-2af1-4fc0-ad67-6e5c16681e771000 Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: REGISTER MESSAGE Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: REGISTER sips:PUBLIC IP:5061 http://204.12.57.221:5061 SIP/2. From: sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181sip%3a588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181;tag=12a8089f-2955-480f-affb-6f Authorization: Digest username=408111, realm=PUBLIC IP, nonce=4c4fbe8600034f58d23302892e98458c1907863d4951, uri=sips: 204.12.57.221:5061, response=df102121f9df50a2e205cd144a ontent-Length: 0 Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: contact expire=null ; header expire =300; contact uri=sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@209.242.149.98:57086;transport=tls Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: This is register, parsing values - 588aa6e4-2af1-4fc0-ad67-6e5c16681e771000; sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181sip%3a588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181; Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: NORMAL-REGISTRATION Jul 28 01:23:48 204548-4 /usr/local/sbin/opensips[21523]: new branch at sip:123...@209.242.149.98:1025;transport=TCP
Re: [OpenSIPS-Users] TCP CONNECT ERROR;
Hi, Have you checked all the things that I told you in the first reply? Probably one of that is not matched and this is why it is not working. Regards, -- Anca Vamanu www.voice-system.ro On 07/28/2010 10:01 AM, Premalatha Kuppan wrote: Can anyone please help me... On Wed, Jul 28, 2010 at 11:16 AM, Premalatha Kuppan premala...@ngintech.com mailto:premala...@ngintech.com wrote: Sorry forgot to attach the cfg file..attached now. On Wed, Jul 28, 2010 at 11:15 AM, Premalatha Kuppan premala...@ngintech.com mailto:premala...@ngintech.com wrote: Still call fails to TC P enabled clients. Please find my config file attached. I dono where iam doing mistake :(..please help Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: REGISTER MESSAGE Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: REGISTER sips:PUBLIC IP http://204.12.57.221:5061SIP/2. From: sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181 mailto:sip%3a588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181;tag=14097453-6b37-4f1d-bb14-43 Authorization: Digest username=408111, realm=PUBLIC IP, nonce=4c4fbe8500013046f7331a315fdc29e6f14851cbedd9, uri=sips:PUBLIC IP:5061 http://204.12.57.221:5061, response=64375192e261bc5f3213bc6f56 ontent-Length: 0 Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: contact expire=null ; header expire =300; contact uri=sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@209.242.149.98:57086;transport=tls Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: This is register, parsing values - 588aa6e4-2af1-4fc0-ad67-6e5c16681e771000; sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181 mailto:sip%3a588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181; Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: NORMAL-REGISTRATION Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: CLIENT BEHIND NAT Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: contact before update sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@209.242.149.98:47043;transport=tls Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: via before update SIP/2.0/TLS 10.11.11.181:57086;rport;branch=z9hG4bKPj48764d96-c980-4584-aba4-2af5c6273165 Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: REGISTER MESSAGE Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: REGISTER sips:204.12.57.221:5061 http://204.12.57.221:5061 SIP/2. From: sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181 mailto:sip%3a588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181;tag=9428c49e-e3bb-4241-b3da-8e ontent-Length: 0a6e4-2af1-4fc0-ad67-6e5c16681e771...@209.242.149.98:47043;transport=tls Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: REGISTER MESSAGE Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: REGISTER sips:PUBLIC IP:5061 http://204.12.57.221:5061 SIP/2. From: sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181 mailto:sip%3a588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181;tag=12a8089f-2955-480f-affb-6f ontent-Length: 0a6e4-2af1-4fc0-ad67-6e5c16681e771...@209.242.149.98:57086;transport=tls Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: REGISTER MESSAGE Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: REGISTER sips:PUBLIC:IP http://204.12.57.221:5061 SIP/2. From: sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181 mailto:sip%3a588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181;tag=9428c49e-e3bb-4241-b3da-8e Authorization: Digest username=408111, realm=PUBLIC IP, nonce=4c4fbe8500025275d997ab9c99b5ab5e0eac036575d0, uri=sips:PUBLIC IP http://204.12.57.221:5061, response=a81e038b357fb069199bb83f32 ontent-Length: 0 Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: contact expire=null ; header expire =0; contact uri=sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@209.242.149.98:47043;transport=tls Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: This is de-registration Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: De-register DONE - 588aa6e4-2af1-4fc0-ad67-6e5c16681e771000 Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: REGISTER MESSAGE Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: REGISTER sips:PUBLIC IP:5061 http://204.12.57.221:5061 SIP/2. From: sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181
Re: [OpenSIPS-Users] TCP CONNECT ERROR;
Yes, i have checked. Have you seen my config file. Probably can you guide me where its going wrong ? On Wed, Jul 28, 2010 at 2:36 PM, Anca Vamanu a...@opensips.org wrote: Hi, Have you checked all the things that I told you in the first reply? Probably one of that is not matched and this is why it is not working. Regards, -- Anca Vamanuwww.voice-system.ro On 07/28/2010 10:01 AM, Premalatha Kuppan wrote: Can anyone please help me... On Wed, Jul 28, 2010 at 11:16 AM, Premalatha Kuppan premala...@ngintech.com wrote: Sorry forgot to attach the cfg file..attached now. On Wed, Jul 28, 2010 at 11:15 AM, Premalatha Kuppan premala...@ngintech.com wrote: Still call fails to TC P enabled clients. Please find my config file attached. I dono where iam doing mistake :(..please help Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: REGISTER MESSAGE Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: REGISTER sips:PUBLIC IP http://204.12.57.221:5061SIP/2. From: sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181sip%3a588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181;tag=14097453-6b37-4f1d-bb14-43 Authorization: Digest username=408111, realm=PUBLIC IP, nonce=4c4fbe8500013046f7331a315fdc29e6f14851cbedd9, uri=sips:PUBLIC IP:5061 http://204.12.57.221:5061, response=64375192e261bc5f3213bc6f56 ontent-Length: 0 Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: contact expire=null ; header expire =300; contact uri= sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@209.242.149.98:57086;transport=tls Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: This is register, parsing values - 588aa6e4-2af1-4fc0-ad67-6e5c16681e771000; sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181sip%3a588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181; Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: NORMAL-REGISTRATION Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: CLIENT BEHIND NAT Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: contact before update sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@209.242.149.98:47043;transport=tls Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: via before update SIP/2.0/TLS 10.11.11.181:57086 ;rport;branch=z9hG4bKPj48764d96-c980-4584-aba4-2af5c6273165 Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: REGISTER MESSAGE Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: REGISTER sips: 204.12.57.221:5061 SIP/2. From: sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181sip%3a588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181;tag=9428c49e-e3bb-4241-b3da-8e ontent-Length: 0a6e4-2af1-4fc0-ad67-6e5c16681e771...@209.242.149.98:47043;transport=tls Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: REGISTER MESSAGE Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: REGISTER sips:PUBLIC IP:5061 http://204.12.57.221:5061 SIP/2. From: sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181sip%3a588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181;tag=12a8089f-2955-480f-affb-6f ontent-Length: 0a6e4-2af1-4fc0-ad67-6e5c16681e771...@209.242.149.98:57086;transport=tls Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: REGISTER MESSAGE Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: REGISTER sips: PUBLIC:IP http://204.12.57.221:5061 SIP/2. From: sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181sip%3a588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181;tag=9428c49e-e3bb-4241-b3da-8e Authorization: Digest username=408111, realm=PUBLIC IP, nonce=4c4fbe8500025275d997ab9c99b5ab5e0eac036575d0, uri=sips:PUBLIC IP http://204.12.57.221:5061, response=a81e038b357fb069199bb83f32 ontent-Length: 0 Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: contact expire=null ; header expire =0; contact uri= sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@209.242.149.98:47043;transport=tls Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: This is de-registration Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: De-register DONE - 588aa6e4-2af1-4fc0-ad67-6e5c16681e771000 Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: REGISTER MESSAGE Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: REGISTER sips:PUBLIC IP:5061 http://204.12.57.221:5061 SIP/2. From: sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181sip%3a588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181;tag=12a8089f-2955-480f-affb-6f Authorization: Digest username=408111, realm=PUBLIC IP, nonce=4c4fbe8600034f58d23302892e98458c1907863d4951, uri=sips: 204.12.57.221:5061, response=df102121f9df50a2e205cd144a ontent-Length: 0 Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: contact expire=null ; header expire =300; contact uri= sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@209.242.149.98:57086;transport=tls Jul 28 01:21:44 204548-4
Re: [OpenSIPS-Users] TCP CONNECT ERROR;
Hi,
Have you seen my cfg file ? I have tried fix_nat_contact() and
fix_nat_register also...:(
Am i missing anything ?
/* uncomment the next line to disable TCP (default on) */
disable_tcp=no
tcp_children=10
tcp_connect_timeout=1
tcp_send_timeout=3
tcp_accept_aliases=1
# - registrar params -
/* uncomment the next line not to allow more than 10 contacts per AOR */
#modparam(registrar, max_contacts, 10)
modparam(registrar,tcp_persistent_flag,7)
modparam(registrar, received_avp, $avp(s:rcv))
#---nathelper params --
modparam(nathelper, received_avp, $avp(s:rcv))
Routing Logic:
if (client_nat_test(3)) {
xlog(CLIENT BEHIND NAT\n);
if (method==SUBSCRIBE)
nat_keepalive();
if (!has_totag()) {
if (method==REGISTER) {
if (proto==UDP)
nat_keepalive();
else
setflag(7);
$avp(s:regrcv) = $source_uri;
fix_nated_register();
}
else if (method==INVITE) {
nat_keepalive();
fix_nated_contact();
fix_nated_sdp(3);
}
}
}
if (is_method(REGISTER))
{
xlog(REGISTER MESSAGE\n);
xlog($mb\n);
# authenticate the REGISTER requests (uncomment to enable
auth)
if (!www_authorize(xxx, subscriber))
{
www_challenge(xx, 0);
exit;
}
###TCP ENABLED CLIENTS
if(proto==TCP)
{
xlog(the SIP message was received over TCP\n);
xlog($(tu{uri.user})\n);
force_tcp_alias();
}
Thanks,
Prem
On Wed, Jul 28, 2010 at 2:44 PM, Premalatha Kuppan
premala...@ngintech.comwrote:
Yes, i have checked. Have you seen my config file. Probably can you guide
me where its going wrong ?
On Wed, Jul 28, 2010 at 2:36 PM, Anca Vamanu a...@opensips.org wrote:
Hi,
Have you checked all the things that I told you in the first reply?
Probably one of that is not matched and this is why it is not working.
Regards,
--
Anca Vamanuwww.voice-system.ro
On 07/28/2010 10:01 AM, Premalatha Kuppan wrote:
Can anyone please help me...
On Wed, Jul 28, 2010 at 11:16 AM, Premalatha Kuppan
premala...@ngintech.com wrote:
Sorry forgot to attach the cfg file..attached now.
On Wed, Jul 28, 2010 at 11:15 AM, Premalatha Kuppan
premala...@ngintech.com wrote:
Still call fails to TC P enabled clients.
Please find my config file attached. I dono where iam doing mistake
:(..please help
Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: REGISTER
MESSAGE
Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: REGISTER
sips:PUBLIC
IP http://204.12.57.221:5061SIP/2. From:
sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181sip%3a588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181;tag=14097453-6b37-4f1d-bb14-43
Authorization: Digest username=408111, realm=PUBLIC IP,
nonce=4c4fbe8500013046f7331a315fdc29e6f14851cbedd9, uri=sips:PUBLIC
IP:5061 http://204.12.57.221:5061,
response=64375192e261bc5f3213bc6f56 ontent-Length: 0
Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: contact
expire=null ; header expire =300; contact uri=
sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@209.242.149.98:57086;transport=tls
Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: This is
register, parsing values - 588aa6e4-2af1-4fc0-ad67-6e5c16681e771000;
sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181sip%3a588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181;
Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]:
NORMAL-REGISTRATION
Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: CLIENT BEHIND
NAT
Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: contact before
update
sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@209.242.149.98:47043;transport=tls
Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: via before
update SIP/2.0/TLS 10.11.11.181:57086
;rport;branch=z9hG4bKPj48764d96-c980-4584-aba4-2af5c6273165
Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: REGISTER
MESSAGE
Jul 28 01:21:43 204548-4 /usr/local/sbin/opensips[21528]: REGISTER sips:
204.12.57.221:5061 SIP/2. From:
sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181sip%3a588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181;tag=9428c49e-e3bb-4241-b3da-8e
ontent-Length:
0a6e4-2af1-4fc0-ad67-6e5c16681e771...@209.242.149.98:47043;transport=tls
Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: REGISTER
MESSAGE
Jul 28 01:21:44 204548-4 /usr/local/sbin/opensips[21528]: REGISTER
sips:PUBLIC
IP:5061 http://204.12.57.221:5061 SIP/2. From:
sip:588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181sip%3a588aa6e4-2af1-4fc0-ad67-6e5c16681e771...@10.11.11.181;tag=12a8089f-2955-480f-affb-6f
ontent-Length:
[OpenSIPS-Users] help, server crashes every hour
Hi,
I'm using opensips 1.6.2, the server handle only subscribe and notify
requests.
I use the mi datagram to publish new status (which I receive via external
XML process).
Every thing works as fine but the server keeps crashing about every hour.
I tested the configuration using two different servers , one clean redhat
5.3 install and the other is gentoo. This happen on both servers.
I've tested different configuration, but no matter what the server keeps
crashing.
this is bt form gdb :
#0 0x2b659d21c836 in mi_publ_rpl_cback (hentity=value optimized out,
reply=0x0) at mi_func.c:316
#1 0x2b659d00ba53 in publ_cback_func (t=value optimized out,
type=value optimized out, ps=0x2b659c330420) at pua_callback.h:72
#2 0x2b659c10534b in run_trans_callbacks (type=512,
trans=0x2b659e37af60, req=0x0, rpl=0x7885f8, code=value optimized out) at
t_hooks.c:208
#3 0x2b659c11da69 in local_reply (t=0x2b659e37af60,
p_msg=0x2b659c341d38, branch=value optimized out, msg_status=value
optimized out,
cancel_bitmap=0x2b659e2fe6d8) at t_reply.c:1339
#4 0x2b659c120389 in reply_received (p_msg=0x7885f8) at t_reply.c:1484
#5 0x00421448 in forward_reply (msg=0x7885f8) at forward.c:559
#6 0x00456182 in receive_msg (
buf=0x754de0 SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP
192.168.167.62;branch=z9hG4bKd94.311a3611.0\r\nTo:
sip:972529984...@192.168.167.62:5060;tag=595f1ec520b32ef8798f023cc8e6d5bc-be8e\r\nFrom:
sip:972529984...@192.168.167.62..., len=431, rcv_info=0x7fff0f218e20) at
receive.c:200
#7 0x0049a354 in udp_rcv_loop () at udp_server.c:492
#8 0x00429c0d in main (argc=3, argv=value optimized out) at
main.c:818
and this is the config file :
loadmodule db_mysql.so
loadmodule sl.so
loadmodule maxfwd.so
loadmodule textops.so
loadmodule tm.so
loadmodule rr.so
loadmodule dialog.so
loadmodule presence.so
loadmodule presence_dialoginfo.so
loadmodule usrloc.so
loadmodule registrar.so
loadmodule pua.so
loadmodule pua_mi.so
loadmodule signaling.so
#loadmodule presence_xml.so
loadmodule mi_fifo.so
#loadmodule pua_usrloc.so
loadmodule mi_datagram.so
loadmodule pua_dialoginfo.so
loadmodule exec.so
modparam(db_mysql, ping_interval, 300)
modparam(mi_datagram, socket_name, udp:127.0.0.1:8808)
# -- presence params --
#modparam(presence_xml, db_url,
mysql://opensips:opensip...@127.0.0.1/opensips)
#modparam(presence_xml, force_active, 1)
modparam(mi_fifo, fifo_name, /tmp/opensips_fifo)
# - setting module-specific parameters ---
# -- usrloc params --
modparam(usrloc, db_mode, 2)
modparam(usrloc, db_url,
mysql://opensips:opensip...@localhost/opensips)
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam(rr, enable_full_lr, 1)
# -- presence and presence_xml params --
modparam(presence, db_url,
mysql://opensips:opensip...@localhost/opensips)
#modparam(presence, server_address, sip:192.168.167.62:5060)
modparam(presence, db_update_period, 0)
modparam(presence, clean_period, 600)
modparam(presence, expires_offset, 15)
#modparam(presence, fallback2db, 1)
# -- pua params --
modparam(pua, db_url, mysql://opensips:opensip...@localhost/opensips)
modparam(pua, update_period, 60)
modparam(pua, default_expires, 3600)
#modparam(pua, max_expires, 600)
modparam(pua_mi, presence_server, sip:192.168.167.62:5060)
#modparam(pua_mi, presence_server, sip:172.30.48.99:5060)
modparam(pua_dialoginfo, presence_server, sip:192.168.167.62:5060)
#modparam(pua_dialoginfo, presence_server, sip:172.30.48.99:5060)
# -- dialog params --
modparam(dialog, db_url,
mysql://opensips:opensip...@localhost/opensips)
modparam(dialog, dlg_flag, 4)
# - request routing logic ---
# main routing logic
route{
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483,Too Many Hops);
exit;
};
if (msg:len = 2048 ) {
sl_send_reply(513, Message too big);
exit;
};
# path determined by record-routing
if (loose_route()) {
# mark routing logic in request
append_hf(P-hint: rr-enforced\r\n);
route(1);
};
if (uri==myself) {
if( is_method(PUBLISH|SUBSCRIBE|NOTIFY))
if(method==NOTIFY)
pua_update_contact();
route(2);
}
}
route[1] {
if (!t_relay()) {
sl_reply_error();
};
exit;
}
route[2]
{
if ( !t_newtran() ){
sl_reply_error();
exit;
};
if(is_method(PUBLISH)){
##//if($hdr(Sender)!= NULL)
##//handle_publish($hdr(Sender));
##//else
handle_publish();
}
else
if( is_method(SUBSCRIBE)){
exec_msg(echo '$fU $tU' /dev/tcp/127.0.0.1/50008);
handle_subscribe();
}
exit;
}
Thanks,
Yaniv
--
View this
Re: [OpenSIPS-Users] help, server crashes every hour
Hi Ynaiv, Can you please run the command 'p mi_hdl' in you gdb console and send back the output? Regards, -- Anca Vamanu www.voice-system.ro On 07/28/2010 01:58 PM, Yaniv Vaknin wrote: Hi, I'm using opensips 1.6.2, the server handle only subscribe and notify requests. I use the mi datagram to publish new status (which I receive via external XML process). Every thing works as fine but the server keeps crashing about every hour. I tested the configuration using two different servers , one clean redhat 5.3 install and the other is gentoo. This happen on both servers. I've tested different configuration, but no matter what the server keeps crashing. this is bt form gdb : #0 0x2b659d21c836 in mi_publ_rpl_cback (hentity=value optimized out, reply=0x0) at mi_func.c:316 #1 0x2b659d00ba53 in publ_cback_func (t=value optimized out, type=value optimized out, ps=0x2b659c330420) at pua_callback.h:72 #2 0x2b659c10534b in run_trans_callbacks (type=512, trans=0x2b659e37af60, req=0x0, rpl=0x7885f8, code=value optimized out) at t_hooks.c:208 #3 0x2b659c11da69 in local_reply (t=0x2b659e37af60, p_msg=0x2b659c341d38, branch=value optimized out, msg_status=value optimized out, cancel_bitmap=0x2b659e2fe6d8) at t_reply.c:1339 #4 0x2b659c120389 in reply_received (p_msg=0x7885f8) at t_reply.c:1484 #5 0x00421448 in forward_reply (msg=0x7885f8) at forward.c:559 #6 0x00456182 in receive_msg ( buf=0x754de0 SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 192.168.167.62;branch=z9hG4bKd94.311a3611.0\r\nTo: sip:972529984...@192.168.167.62:5060;tag=595f1ec520b32ef8798f023cc8e6d5bc-be8e\r\nFrom: sip:972529984...@192.168.167.62..., len=431, rcv_info=0x7fff0f218e20) at receive.c:200 #7 0x0049a354 in udp_rcv_loop () at udp_server.c:492 #8 0x00429c0d in main (argc=3, argv=value optimized out) at main.c:818 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Mediaproxy not updating with new SDPs
Saul, Could you verify you got the trace file I sent? I mailed it directly to you so I wouldn't have to worry about obfuscation on the IP addresses and want to make sure I didn't get caught in a spam filter or anything. Richard On Jul 27, 2010, at 7:48 AM, Richard Revels wrote: Re RFC4317, yes that is the scenario. My complaint here has nothing to do with opensips. I was just venting about the UA sending the reinvite so quickly after the ACK that it sometimes reaches the destination first. Let's just say that not all endpoints handle this as gracefully as they could. : Re reinvite, I thought I understood engage_media_proxy would handle reinvites automagically. Just making sure. Thank you for verifying that. Re UA port, yes it is advertising one port and using another. A picture is worth a thousand words. I'll send the trace. Richard On Jul 27, 2010, at 4:21 AM, Saúl Ibarra Corretgé wrote: Hi Richard, On 24/07/10 23:05, Richard Revels wrote: Saul (and everyone) good afternoon. I think I've come across a call flow that Mediaproxy would be expected to handle but is not. Umm, it's like sacrilege I know; but I think Mediaproxy may have a bug. 1) The originator sends INVITE with a couple of codec choices. 2) engage_media proxy is called and the INVITE is forwarded 3) 200 comes back with a couple of codec choices and m line indicates rtp will source from port 35210 (and indeed rtp starts sourcing from there 4) The ACK comes back and for 10 milliseconds life is good Then the Originator reinvites with only the single codec choice that it believes was negotiated. Standard procedure for this device and allowed and all that but really, 10 milliseconds?? Anyway... So, would this be RFC4317 section 2.2? THat is: user A invites user B with 3 codecs but he only supports one at a time. Then B answers with 2 codecs and A reinvites with a single codec. Is this right? 5) reinvite is forwarded without any mediaproxy calls or anything from the script If you are using engage_media_proxy this is done automatically by the dialog module. 6) The destination sends back a 200 but this time claims it will source from port 1082. 7) The destination rtp continues to source from the original port 35210 So, the UA said he is using port 1082, but is he actually using it? Or is he advertising one port and then using a different one? So, the result here is that rtp comes from the origination and is forwarded, by Mediaproxy, to port 35210 at the destination where it is happily accepted. Media also flows from the destination from port 35210 but it is not forwarded which results in one way audio on the call. It will be Monday before I am able to request more calls to be made but I'm kind of thinking the firewall may be the cause of the port not changing and the UA is actually sending from a new port. Maybe not though. In either case, should the Mediaproxy have waited for a rtp packet on any port from the (call) destination IP after the 200 to the reinvite and set a new connection track rule after it saw one? Session information is updated with any reinvite that could take place. Could you provide me with a SIP trace for the whole call? Regards, -- Saúl Ibarra Corretgé AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] troubleshooting memory problems
Just a quick follow-up on this. I was cleaning up this server today (plus the couple problems I was having with b2bua - refer scenario were fixed with last bug fix) and realized that there are times when I can be pretty dense. I was somewhat perplexed as to why I didn't get the debug output I was looking for with only the DDBG_QM_MALLOC uncommented but Bogdan was pretty firm that should be all that is needed. Commenting them back out today, it finally hit home it's a continued line and commenting out one line below another doesn't really do anything. : Move DDBG_QM_MALLOC above DDBG_F_MALLOC in the file and THEN uncomment DDBG_QM_MALLOC. Hopefully this will help someone down the road. Should be obvious but got past me for quite a while. DOAH! Richard On May 27, 2010, at 10:35 AM, Bogdan-Andrei Iancu wrote: Hi Richard, Richard Revels wrote: In Makefile.defs uncomment -DDBG_QM_MALLOC \ -DDBG_F_MALLOC \ use only DDBG_QM_MALLOC !! In script set debug=6 memlog=6 Restart and let run for a while. Then cat /var/log/opensips-msg | egrep 'freeing|DBG:core:fm_malloc.*called' | sed -e 's/.*free.*\: \(.*\)/\1-mfree/' -e 's/.*malloc.*\: \(.*\)/\1-malloc/' | sort | uniq -c Adjust path for wherever you are logging of course. Your output will have something like 3015 parse_contact(81)-malloc 3015 parse_contact(81)-mfree 3015 parse_contacts(192)-malloc 3015 parse_contacts(192)-mfree 19592 parse_from_header(63)-malloc 19592 parse_from_header(63)-mfree 335368 parse_headers(309)-malloc 335368 parse_headers(309)-mfree for all the calls that are fine. Then something like 14922 do_parse_rr_body(65)-malloc 8989 do_parse_rr_body(65)-mfree or 9016 sip_msg_cloner(437)-malloc 6003 sip_msg_cloner(437)-mfree That is not relevant as a mem block can be allocated in function X and freed in function Y, so you cannot correlate the numbers. for calls that need further looking into. You'll probably want to go grep out the problem values in the log to get more information about what's calling them. I'm chasing a fairly nasty memory leak (shared memory) right now and thought I would document / share some of the methods we use for this type of thing. See: http://www.opensips.org/Resources/DocsTsMem - try to get a dump to see if there are leaks. Regards, Bogdan Richard ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] B2BUA Error
Hi,
Iam using Opensips as proxy and B2BUA. But B2BUA doesn't seems to work. Iam
getting following error.
This is to Handle REFER request. From Asterisk Iam sending REFER request to
Opensips. But getting following error.
any insight ?
1. From Asterisk:
REFER sip:4081112...@:5060 SIP/2.0
Via: SIP/2.0/UDP yy:5070;branch=z9hG4bK5a523824;rport
Route: sip:y;lr=on
From: sip:7081110...@xxx;tag=as6ed72d65
To: 408111 sip:4081112...@yy;tag=1549130118
Contact: sip:ivr_pic...@yy:5070
Call-ID: 0e1a436f-6a3a-7537-d2ff-b6d5fff56...@xxx
CSeq: 102 REFER
User-Agent: Asterisk PBX
Max-Forwards: 70
Refer-To: sip:999_44_1...@xx:41452;transport=tcp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Referred-By: sip:ivr_pic...@xx:5070
2. Opensips.cfg
if (method==REFER)
{
xlog(Call from Asterisk PROTOCOL=$pr PORT=$op PROTOCOL
PORT=$oP RQT=$oU \n);
b2b_init_request(refer);
}
3. Opensips Log:
Jul 28 09:17:40 204548-4 /usr/local/sbin/opensips[6199]:
ERROR:b2b_entities:b2b_new_dlg: no Content-Length header found!
Jul 28 09:17:40 204548-4 /usr/local/sbin/opensips[6199]:
ERROR:b2b_entities:server_new: failed to create new dialog structure entry
Jul 28 09:17:40 204548-4 /usr/local/sbin/opensips[6199]:
ERROR:b2b_logic:b2b_process_scenario_init: failed to create new b2b server
instance
Jul 28 09:17:41 204548-4 /usr/local/sbin/opensips[6199]:
ERROR:b2b_entities:b2b_new_dlg: no Content-Length header found!
Jul 28 09:17:41 204548-4 /usr/local/sbin/opensips[6199]:
ERROR:b2b_entities:server_new: failed to create new dialog structure entry
Jul 28 09:17:41 204548-4 /usr/local/sbin/opensips[6199]:
ERROR:b2b_logic:b2b_process_scenario_init: failed to create new b2b server
instance
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Re: [OpenSIPS-Users] B2BUA Error
Hi,
You made the classic error due to not reading the documentation
carefully - you need to call b2b_init_request on the initial Invite( and
only on it). Do not call it on REFER.
Regards,
--
Anca Vamanu
www.voice-system.ro
On 07/28/2010 04:35 PM, Premalatha Kuppan wrote:
Hi,
Iam using Opensips as proxy and B2BUA. But B2BUA doesn't seems to
work. Iam getting following error.
This is to Handle REFER request. From Asterisk Iam sending REFER
request to Opensips. But getting following error.
any insight ?
1. From Asterisk:
REFER sip:4081112...@:5060 SIP/2.0
Via: SIP/2.0/UDP yy:5070;branch=z9hG4bK5a523824;rport
Route: sip:y;lr=on
From: sip:7081110...@xxx;tag=as6ed72d65
To: 408111 sip:4081112...@yy;tag=1549130118
Contact: sip:ivr_pic...@yy:5070
Call-ID: 0e1a436f-6a3a-7537-d2ff-b6d5fff56...@xxx
CSeq: 102 REFER
User-Agent: Asterisk PBX
Max-Forwards: 70
Refer-To: sip:999_44_1...@xx:41452;transport=tcp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Referred-By: sip:ivr_pic...@xx:5070
2. Opensips.cfg
if (method==REFER)
{
xlog(Call from Asterisk PROTOCOL=$pr PORT=$op PROTOCOL
PORT=$oP RQT=$oU \n);
b2b_init_request(refer);
}
3. Opensips Log:
Jul 28 09:17:40 204548-4 /usr/local/sbin/opensips[6199]:
ERROR:b2b_entities:b2b_new_dlg: no Content-Length header found!
Jul 28 09:17:40 204548-4 /usr/local/sbin/opensips[6199]:
ERROR:b2b_entities:server_new: failed to create new dialog structure entry
Jul 28 09:17:40 204548-4 /usr/local/sbin/opensips[6199]:
ERROR:b2b_logic:b2b_process_scenario_init: failed to create new b2b
server instance
Jul 28 09:17:41 204548-4 /usr/local/sbin/opensips[6199]:
ERROR:b2b_entities:b2b_new_dlg: no Content-Length header found!
Jul 28 09:17:41 204548-4 /usr/local/sbin/opensips[6199]:
ERROR:b2b_entities:server_new: failed to create new dialog structure entry
Jul 28 09:17:41 204548-4 /usr/local/sbin/opensips[6199]:
ERROR:b2b_logic:b2b_process_scenario_init: failed to create new b2b
server instance
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Re: [OpenSIPS-Users] B2BUA Error
But, from asterisk iam getting only REFER, no INVITE request. So, it emans i
can't make it to work as B2BUA in my scenario ?
On Wed, Jul 28, 2010 at 7:11 PM, Anca Vamanu a...@opensips.org wrote:
Hi,
You made the classic error due to not reading the documentation carefully -
you need to call b2b_init_request on the initial Invite( and only on it). Do
not call it on REFER.
Regards,
--
Anca Vamanuwww.voice-system.ro
On 07/28/2010 04:35 PM, Premalatha Kuppan wrote:
Hi,
Iam using Opensips as proxy and B2BUA. But B2BUA doesn't seems to work. Iam
getting following error.
This is to Handle REFER request. From Asterisk Iam sending REFER request to
Opensips. But getting following error.
any insight ?
1. From Asterisk:
REFER sip:4081112...@:5060 SIP/2.0
Via: SIP/2.0/UDP yy:5070;branch=z9hG4bK5a523824;rport
Route: sip:y;lr=on
From: sip:7081110...@xxx;tag=as6ed72d65
To: 408111 sip:4081112...@yy;tag=1549130118
Contact: sip:ivr_pic...@yy:5070
Call-ID: 0e1a436f-6a3a-7537-d2ff-b6d5fff56...@xxx
CSeq: 102 REFER
User-Agent: Asterisk PBX
Max-Forwards: 70
Refer-To: sip:999_44_1...@xx:41452;transport=tcp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Referred-By: sip:ivr_pic...@xx:5070
2. Opensips.cfg
if (method==REFER)
{
xlog(Call from Asterisk PROTOCOL=$pr PORT=$op PROTOCOL
PORT=$oP RQT=$oU \n);
b2b_init_request(refer);
}
3. Opensips Log:
Jul 28 09:17:40 204548-4 /usr/local/sbin/opensips[6199]:
ERROR:b2b_entities:b2b_new_dlg: no Content-Length header found!
Jul 28 09:17:40 204548-4 /usr/local/sbin/opensips[6199]:
ERROR:b2b_entities:server_new: failed to create new dialog structure entry
Jul 28 09:17:40 204548-4 /usr/local/sbin/opensips[6199]:
ERROR:b2b_logic:b2b_process_scenario_init: failed to create new b2b server
instance
Jul 28 09:17:41 204548-4 /usr/local/sbin/opensips[6199]:
ERROR:b2b_entities:b2b_new_dlg: no Content-Length header found!
Jul 28 09:17:41 204548-4 /usr/local/sbin/opensips[6199]:
ERROR:b2b_entities:server_new: failed to create new dialog structure entry
Jul 28 09:17:41 204548-4 /usr/local/sbin/opensips[6199]:
ERROR:b2b_logic:b2b_process_scenario_init: failed to create new b2b server
instance
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Re: [OpenSIPS-Users] B2BUA Error
Hi, No, it won't work without passing all the dialog messages through opensips. Regards, -- Anca Vamanu www.voice-system.ro On 07/28/2010 04:45 PM, Premalatha Kuppan wrote: But, from asterisk iam getting only REFER, no INVITE request. So, it emans i can't make it to work as B2BUA in my scenario ? On Wed, Jul 28, 2010 at 7:11 PM, Anca Vamanu a...@opensips.org mailto:a...@opensips.org wrote: Hi, You made the classic error due to not reading the documentation carefully - you need to call b2b_init_request on the initial Invite( and only on it). Do not call it on REFER. Regards, -- Anca Vamanu www.voice-system.ro http://www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] B2BUA Error
Sorry. I couldn't get you. Actually, 1. User is making initial INVITE to opensips(proxy), 2.opensips will forward it to Asterisk for IVR, 3.now after IVR, asterisk transfer the call to destination via Opensips. As i mentioned earlier since asterisk 1.4x has no support to TCP/TLS, i want opensips to handle the REFER request and connect the peers. Here, if i call b2b_init_request during initial INVITE, will my problem get solved ? Or is that making Opensisp (Proxy) to act as well as B2BUA will not make logic in my scenario. On Wed, Jul 28, 2010 at 7:24 PM, Anca Vamanu a...@opensips.org wrote: Hi, No, it won't work without passing all the dialog messages through opensips. Regards, -- Anca Vamanuwww.voice-system.ro On 07/28/2010 04:45 PM, Premalatha Kuppan wrote: But, from asterisk iam getting only REFER, no INVITE request. So, it emans i can't make it to work as B2BUA in my scenario ? On Wed, Jul 28, 2010 at 7:11 PM, Anca Vamanu a...@opensips.org wrote: Hi, You made the classic error due to not reading the documentation carefully - you need to call b2b_init_request on the initial Invite( and only on it). Do not call it on REFER. Regards, -- Anca Vamanuwww.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] B2BUA Error
On 07/28/2010 05:12 PM, Premalatha Kuppan wrote: Sorry. I couldn't get you. Actually, 1. User is making initial INVITE to opensips(proxy), 2.opensips will forward it to Asterisk for IVR, 3.now after IVR, asterisk transfer the call to destination via Opensips. As i mentioned earlier since asterisk 1.4x has no support to TCP/TLS, i want opensips to handle the REFER request and connect the peers. Here, if i call b2b_init_request during initial INVITE, will my problem get solved ? YES. Or is that making Opensisp (Proxy) to act as well as B2BUA will not make logic in my scenario. On Wed, Jul 28, 2010 at 7:24 PM, Anca Vamanu a...@opensips.org mailto:a...@opensips.org wrote: Hi, No, it won't work without passing all the dialog messages through opensips. Regards, -- Anca Vamanu www.voice-system.ro http://www.voice-system.ro On 07/28/2010 04:45 PM, Premalatha Kuppan wrote: But, from asterisk iam getting only REFER, no INVITE request. So, it emans i can't make it to work as B2BUA in my scenario ? On Wed, Jul 28, 2010 at 7:11 PM, Anca Vamanu a...@opensips.org mailto:a...@opensips.org wrote: Hi, You made the classic error due to not reading the documentation carefully - you need to call b2b_init_request on the initial Invite( and only on it). Do not call it on REFER. Regards, -- Anca Vamanu www.voice-system.ro http://www.voice-system.ro ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Anca Vamanu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] B2BUA Error
I have changed the .cfg file. But still getting the same problem,
# main request routing logic
route{
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483,Too Many Hops);
exit;
}
if (is_method(INVITE))
{ b2b_init_request(refer); }
Jul 28 10:35:15 204548-4 /usr/local/sbin/opensips[8837]:
ERROR:b2b_entities:b2b_new_dlg: no Content-Length header found!
Jul 28 10:35:15 204548-4 /usr/local/sbin/opensips[8837]:
ERROR:b2b_entities:server_new: failed to create new dialog structure entry
Jul 28 10:35:15 204548-4 /usr/local/sbin/opensips[8837]:
ERROR:b2b_logic:b2b_process_scenario_init: failed to create new b2b server
instance
Jul 28 10:35:16 204548-4 /usr/local/sbin/opensips[8837]:
ERROR:b2b_entities:b2b_new_dlg: no Content-Length header found!
Jul 28 10:35:16 204548-4 /usr/local/sbin/opensips[8837]:
ERROR:b2b_entities:server_new: failed to create new dialog structure entry
Jul 28 10:35:16 204548-4 /usr/local/sbin/opensips[8837]:
ERROR:b2b_logic:b2b_process_scenario_init: failed to create new b2b server
instance
Jul 28 10:35:16 204548-4 /usr/local/sbin/opensips[8838]: ACC: transaction
answered:
timestamp=1280327716;method=BYE;from_tag=as3b753c6d;to_tag=790088202;call_id=3ec3441c-5d0b-eb6e-51d3-f4e482048...@xx;code=200;reason=Ok
Jul 28 10:35:17 204548-4 /usr/local/sbin/opensips[8837]:
ERROR:b2b_entities:b2b_new_dlg: no Content-Length header found!
Jul 28 10:35:17 204548-4 /usr/local/sbin/opensips[8837]:
ERROR:b2b_entities:server_new: failed to create new dialog structure entry
Jul 28 10:35:17 204548-4 /usr/local/sbin/opensips[8837]:
ERROR:b2b_logic:b2b_process_scenario_init: failed to create new b2b server
instance
Jul 28 10:35:19 204548-4 /usr/local/sbin/opensips[8838]:
ERROR:b2b_entities:b2b_new_dlg: no Content-Length header found!
Jul 28 10:35:19 204548-4 /usr/local/sbin/opensips[8838]:
ERROR:b2b_entities:server_new: failed to create new dialog structure entry
Jul 28 10:35:19 204548-4 /usr/local/sbin/opensips[8838]:
ERROR:b2b_logic:b2b_process_scenario_init: failed to create new b2b server
instance
Jul 28 10:35:23 204548-4 /usr/local/sbin/opensips[8837]:
ERROR:b2b_entities:b2b_new_dlg: no Content-Length header found!
Jul 28 10:35:23 204548-4 /usr/local/sbin/opensips[8837]:
ERROR:b2b_entities:server_new: failed to create new dialog structure entry
Jul 28 10:35:23 204548-4 /usr/local/sbin/opensips[8837]:
ERROR:b2b_logic:b2b_process_scenario_init: failed to create new b2b server
instance
Jul 28 10:35:27 204548-4 /usr/local/sbin/opensips[8838]:
ERROR:b2b_entities:b2b_new_dlg: no Content-Length header found!
Jul 28 10:35:27 204548-4 /usr/local/sbin/opensips[8838]:
ERROR:b2b_entities:server_new: failed to create new dialog structure entry
Jul 28 10:35:27 204548-4 /usr/local/sbin/opensips[8838]:
ERROR:b2b_logic:b2b_process_scenario_init: failed to create new b2b server
instance
Jul 28 10:35:31 204548-4 /usr/local/sbin/opensips[8837]:
ERROR:b2b_entities:b2b_new_dlg: no Content-Length header found!
Jul 28 10:35:31 204548-4 /usr/local/sbin/opensips[8837]:
ERROR:b2b_entities:server_new: failed to create new dialog structure entry
Jul 28 10:35:31 204548-4 /usr/local/sbin/opensips[8837]:
ERROR:b2b_logic:b2b_process_scenario_init: failed to create new b2b server
instance
On Wed, Jul 28, 2010 at 8:05 PM, Anca Vamanu a...@opensips.org wrote:
On 07/28/2010 05:12 PM, Premalatha Kuppan wrote:
Sorry. I couldn't get you.
Actually,
1. User is making initial INVITE to opensips(proxy),
2.opensips will forward it to Asterisk for IVR,
3.now after IVR, asterisk transfer the call to destination via Opensips. As
i mentioned earlier since asterisk 1.4x has no support to TCP/TLS, i want
opensips to handle the REFER request and connect the peers.
Here, if i call b2b_init_request during initial INVITE, will my problem get
solved ?
YES.
Or is that making Opensisp (Proxy) to act as well as B2BUA will not make
logic in my scenario.
On Wed, Jul 28, 2010 at 7:24 PM, Anca Vamanu a...@opensips.org wrote:
Hi,
No, it won't work without passing all the dialog messages through
opensips.
Regards,
--
Anca Vamanuwww.voice-system.ro
On 07/28/2010 04:45 PM, Premalatha Kuppan wrote:
But, from asterisk iam getting only REFER, no INVITE request. So, it emans
i can't make it to work as B2BUA in my scenario ?
On Wed, Jul 28, 2010 at 7:11 PM, Anca Vamanu a...@opensips.org wrote:
Hi,
You made the classic error due to not reading the documentation carefully
- you need to call b2b_init_request on the initial Invite( and only on it).
Do not call it on REFER.
Regards,
--
Anca Vamanuwww.voice-system.ro
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Re: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius
Check the radiusclient.conf to make sure that your dictionary is mapped to the right path. It be a good idea to check your radius log as well. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/FW-Error-when-setting-OpenSips-with-Radius-tp5342015p5345781.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] question about uac_auth and gateway authentication
Hi
I'm try to configure opensips for the following scenario:
- 'A' VoIP Phone
- 'B' My opensips proxy
- 'C' PSTN Gateway with auth
The problem is: 'A' need to authenticate for 'B', it's OK. 'A' sends
an INVITE to 'B' with auth credentials, to a public PSTN number. But i
need to authenticate to 'C'. I'm trying the uac_auth but, when i'm
running wireshark i'm seeing 'B' sending the wrong credentials (sends
'A' to 'B', not 'B' to 'C'). How can i use uac_auth? What is the the
correct way for gateway auth? If i try to replace some words in
failure_route with 'replace_all', is working so opensips processing
failure_route[2], but don't insert the proper authentication
information.
from my route script (with replaced fake host names, and passwords):
modparam(uac,credential,username:domain:password)
route {
t_on_failure(2);
t_relay( udp:ip_addr:5060 );
...
}
failure_route[2] {
uac_auth();
t_relay(udp:ip_addr:5060);
}
Czo
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Re: [OpenSIPS-Users] help, server crashes every hour
Hi Anca, This is the result : $1 = (struct mi_handler *) 0x2b659e37adc0 Yaniv -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/help-server-crashes-every-hour-tp5345496p5345924.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Load balancer retcode question on OpenSIPS 1.6.2
I am using OpenSIPS 1.6.2 and followed the tutorial
http://www.opensips.org/Resources/DocsTutLoadbalancing to use the load
balancer module.
The Tutorial use $retcode0 for Service Full reply. I get $rectcode = 1
instead of 0. What is the correct retcode load_balance(id,resource) when
resource is full?
This work for me
if ( $retcode=1 ) {
sl_send_reply(500,Service full);
exit;
}
--
View this message in context:
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Re: [OpenSIPS-Users] help, server crashes every hour
Hi Anca, This is the command result: $1 = (struct mi_handler *) 0x2b659e37adc0 Thanks, Yaniv On 07/28/2010 01:58 PM, Yaniv Vaknin wrote: Hi, I'm using opensips 1.6.2, the server handle only subscribe and notify requests. I use the mi datagram to publish new status (which I receive via external XML process). Every thing works as fine but the server keeps crashing about every hour. I tested the configuration using two different servers , one clean redhat 5.3 install and the other is gentoo. This happen on both servers. I've tested different configuration, but no matter what the server keeps crashing. this is bt form gdb : #0 0x2b659d21c836 in mi_publ_rpl_cback (hentity=value optimized out, reply=0x0) at mi_func.c:316 #1 0x2b659d00ba53 in publ_cback_func (t=value optimized out, type=value optimized out, ps=0x2b659c330420) at pua_callback.h:72 #2 0x2b659c10534b in run_trans_callbacks (type=512, trans=0x2b659e37af60, req=0x0, rpl=0x7885f8, code=value optimized out) at t_hooks.c:208 #3 0x2b659c11da69 in local_reply (t=0x2b659e37af60, p_msg=0x2b659c341d38, branch=value optimized out, msg_status=value optimized out, cancel_bitmap=0x2b659e2fe6d8) at t_reply.c:1339 #4 0x2b659c120389 in reply_received (p_msg=0x7885f8) at t_reply.c:1484 #5 0x00421448 in forward_reply (msg=0x7885f8) at forward.c:559 #6 0x00456182 in receive_msg ( buf=0x754de0 SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 192.168.167.62;branch=z9hG4bKd94.311a3611.0\r\nTo: sip:972529984...@192.168.167.62:5060 ;tag=595f1ec520b32ef8798f023cc8e6d5bc-be8e\r\nFrom: sip:972529984...@192.168.167.62 sip%3a972529984...@192.168.167.62..., len=431, rcv_info=0x7fff0f218e20) at receive.c:200 #7 0x0049a354 in udp_rcv_loop () at udp_server.c:492 #8 0x00429c0d in main (argc=3, argv=value optimized out) at main.c:818 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Load balancer retcode question on OpenSIPS 1.6.2
I figured it out.
This work all the time
if ( uri=~sip:92[1-9][0-...@.* ) {
load_balance(27,white);
} else if ( uri=~sip:3392[1-9][0-...@.* ) {
load_balance(27,grey); #
}
if ( $retcode 0 ) {
sl_send_reply(500,Service full);
exit;
}
This work sometime
if ( uri=~sip:92[1-9][0-...@.* ) {
load_balance(27,white);
}
if ( uri=~sip:3392[1-9][0-...@.* ) {
load_balance(27,grey); #
}
if ( $retcode 0 ) {
sl_send_reply(500,Service full);
exit;
}
--
View this message in context:
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[OpenSIPS-Users] Location info when using OpenSIPS as outbound proxy
Hi, using opensips 1.6.2. We are using Opensips as an outbound proxy which also does local routing of 302 responses. IF we have a client UAC1 which registers with a provider using OpenSIPS only as an outbound proxy is it possible to save the location info of UAC1, reason being if there is a call to UAC2 for instance from the provider which is redirected with 302, then our Opensips can handle the redirection because it knows where UAC1 is without having it to go back to the provider's proxy server. Maybe we need to do 2 registrations one with OpenSIPS and one with the provider but if we can do this with just the one it would be a neat solution. Secondly if we do save the location info of UAC1 at the opensips proxy, what url would need to go in contact field of the 302, would it be u...@provider.com - opensips needs soemthing ot be able to send to UAC1 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Mediaproxy not updating with new SDPs
No need to apologize. I'm intruding on your time. Just wanted to make sure I didn't sit in a spam filter for a week or anything. : I found I had the same problem if I used use_media_proxy in the invite and reinvite. However, since I am assured of the reinvite on every call from this source I just set a dialog flag in the invite to indicate I needed media proxy and waited to turn it on for the first time in the reinvite. That is doing the job for now. I suspect most users behind a Belkin home wifi router will cause this problem. On the particular model we have been working with, there seems to be no way to turn off the SIP ALG so it is particularly irritating. Anyway, I have a working solution and if you find something that can fix this while still using engage_media_proxy() that will be even better. Richard On Jul 28, 2010, at 8:16 AM, Saúl Ibarra Corretgé wrote: On 28/07/10 14:01, Richard Revels wrote: Saul, Could you verify you got the trace file I sent? I mailed it directly to you so I wouldn't have to worry about obfuscation on the IP addresses and want to make sure I didn't get caught in a spam filter or anything. I didn't have time yet, but I got the email, sorry for not ack-ing it :-) Will look into it. Regards, -- Saúl Ibarra Corretgé AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] proxy_authorize(,subscriber) bug ??
Hi Bogdan,
My authentication route is as follow,
if (!allow_trusted()) {
if (!proxy_authorize(,subscriber)) {
if(!lookup(location) ){
proxy_challenge(,0);
exit;
}
} else if (!check_from()) {
sl_send_reply(403, Spoofed From-URI detected);
xlog(L_INFO,Spoofed From-URI detected ! from -- $fu -- IP
$si PORT:$sp);
exit;
}
if(is_present_hf(Proxy-Authorization)){
consume_credentials();
}
}
This route is before the dispatch route (t_relay())
I think retransmitted INVITEs get block by this route so If I use the
t_check_trans() as follow will I able to absorb the retransmitted INVITE ?
if (!allow_trusted()) {
if (!proxy_authorize(,subscriber)) {
if(!lookup(location) ! t_check_trans() ){
proxy_challenge(,0);
exit;
}
} else if (!check_from()) {
sl_send_reply(403, Spoofed From-URI detected);
xlog(L_INFO,Spoofed From-URI detected ! from -- $fu -- IP
$si PORT:$sp);
exit;
}
if(is_present_hf(Proxy-Authorization)){
consume_credentials();
}
}
modparam(auth, disable_nonce_check, 1) setting this is not a good idea i
think.
thanks
From: Pasan Meemaduma pasan...@ymail.com
To: OpenSIPS users mailling list users@lists.opensips.org
Sent: Monday, July 12, 2010 16:46:26
Subject: Re: [OpenSIPS-Users] proxy_authorize(,subscriber) bug ??
Hi Bogdan,
Thanks for the quick reply,
What I now suspect is the security mechanism for stale nonces introduced in
later 1.4 causing this. The identical configuration works fine with opensips 1.4
This problem started to appear after I upgrade server from openser to opensips
about a month ago.
Loosing registration is the most worst problem since its affecting incoming
calls.
For the moment what I did was add the following in my opensips.cfg after going
through the mailing list archives.
modparam(auth, disable_nonce_check, 1)
As I understood opensips reject nonce which is used before even if it send with
correct credentials. This could be the problem that Re-INVITEs get 407 .
I can't do much changes to observe more debuging information like setting set
debug =6 as this is a production server.
I'm going to apply the new setting modparam(auth, disable_nonce_check, 1)
tomorrow on our offpeak time and see whether it will resolve the problem.
I'll get back to here tomorrow with the results.
From: Bogdan-Andrei Iancu bog...@voice-system.ro
To: OpenSIPS users mailling list users@lists.opensips.org
Sent: Monday, July 12, 2010 15:46:18
Subject: Re: [OpenSIPS-Users] proxy_authorize(,subscriber) bug ??
Hi Pasan,
first, for non-REGISTER requests use only the proxy_() functions.
For debugging the failure, try:
1) print the return code of the proxy_authorize() (use $retcode) - see
http://www.opensips.org/html/docs/modules/1.6.x/auth_db.html#id228340
2) set debug =6 and post the log corresponding to the INVITE processing .
Regards,
Bogdan
Pasan Meemaduma wrote:
Hi All,
I'm having trouble with my authentication routine with opensips 1.5
I'm currently using opensips 1.5.3-1
And there are lot of voip equipments using this production server.
problem is that sometimes for some sip clients
proxy_authorize(,subscriber) returns false even with correct
credentials.
basically most of the times this happens to Re-INVITEs in a dialogue
(messages with Proxy-Authorization Header).
This is causing in progress calls being failed. sip client gives up
when it changes again.
And another problem is with www_authorize(, subscriber)
It has the same problem returns false even with correct credentials.
and this happens randomly so , its hard to figure out why .
does any one else having problem with simillar issues with using these
routines ?
Is it a bug in these routines ?
Is there a new release for 1.5 branch which has fixed this sort of a
problem.
any help on this would be very appreciated.
currently server has more than 8000 entries in location table at any
given time and handles more than 3000 calls per day.
following is one such sip trace that i got from a call
Even the re- INVITE has correct Proxy-Authorization header present
opensips change it again.
U 2010/06/24 16:03:40.466974 y.y.y.y:5060 - x.x.x.x:5060
INVITE sip:1234567...@x.x.x.x SIP/2.0.
To: sip:1234567...@x.x.x.x.
From: abcdefgh sip:abcde...@x.x.x.x;tag=252070.
Call-ID: 44460...@192.168.1.20.
CSeq: 5 INVITE.
Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK155910d13;rport.
Allow: ACK,BYE,CANCEL,INVITE,INFO,NOTIFY,OPTIONS,PRACK,REFER,UPDATE.
Contact: sip:abcde...@192.168.1.20:5060.
Supported: replaces,precondition.
Accept: application/sdp,application/cpim-pidf+xml.
Expires: 240.
