Re: [OpenSIPS-Users] Strange errors forwarding requests
The destination (in this case) is the 1st server in the loadbalancer list (as
there are no other calls).
I've upgraded this machine to ubuntu 10 (from 8) and started getting Connection
Tracking drop messages in my
syslog. I've disabled connection tracking and the issue hasn't appeared since...
Op 5-5-2010 12:24, Bogdan-Andrei Iancu schreef:
Hi Erik,
have you tried to print the destination of the requests that fail?
regards,
Bogdan
Erik Versaevel wrote:
Hi All,
I attempted an migration last night (from our current environment to this
new setup) but i ran into this
problem as soon as i tried to make some test calls, funny thing is i can't
get it reproduced :/ Any clues
on how to debug this any further?
Kind regards,
Erik
Op 27-4-2010 15:17, Erik Versaevel schreef:
Hi all,
I'm building a setup in which opensips is acting as registar for my
endpoints and loadbalancing
calls made by those endpoint over an cluster of asterisk machines. (so that
if we need more asterisk
power, we just have to add another destination to the loadbalancer module)
Opensips is listening on multiple IP addresses and uses the loadbalancer
module to poll my asterisk
machines and select the destination.
My problem is that every now and then opensips fails to forward an invite
to my asterisk cluster and
generates
ERROR:core:udp_send: sendto(sock,0x77b81280,1353,0,0x77b81b04,16):
Operation not permitted(1)
there is some iptables filtering on this machine, however it is not showing
drops in the logfile (and it keeps
occuring even without any iptable rules).
I tried stracing opensips but all i get is:
opensipstrace.7423:sendto(6, INVITE
sip:e164_dst_phone...@opensips_ip_address SIP/2.0
Record-Route:
sip:OPENSIPS_IP_ADDRESS;lr=on;ftag=AI05ED431A05432EB8;nat=yes;did=fd6.e1f16fe3;vsf=AAMIBgl3AggLFgF5HAAFGhwBHzE3NC44MQ--
Via: SIP/2.0/UDP OPENSIPS_IP_ADDRESS;branch=z9hG4bK3177.1e0e38b7.0
Via: SIP/2.0/UDP
192.168.178.44:5060;received=CPE_IP_ADDRESS;rport=61008;branch=z9hG4bK2010Apr222938466E164_DST_PHONE_NR
To: sip:e164_dst_phone...@opensips_ip_address
From: \3961\ sip:3...@opensips_ip_address;tag=AI05ED431A05432EB8
Call-ID: aif001c45e85f79...@192.168.178.44
CSeq: 2 INVITE
Max-Forwards: 69
Contact: sip:e164phone...@cpe_ip_address:61008;line=AIF8F01E8DF866D7CB
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER
Allow-Events: dialog,message-summary
P-Preferred-Identity: sip:e164phone...@opensips_ip_address
Privacy: none
User-Agent: SomeStrangeDude
Content-Type: application/sdp
Content-Length: 324
I-FromDisp: null
I-FromUri: E164PHONE_NR
I-CustId: 3961
v=0
o=intelligate 1133701155 1133701155 IN IP4 192.168.178.44
s=call
c=IN IP4 CPE_IP_ADDRESS
t=0 0
m=audio 5004 RTP/AVP 18 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
a=direction:active
a=oldmediaip:192.168.178.44
, 1253, 0, {sa_family=AF_INET, sin_port=htons(5060),
sin_addr=inet_addr(ASTERISK_IP_ADDRESS)}, 16) = -1 EPERM (Operation not
permitted)
I also use the uac_replace_from() to mangle the from header so asterisk
uses the correct user/peer/client to connect the call (codec/dialplan etc).
I'm having trouble reproducing the error as it's not allways occuring, the
errors i straced where mainly the initial invite towards my asterisk
cluster and a few 200 OK's which didn't get processed correctly.
Any clues on how to debug this further?
Kind regards,
Erik Versaevel
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Re: [OpenSIPS-Users] Rtpproxy behind the NAT
I have done no recording so I can't speak to those formats used. -dg On Thu, May 6, 2010 at 6:33 AM, Indiver nehru.i...@gmail.com wrote: Thanks for your response Daniel. I included the patch and it's started working !. Now i'm testing the rtpproxy with different scenarios. Both clients outside NAT , behind the NAT etc. I included recording option also,but the concern part about the rtpproxy is that after every call we have to manually convert recorded RTP files in to single wave file using rtpbreak and sox software. Does there any of chance of direct wav file creation when recording option enabled with out doing all this stuff. Regards, Nehru. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Rtpproxy-behind-the-NAT-tp5008041p5012493.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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPS failover
Hi, try using any software specialized in monitoring or testing applications (monit, nagios, sipsak, etc) to detect the failure of opensips service and to trigger heartbeat switching. Regards, Bogdan rajib deka wrote: Hi Bogdan, I have seen that heartbeat is supporting box level failure but not service level failure. Means if the box itself is down heartbeat will activate the VIP of back up server, but if OpenSIPS service is down heartbeat will not activate backup VIP. I have checked with ldirectord also but I am not getting any good result. Is there anyway to handle this service level failure ? Please let me know as we have implemented OpenSIPS as our core routing agent for the whole enterprise. Regards Rajib On Wed, May 5, 2010 at 2:07 PM, rajib deka raji...@gmail.com mailto:raji...@gmail.com wrote: Thanks Bogdan. I am trying to achieve the same using hearbeat and almost done. Regards Rajib On Wed, May 5, 2010 at 2:03 PM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Hi Rajib, OpenSIPS does not have anything builtin for HA - this is something you achieve using other software (like vrrp, heartbeat, etc). But OpenSIPS provides some mechanism for syncronizing the active and the backup instance - a way of replicating data : like t_replicate() function in TM module that you can use for replicating REGISTERs to the backup. Regards, Bogdan rajib deka wrote: Hi List, Cn some some please explain me how OpenSIPS supports HA and redundancy? I have not found anything on this on the site. I have an implementation with virtual ip for HA and its working fine. I just want to know is there any internal module or configuration with OpenSIPS using which we can achieve HA and redundancy. -- Rajib Deka Software Engineer Servion Global Solution Chennai, India Mobile No: + 91 80157 09130 ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu 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 -- Rajib Deka Software Engineer Servion Global Solution Chennai, India Mobile No: + 91 80157 09130 -- Rajib Deka Software Engineer Servion Global Solution Chennai, India Mobile No: + 91 80157 09130 ___ 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
Re: [OpenSIPS-Users] Outbound gateway selection
Hi Alejandro,
Alejandro Recarey wrote:
I am using OpenSIPS to load balance incoming calls between my asterisk
boxes, with great results! The LB module is perfect for this.
I am currently configuring OpenSIPS to also act as an outbound proxy
to my asterisk boxes. This is because some VoIP providers want you to
send them all calls from a single IP address.
However, I have a problem with the setup. The authentification of the
incoming calls is done with the permission.so module and works
perfectly. However, because the destination URI always has the IP
address of my OpenSIPS box, I have no way to distinguish what carrier
(IP address) I should actually be sending the call to.
The only solution I have found so far is to add an internal prefix
(for example 1# for carrier 1 with IP X, 2# for carrier 2 with IP Y
etc), this way I can match the uri like
if(uri=~^sip:1#[0-9]*+@){
rewritehostport(x.x.x.x);
route(1)
}
if(uri=~^sip:2#[0-9]*+@){
rewritehostport(y.y.y.y);
route(1)
}
However, I have to strip this prefix when forwarding the invite to my
carrier, because my carrier will not accept the prefix. This is easily
done with the dialplan module,
I assume you do the change over RURI only (removing this prefix).
but I also have to _add it back_ when
forwarding RE-INVITES, TRYING, ACK or other messages that come from my
carrier, so that asterisk does not get confused when to: or uri tags
do not match.
asterisk should not be confused at all - according to SIP, proxies on
the way may change the RURI without breaking anything. The idea is to
change only the RURI (for original INVITE), so that the SIP dialog will
be consistent for its whole duration.
Regards,
Bogdan
This seems like an ugly solution, or maybe I don't know
how to set it up?
What is the best way to setup this scenario?
Thank you for your help!
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--
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www.voice-system.ro
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Re: [OpenSIPS-Users] Strange errors forwarding requests
So, after all, it was a network layer configuration issue... :)
Regards,
Bogdan
Erik Versaevel wrote:
The destination (in this case) is the 1st server in the loadbalancer list (as
there are no other calls).
I've upgraded this machine to ubuntu 10 (from 8) and started getting
Connection Tracking drop messages in my
syslog. I've disabled connection tracking and the issue hasn't appeared
since...
Op 5-5-2010 12:24, Bogdan-Andrei Iancu schreef:
Hi Erik,
have you tried to print the destination of the requests that fail?
regards,
Bogdan
Erik Versaevel wrote:
Hi All,
I attempted an migration last night (from our current environment to this
new setup) but i ran into this
problem as soon as i tried to make some test calls, funny thing is i can't
get it reproduced :/ Any clues
on how to debug this any further?
Kind regards,
Erik
Op 27-4-2010 15:17, Erik Versaevel schreef:
Hi all,
I'm building a setup in which opensips is acting as registar for my
endpoints and loadbalancing
calls made by those endpoint over an cluster of asterisk machines. (so
that if we need more asterisk
power, we just have to add another destination to the loadbalancer module)
Opensips is listening on multiple IP addresses and uses the loadbalancer
module to poll my asterisk
machines and select the destination.
My problem is that every now and then opensips fails to forward an invite
to my asterisk cluster and
generates
ERROR:core:udp_send: sendto(sock,0x77b81280,1353,0,0x77b81b04,16):
Operation not permitted(1)
there is some iptables filtering on this machine, however it is not
showing drops in the logfile (and it keeps
occuring even without any iptable rules).
I tried stracing opensips but all i get is:
opensipstrace.7423:sendto(6, INVITE
sip:e164_dst_phone...@opensips_ip_address SIP/2.0
Record-Route:
sip:OPENSIPS_IP_ADDRESS;lr=on;ftag=AI05ED431A05432EB8;nat=yes;did=fd6.e1f16fe3;vsf=AAMIBgl3AggLFgF5HAAFGhwBHzE3NC44MQ--
Via: SIP/2.0/UDP OPENSIPS_IP_ADDRESS;branch=z9hG4bK3177.1e0e38b7.0
Via: SIP/2.0/UDP
192.168.178.44:5060;received=CPE_IP_ADDRESS;rport=61008;branch=z9hG4bK2010Apr222938466E164_DST_PHONE_NR
To: sip:e164_dst_phone...@opensips_ip_address
From: \3961\ sip:3...@opensips_ip_address;tag=AI05ED431A05432EB8
Call-ID: aif001c45e85f79...@192.168.178.44
CSeq: 2 INVITE
Max-Forwards: 69
Contact: sip:e164phone...@cpe_ip_address:61008;line=AIF8F01E8DF866D7CB
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER
Allow-Events: dialog,message-summary
P-Preferred-Identity: sip:e164phone...@opensips_ip_address
Privacy: none
User-Agent: SomeStrangeDude
Content-Type: application/sdp
Content-Length: 324
I-FromDisp: null
I-FromUri: E164PHONE_NR
I-CustId: 3961
v=0
o=intelligate 1133701155 1133701155 IN IP4 192.168.178.44
s=call
c=IN IP4 CPE_IP_ADDRESS
t=0 0
m=audio 5004 RTP/AVP 18 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
a=direction:active
a=oldmediaip:192.168.178.44
, 1253, 0, {sa_family=AF_INET, sin_port=htons(5060),
sin_addr=inet_addr(ASTERISK_IP_ADDRESS)}, 16) = -1 EPERM (Operation not
permitted)
I also use the uac_replace_from() to mangle the from header so asterisk
uses the correct user/peer/client to connect the call (codec/dialplan etc).
I'm having trouble reproducing the error as it's not allways occuring, the
errors i straced where mainly the initial invite towards my asterisk
cluster and a few 200 OK's which didn't get processed correctly.
Any clues on how to debug this further?
Kind regards,
Erik Versaevel
___
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http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Erik Versaevel
--
Bogdan-Andrei Iancu
www.voice-system.ro
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Re: [OpenSIPS-Users] Strange errors forwarding requests
That might be :)
I'm now running into problems with the dialog module (which i use to limit
concurrent calls).
Calls seem to stick in the dialog module (thus denying additional calls) while
the endpoint isn't
listing the same amount of calls :/
Regards,
Erik
Op 6-5-2010 11:02, Bogdan-Andrei Iancu schreef:
So, after all, it was a network layer configuration issue... :)
Regards,
Bogdan
Erik Versaevel wrote:
The destination (in this case) is the 1st server in the loadbalancer
list (as there are no other calls).
I've upgraded this machine to ubuntu 10 (from 8) and started getting
Connection Tracking drop messages in my
syslog. I've disabled connection tracking and the issue hasn't
appeared since...
Op 5-5-2010 12:24, Bogdan-Andrei Iancu schreef:
Hi Erik,
have you tried to print the destination of the requests that fail?
regards,
Bogdan
Erik Versaevel wrote:
Hi All,
I attempted an migration last night (from our current environment to
this new setup) but i ran into this
problem as soon as i tried to make some test calls, funny thing is i
can't get it reproduced :/ Any clues
on how to debug this any further?
Kind regards,
Erik
Op 27-4-2010 15:17, Erik Versaevel schreef:
Hi all,
I'm building a setup in which opensips is acting as registar for my
endpoints and loadbalancing
calls made by those endpoint over an cluster of asterisk machines.
(so that if we need more asterisk
power, we just have to add another destination to the loadbalancer
module)
Opensips is listening on multiple IP addresses and uses the
loadbalancer module to poll my asterisk
machines and select the destination.
My problem is that every now and then opensips fails to forward an
invite to my asterisk cluster and
generates
ERROR:core:udp_send:
sendto(sock,0x77b81280,1353,0,0x77b81b04,16): Operation not
permitted(1)
there is some iptables filtering on this machine, however it is not
showing drops in the logfile (and it keeps
occuring even without any iptable rules).
I tried stracing opensips but all i get is:
opensipstrace.7423:sendto(6, INVITE
sip:e164_dst_phone...@opensips_ip_address SIP/2.0
Record-Route:
sip:OPENSIPS_IP_ADDRESS;lr=on;ftag=AI05ED431A05432EB8;nat=yes;did=fd6.e1f16fe3;vsf=AAMIBgl3AggLFgF5HAAFGhwBHzE3NC44MQ--
Via: SIP/2.0/UDP OPENSIPS_IP_ADDRESS;branch=z9hG4bK3177.1e0e38b7.0
Via: SIP/2.0/UDP
192.168.178.44:5060;received=CPE_IP_ADDRESS;rport=61008;branch=z9hG4bK2010Apr222938466E164_DST_PHONE_NR
To: sip:e164_dst_phone...@opensips_ip_address
From: \3961\
sip:3...@opensips_ip_address;tag=AI05ED431A05432EB8
Call-ID: aif001c45e85f79...@192.168.178.44
CSeq: 2 INVITE
Max-Forwards: 69
Contact:
sip:e164phone...@cpe_ip_address:61008;line=AIF8F01E8DF866D7CB
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER
Allow-Events: dialog,message-summary
P-Preferred-Identity: sip:e164phone...@opensips_ip_address
Privacy: none
User-Agent: SomeStrangeDude
Content-Type: application/sdp
Content-Length: 324
I-FromDisp: null
I-FromUri: E164PHONE_NR
I-CustId: 3961
v=0
o=intelligate 1133701155 1133701155 IN IP4 192.168.178.44
s=call
c=IN IP4 CPE_IP_ADDRESS
t=0 0
m=audio 5004 RTP/AVP 18 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
a=direction:active
a=oldmediaip:192.168.178.44
, 1253, 0, {sa_family=AF_INET, sin_port=htons(5060),
sin_addr=inet_addr(ASTERISK_IP_ADDRESS)}, 16) = -1 EPERM
(Operation not permitted)
I also use the uac_replace_from() to mangle the from header so
asterisk uses the correct user/peer/client to connect the call
(codec/dialplan etc).
I'm having trouble reproducing the error as it's not allways
occuring, the errors i straced where mainly the initial invite
towards my asterisk
cluster and a few 200 OK's which didn't get processed correctly.
Any clues on how to debug this further?
Kind regards,
Erik Versaevel
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Erik Versaevel
Erik Versaevel
--
Core Network Engineer
Infopact Network Solutions
Hoogvlietsekerkweg 170
3194 AM Rotterdam Hoogvliet
Telefoon+31 (0)88 - 4636777
Fax +31 (0)88 - 4636799
Mobile +31 (0)6 - 6070
e.versae...@infopact.nl
www.infopact.nl
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Re: [OpenSIPS-Users] Strange errors forwarding requests
What is the state of the dialog stuck in memory? (use dlg_list MI
command to see them).
Also check that the sequential requests (ACK, re-INVITE, BYE) do go
through your proxy (via loose_route).
Regards,
Bogdan
Erik Versaevel - InfoPact Netwerkdiensten wrote:
That might be :)
I'm now running into problems with the dialog module (which i use to limit
concurrent calls).
Calls seem to stick in the dialog module (thus denying additional calls)
while the endpoint isn't
listing the same amount of calls :/
Regards,
Erik
Op 6-5-2010 11:02, Bogdan-Andrei Iancu schreef:
So, after all, it was a network layer configuration issue... :)
Regards,
Bogdan
Erik Versaevel wrote:
The destination (in this case) is the 1st server in the loadbalancer
list (as there are no other calls).
I've upgraded this machine to ubuntu 10 (from 8) and started getting
Connection Tracking drop messages in my
syslog. I've disabled connection tracking and the issue hasn't
appeared since...
Op 5-5-2010 12:24, Bogdan-Andrei Iancu schreef:
Hi Erik,
have you tried to print the destination of the requests that fail?
regards,
Bogdan
Erik Versaevel wrote:
Hi All,
I attempted an migration last night (from our current environment to
this new setup) but i ran into this
problem as soon as i tried to make some test calls, funny thing is i
can't get it reproduced :/ Any clues
on how to debug this any further?
Kind regards,
Erik
Op 27-4-2010 15:17, Erik Versaevel schreef:
Hi all,
I'm building a setup in which opensips is acting as registar for my
endpoints and loadbalancing
calls made by those endpoint over an cluster of asterisk machines.
(so that if we need more asterisk
power, we just have to add another destination to the loadbalancer
module)
Opensips is listening on multiple IP addresses and uses the
loadbalancer module to poll my asterisk
machines and select the destination.
My problem is that every now and then opensips fails to forward an
invite to my asterisk cluster and
generates
ERROR:core:udp_send:
sendto(sock,0x77b81280,1353,0,0x77b81b04,16): Operation not
permitted(1)
there is some iptables filtering on this machine, however it is not
showing drops in the logfile (and it keeps
occuring even without any iptable rules).
I tried stracing opensips but all i get is:
opensipstrace.7423:sendto(6, INVITE
sip:e164_dst_phone...@opensips_ip_address SIP/2.0
Record-Route:
sip:OPENSIPS_IP_ADDRESS;lr=on;ftag=AI05ED431A05432EB8;nat=yes;did=fd6.e1f16fe3;vsf=AAMIBgl3AggLFgF5HAAFGhwBHzE3NC44MQ--
Via: SIP/2.0/UDP OPENSIPS_IP_ADDRESS;branch=z9hG4bK3177.1e0e38b7.0
Via: SIP/2.0/UDP
192.168.178.44:5060;received=CPE_IP_ADDRESS;rport=61008;branch=z9hG4bK2010Apr222938466E164_DST_PHONE_NR
To: sip:e164_dst_phone...@opensips_ip_address
From: \3961\
sip:3...@opensips_ip_address;tag=AI05ED431A05432EB8
Call-ID: aif001c45e85f79...@192.168.178.44
CSeq: 2 INVITE
Max-Forwards: 69
Contact:
sip:e164phone...@cpe_ip_address:61008;line=AIF8F01E8DF866D7CB
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER
Allow-Events: dialog,message-summary
P-Preferred-Identity: sip:e164phone...@opensips_ip_address
Privacy: none
User-Agent: SomeStrangeDude
Content-Type: application/sdp
Content-Length: 324
I-FromDisp: null
I-FromUri: E164PHONE_NR
I-CustId: 3961
v=0
o=intelligate 1133701155 1133701155 IN IP4 192.168.178.44
s=call
c=IN IP4 CPE_IP_ADDRESS
t=0 0
m=audio 5004 RTP/AVP 18 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
a=direction:active
a=oldmediaip:192.168.178.44
, 1253, 0, {sa_family=AF_INET, sin_port=htons(5060),
sin_addr=inet_addr(ASTERISK_IP_ADDRESS)}, 16) = -1 EPERM
(Operation not permitted)
I also use the uac_replace_from() to mangle the from header so
asterisk uses the correct user/peer/client to connect the call
(codec/dialplan etc).
I'm having trouble reproducing the error as it's not allways
occuring, the errors i straced where mainly the initial invite
towards my asterisk
cluster and a few 200 OK's which didn't get processed correctly.
Any clues on how to debug this further?
Kind regards,
Erik Versaevel
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Erik Versaevel
--
Bogdan-Andrei Iancu
www.voice-system.ro
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[OpenSIPS-Users] Sipp to test OPENSIPS
Hi everybody, I try to generate SIP traffic to test the power of OpenSIPS but I don't know how to do a simple test. I did : /sipp -sn uas and ./sipp -sn uac 192.168.0.190 (ip of my opensips) but all calls are failed. I have not understood yet how to do the test yet. Can someone help me? Thanks Samy. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Sipp-to-test-OPENSIPS-tp5013501p5013501.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] Nat Problem
Hi,
I've configured OpenSIPs using Nathelper module and rtpproxy. the problem
I'm facing is when I try to register my softphone, it got registered but as
I issue the command opensipsctl ul show, in contact header the IP is private
not public. The configuration of OpenSIPs is listed down below;
loadmodule dispatcher.so
loadmodule avpops.so
loadmodule permissions.so
loadmodule aaa_radius.so
loadmodule auth_aaa.so
#loadmodule auth_diameter.so
loadmodule nathelper.so
#Settings For
Radius-
#modparam(auth_diameter, diameter_client_host, localhost)
modparam(aaa_radius,
radius_config,/usr/local/etc/radiusclient-ng/radiusclient.conf)
modparam(acc, aaa_url,
radius:/usr/local/etc/radiusclient-ng/radiusclient.conf)
modparam(acc, aaa_flag, 2)
modparam(acc, aaa_missed_flag, 3)
modparam(acc, aaa_extra,User-Name=$Au; \
Calling-Station-Id=$from; \
Called-Station-Id=$to; \
Sip-Translated-Request-URI=$ruri; \
Sip-RPid=$avp(s:rpid); \
Source-IP=$si; \
Source-Port=$sp; \
Canonical-URI=$avp(s:can_uri); \
Billing-Party=$avp(s:billing_party); \
Divert-Reason=$avp(s:divert_reason); \
X-RTP-Stat=$hdr(X-RTP-Stat); \
Contact=$hdr(contact); \
Event=$hdr(event); \
SIP-Proxy-IP=$avp(s:sip_proxy_ip); \
ENUM-TLD=$avp(s:enum_tld))
modparam(auth_aaa,aaa_url,radius:/usr/local/etc/radiusclient-ng/radiusclient.conf)
modparam(auth, rpid_prefix, sip:)
modparam(auth, rpid_suffix, @77.66.2.137;screen=yes;privacy=off)
#modparam(auth, rpid_suffix, @203.215.179.54;screen=yes;privacy=off)
modparam(auth, rpid_avp, $avp(s:rpid))
#modparam(uri,service_type,10)
# - setting module-specific parameters ---
modparam(dispatcher, db_url, mysql://opensips:opensip...@localhost
/opensips)
modparam(permissions, db_url, mysql://opensips:opensip...@localhost
/opensips)
#- setting NAT module parameters -
modparam(nathelper,ping_nated_only,1)
modparam(nathelper, natping_interval, 30)
modparam(nathelper,natping_processes,1)
modparam(nathelper,rtpproxy_sock,udp:127.0.0.1:7890)
#modparam(nathelper,rtpproxy_sock, )
modparam(nathelper,received_avp,$avp(i:42))
#modparam(nathelper, sipping_bflag, 7)
modparam(usrloc, nat_bflag, 6)
route{
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483,Too Many Hops);
exit;
}
#NAT detection
log(# Go to Route 3 for NAT
Detection #);
route(3);
if (has_totag()) {
if (loose_route()) {
if (is_method(BYE)) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction
fails
} else if (is_method(INVITE)) {
record_route();
}
route(1);
} else {
if ( is_method(ACK) ) {
if ( t_check_trans() ) {
# non loose-route, but stateful ACK;
must be an ACK after
# a 487 or e.g. 404 from upstream
server
t_relay();
exit;
} else {
# ACK without matching transaction
-
# ignore and discard
exit;
}
}
sl_send_reply(404,Not here);
}
exit;
}
#initial requests
# CANCEL processing
if (is_method(CANCEL))
{
if (t_check_trans())
t_relay();
exit;
}
t_check_trans();
# preloaded route checking
if (loose_route()) {
xlog(L_ERR,
Attempt to route with preloaded Route's
[$fu/$tu/$ru/$ci]);
if (!is_method(ACK))
sl_send_reply(403,Preload Route denied);
exit;
}
# record routing
if (!is_method(REGISTER|MESSAGE))
record_route();
$avp(s:checksrc) = check_source_address(0);
log(###\n);
Re: [OpenSIPS-Users] OpenSIPS failover
Hi Rajib, I have several customers using opensips in HA, and I use in all of then Keepalived (instead of Heartbeat), and the Sipsak to test the service level, so Sipsak send an OPTIONS request to Opensips and if does not receive the answer, Sipsak trigger the failover in the Keepalived. In other words we implemented a custom SIP Check plugin to keepalived using Sipsak. I think you can try the same think with Heartbeat. Let me know if you need some help in this directions. Best regards, Antonio Em 06/05/2010 05:46, Bogdan-Andrei Iancu bog...@voice-system.roescreveu: Hi, try using any software specialized in monitoring or testing applications (monit, nagios, sipsak, etc) to detect the failure of opensips service and to trigger heartbeat switching. Regards, Bogdan rajib deka wrote: Hi Bogdan, I have seen that heartbeat is supporting box level failure but not service level... mailto:raji...@gmail.com wrote: Thanks Bogdan. I am trying to achieve the same using he... bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Hi Rajib, ... Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- ... 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 -- Rajib Dek... ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Sipp to test OPENSIPS
Hi Samoh, It could be because your calls are not allowed at the Opensips, the default uri of the sipp is something like service@ope.nsi.ps.ip. try to Ngrep it to see the uri and allow it at opensips or change at sipp. Regards. - - - Andrey Cassemiro (11) 6343-0411 Date: Thu, 6 May 2010 03:27:26 -0700 From: dahmani.s...@gmail.com To: users@lists.opensips.org Subject: [OpenSIPS-Users] Sipp to test OPENSIPS Hi everybody, I try to generate SIP traffic to test the power of OpenSIPS but I don't know how to do a simple test. I did : /sipp -sn uas and ./sipp -sn uac 192.168.0.190 (ip of my opensips) but all calls are failed. I have not understood yet how to do the test yet. Can someone help me? Thanks Samy. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Sipp-to-test-OPENSIPS-tp5013501p5013501.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 _ VOCÊ PODE TER 25 GB GRATUITOS PARA ARMAZENAR SEUS ARQUIVOS NA WEB. VEJA AQUI COMO. http://www.windowslive.com.br/public/product.aspx/view/1?cname=skydriveocid=Hotmail:MSN:Messenger:Tagline:1x1:skydrive:-___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Sipp to test OPENSIPS
Hi Andrey, I use my opensips without authentification, I used ngrep and I noticed that it try to join serv...@192.168.0.190 but the opensips answer don't found it, I don't know if I must start another sipp process which will be the serv...@192.168.0.190 ??! Thanks. Sam. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Sipp-to-test-OPENSIPS-tp5013501p5014016.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
Re: [OpenSIPS-Users] Nat Problem
Hi Ahmed, check the following things: 1) you do fix_nated_register() before save(location) 2) the received_avp param has the same value in registrar and nathelper module 3) you configured the nat_bflag param in usrloc module and you are setting it before save(location) Regards, Bogdan Ahmed Munir wrote: Hi, I've configured OpenSIPs using Nathelper module and rtpproxy. the problem I'm facing is when I try to register my softphone, it got registered but as I issue the command opensipsctl ul show, in contact header the IP is private not public. The configuration of OpenSIPs is listed down below; ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] drouting crash in 1.6.2
Hi, I am just moving to 1.6.2 and it works great except for the drouting module crashes out under load. I saw this happening before in the forum but never saw a resolution. Is there a fix for this? The same config works under 1.5.1. When the crash happens the threads just go away one by one and leave no core. Here is an entry from the log: /usr/local/opensips/sbin/opensips[14398]: Memory status (pkg): /usr/local/opensips/sbin/opensips[14398]: fm_status (0x81b3d60): /usr/local/opensips/sbin/opensips[14398]: heap size= 1048576 /usr/local/opensips/sbin/opensips[14398]: used= 111464, used+overhead=136132, free=937112 /usr/local/opensips/sbin/opensips[14398]: max used (+overhead)= 154524 /usr/local/opensips/sbin/opensips[14398]: dumping free list: /usr/local/opensips/sbin/opensips[14398]: hash = 2049 fragments no.: 1, unused: 0 bucket size: 16384 - 32768 (first 21744) /usr/local/opensips/sbin/opensips[14398]: hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 915368) /usr/local/opensips/sbin/opensips[14398]: TOTAL: 2 free fragments = 937112 free bytes /usr/local/opensips/sbin/opensips[14398]: TOTAL: 937112 large bytes /usr/local/opensips/sbin/opensips[14398]: TOTAL: 12 overhead /usr/local/opensips/sbin/opensips[14398]: - -- -- Jim Wiegand --- Explicit coquina que est optima medicina. -- -- Jim Wiegand --- Explicit coquina que est optima medicina. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Stun Module - OpenSIPS won't start
I just compiled the STUN module on Debian 5.0.4 x64 and added the following stun parameters to my config port=5060 /* uncomment and configure the following line if you want opensips to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:173.*.*.134:5060 #** # - stun params - modparam(stun,primary_ip,173.*.*.134) modparam(stun,primary_port,5060) When I try and start OpenSIPS it fails and I see the following errors in syslog. Any idea whats up? May 6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]: DBG:core:init_mod: initializing module stun May 6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]: DBG:core:grep_sock_info: checking if host==us: 14==14 [173.*.*.134] == [173.*.*.134] May 6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]: DBG:core:grep_sock_info: checking if port 5060 matches port 5060 May 6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]: DBG:core:grep_sock_info: checking if host==us: 14==14 [173.*.*.134] == [173.*.*.134] May 6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]: DBG:core:grep_sock_info: checking if port 5060 matches port 3478 May 6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]: DBG:stun:stun_mod_init: socketfd2 not found May 6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]: DBG:core:grep_sock_info: checking if host==us: 13==14 [192.168.2.143] = = [173.*.*.134] May 6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]: DBG:core:grep_sock_info: checking if port 5060 matches port 5060 May 6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]: DBG:stun:stun_mod_init: socketfd3 not found May 6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]: DBG:core:grep_sock_info: checking if host==us: 13==14 [192.168.2.143] = = [173.*.*.134] May 6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]: DBG:core:grep_sock_info: checking if port 5060 matches port 3478 May 6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]: DBG:stun:stun_mod_init: socketfd4 not found May 6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]: DBG:stun:stun_mod_init: stun init failed May 6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]: ERROR:core:init_mod: failed to initialize module stun May 6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]: ERROR:core:main: error while initializing modules -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Stun-Module-OpenSIPS-won-t-start-tp5017728p5017728.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
