[OpenSIPS-Users] Hangup does not work
Hi, I'm using OpenSIPS 1.11 together with CSipSimple clients on Android. There option ICE but not STUN is enabled. Now I noticed that disconnecting (hang up) after a phone call does not work. Also when pressing the red hang up-button in the App, connection is still alive and both sides can communicate with each other. Then after some time (10..20 seconds?) a timeout error is shown on both sides and connection is closed. Any idea if this could be a problem of OpenSIPS and how to solve it? It doesn't matters if I use TLS or not. Thanks! ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] strange behavior use_next_gw
I see a strange behaviour using the route_to_carrier and use_next_gw function: If I use a sip client whichs adds :5060 to the ruri and I receive a 503 Service unavailable from the first gateway, then the ruri sent to the next gateway (using the use_next_gw() function) is mixed up. The first prefix is still there preceded by the prefix from the next gateway with the original number four digets short. Let me explain: The way it works ok (note, no :5060) INVITE sip:31206655...@mysipserver.org SIP/2.0 - INVITE sip:123431206655443@10.20.30.40 SIP/2.0 - 503 Service unavailable - INVITE sip:876531206655443@20.30.40.50 SIP/2.0 The problem i see (note, :5060) INVITE sip:31206655...@mysipserver.org SIP/2.0 - INVITE sip:123431206655443@10.20.30.40 SIP/2.0 - 503 Service unavailable - INVITE sip:876512343120665@20.30.40.50 SIP/2.0 So I notice the first prefix is still there: 8765 1234 3120665 I can reproduce this by using a 3CX soft client (then it goes wrong). Here below the output of 'opensipsctl dr show' dr gateways ++--+--+---+---++---++---++-+ | id | gwid | type | address | strip | pri_prefix | attrs | probe_mode | state | socket | description | ++--+--+---+---++---++---++-+ | 1 | 1|1 | 10.20.30.40 | 0 | 1234 | NULL | 0 | 0 | NULL | Gateway A | | 2 | 2|2 | 20.30.40.50 | 0 | 8765 | NULL | 0 | 0 | NULL | Gateway B | ++--+--+---+---++---++---++-+ dr groups ++--++-+-+ | id | username | domain | groupid | description | ++--++-+-+ | 1 | .* | .* | 1 | Inbound | | 2 | .* | .* | 2 | Outbound| ++--++-+-+ dr carriers ++---++---+---+---+---+ | id | carrierid | gwlist | flags | state | attrs | description | ++---++---+---+---+---+ | 1 | 1 | 1,2| 0 | 0 | NULL | Carrier A | | 2 | 2 | 2,1| 0 | 0 | NULL | Carrier B | ++---++---+---+---+---+ dr rules ++-++-+--+-++---+-+ | ruleid | groupid | prefix | timerec | priority | routeid | gwlist | attrs | description | ++-++-+--+-++---+-+ | 1 | 1 | 416555 | |0 | NULL| 1 | NULL | Inbound | | 2 | 2 | 416555 | |0 | NULL| 1 | NULL | Outbound| ++-++-+--+-++---+-+ Is this a bug? I can give more info if wanted... -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/strange-behavior-use-next-gw-tp7595700.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] Logwatch-rules for OpenSIPS?
Done - thanks :-) 2015-03-09 10:21 GMT+01:00 Liviu Chircu li...@opensips.org: Hello Karl, We haven't added any logwatch files/scripts yet, but this looks like an excellent suggestion for a Feature Request on GitHub [1] [1] https://github.com/OpenSIPS/opensips/issues?q=is%3Aopen+is%3Aissue+label%3A%22feature+request%22 Best regards, Liviu Chircu OpenSIPS Developerhttp://www.opensips-solutions.com On 09.03.2015 11:10, Karl Karpfen wrote: Hi, I'm using version 1.11 from git - are there some predefined logwatch rules available I can use on Ubuntu 12.04? Thanks! ___ Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ 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] URGENT: ERROR:siptrace:pipport2su: bad protocol udp
Superb, definitely going to give a try, I have a silly question. Can I apply
that patch manually on my beach because if I try new master then fires it
require any change in mysql?
Reason we have couple of box running 2.1.x so want to keep it same. But any way
let me test code first. And get back to you.
--
Sent from my iPhone
On Mar 9, 2015, at 5:03 AM, Răzvan Crainea raz...@opensips.org wrote:
Hi, Satish!
Indeed, there was an issue with the protocol check. I fixed it in the master
branch. Please give it another try and let me know if you still have problems.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 03/08/2015 09:31 PM, Satish Patel wrote:
I got your point, but our plan is to use 2.1.x and we are already using it
since last 6 month without issue.
But it should work in 2.1.x right?
On Sun, Mar 8, 2015 at 3:20 PM, shah...@voip-demos.com wrote:
OpenSIPS v2.x is much different then v1.x. So, if you are not familiar with
it, you should better use 1.x
Thank you.
On 2015-03-08 19:52, Satish Patel wrote:
I tried same configuration on 1.11 version and it works! so look like
something wrong in 2.1.x version please fix that bug as soon as possible
On Sun, Mar 8, 2015 at 2:04 PM, Satish Patel satish@gmail.com wrote:
sorry for push but it wired error!
I have configure siptrace to send packet to Homer but getting following
error in logs
ERROR:siptrace:pipport2su: bad protocol udp
ERROR:siptrace:pipport2su: bad protocol udp
ERROR:siptrace:pipport2su: bad protocol udp
Opensips 2.1.x
SIP Capture agent
loadmodule siptrace.so
modparam(siptrace, duplicate_uri, sip:192.168.1.200:9060)
modparam(siptrace, duplicate_with_hep, 1)
modparam(siptrace, trace_to_database, 0)
modparam(siptrace, trace_flag, 22)
modparam(siptrace, trace_on, 1)
#HEPv2 == timestamp will be included to HEP header
modparam(siptrace, hep_version, 1)
modparam(siptrace, db_url,
mysql://opensips:xx@localhost/opensips)
...
...
route{
setflag(22);
sip_trace();
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Re: [OpenSIPS-Users] URGENT: ERROR:siptrace:pipport2su: bad protocol udp
Sorry It was branch
My iPhone is over smart :(
--
Sent from my iPhone
On Mar 9, 2015, at 9:12 AM, Satish Patel satish@gmail.com wrote:
Superb, definitely going to give a try, I have a silly question. Can I apply
that patch manually on my beach because if I try new master then fires it
require any change in mysql?
Reason we have couple of box running 2.1.x so want to keep it same. But any
way let me test code first. And get back to you.
--
Sent from my iPhone
On Mar 9, 2015, at 5:03 AM, Răzvan Crainea raz...@opensips.org wrote:
Hi, Satish!
Indeed, there was an issue with the protocol check. I fixed it in the master
branch. Please give it another try and let me know if you still have
problems.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 03/08/2015 09:31 PM, Satish Patel wrote:
I got your point, but our plan is to use 2.1.x and we are already using it
since last 6 month without issue.
But it should work in 2.1.x right?
On Sun, Mar 8, 2015 at 3:20 PM, shah...@voip-demos.com wrote:
OpenSIPS v2.x is much different then v1.x. So, if you are not familiar
with it, you should better use 1.x
Thank you.
On 2015-03-08 19:52, Satish Patel wrote:
I tried same configuration on 1.11 version and it works! so look like
something wrong in 2.1.x version please fix that bug as soon as possible
On Sun, Mar 8, 2015 at 2:04 PM, Satish Patel satish@gmail.com
wrote:
sorry for push but it wired error!
I have configure siptrace to send packet to Homer but getting
following error in logs
ERROR:siptrace:pipport2su: bad protocol udp
ERROR:siptrace:pipport2su: bad protocol udp
ERROR:siptrace:pipport2su: bad protocol udp
Opensips 2.1.x
SIP Capture agent
loadmodule siptrace.so
modparam(siptrace, duplicate_uri, sip:192.168.1.200:9060)
modparam(siptrace, duplicate_with_hep, 1)
modparam(siptrace, trace_to_database, 0)
modparam(siptrace, trace_flag, 22)
modparam(siptrace, trace_on, 1)
#HEPv2 == timestamp will be included
to HEP header
modparam(siptrace, hep_version, 1)
modparam(siptrace, db_url,
mysql://opensips:xx@localhost/opensips)
...
...
route{
setflag(22);
sip_trace();
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Re: [OpenSIPS-Users] strange behavior use_next_gw
I tested a quick 'fix' which works:
$ru = sip: + $rU + @ + $rd;
if(route_to_carrier($avp(droute)))
{
--
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[OpenSIPS-Users] execute lb_status remotely to disable a node
Is there a way to execute lb_status remotely to disable a node from load balancer? So I have an inbound Opensips that takes all the calls and load balance to multiple asterisk. And then I have another obound opensips that will terminate call from the asterisk server. The call flow goes: Inbound Opensips -- multiple Asterisk -- outbound Opensips. Inbound opensips send the calls across multiple asterisk and each asterisk send call (1 to 1) to an outbound opensips. When the outbound opensips goes down, inbound opensips will keep on sending call to the asterisk not know it can no longer terminate call (since the outbound opensips is down). Thus I want to be able to have asterisk send the opensipsctl fifo lb_status x,0 to the inbound opensip so that it will not keep sending call to it. Is that possible? Thx! -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/execute-lb-status-remotely-to-disable-a-node-tp7595707.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] URGENT: ERROR:siptrace:pipport2su: bad protocol udp
Thanks Razvan,
It is working great!! you guys are awesome!
On Mon, Mar 9, 2015 at 9:43 AM, Satish Patel satish@gmail.com wrote:
Sorry It was branch
My iPhone is over smart :(
--
Sent from my iPhone
On Mar 9, 2015, at 9:12 AM, Satish Patel satish@gmail.com wrote:
Superb, definitely going to give a try, I have a silly question. Can I
apply that patch manually on my beach because if I try new master then
fires it require any change in mysql?
Reason we have couple of box running 2.1.x so want to keep it same. But
any way let me test code first. And get back to you.
--
Sent from my iPhone
On Mar 9, 2015, at 5:03 AM, Răzvan Crainea raz...@opensips.org wrote:
Hi, Satish!
Indeed, there was an issue with the protocol check. I fixed it in the
master branch. Please give it another try and let me know if you still have
problems.
Best regards,
Răzvan Crainea
OpenSIPS Solutionswww.opensips-solutions.com
On 03/08/2015 09:31 PM, Satish Patel wrote:
I got your point, but our plan is to use 2.1.x and we are already using
it since last 6 month without issue.
But it should work in 2.1.x right?
On Sun, Mar 8, 2015 at 3:20 PM, shah...@voip-demos.com wrote:
OpenSIPS v2.x is much different then v1.x. So, if you are not familiar
with it, you should better use 1.x
Thank you.
On 2015-03-08 19:52, Satish Patel wrote:
I tried same configuration on 1.11 version and it works! so look like
something wrong in 2.1.x version please fix that bug as soon as possible
On Sun, Mar 8, 2015 at 2:04 PM, Satish Patel satish@gmail.com
wrote:
sorry for push but it wired error!
I have configure siptrace to send packet to Homer but getting
following error in logs
ERROR:siptrace:pipport2su: bad protocol udp
ERROR:siptrace:pipport2su: bad protocol udp
ERROR:siptrace:pipport2su: bad protocol udp
Opensips 2.1.x
SIP Capture agent
loadmodule siptrace.so
modparam(siptrace, duplicate_uri, sip:192.168.1.200:9060)
modparam(siptrace, duplicate_with_hep, 1)
modparam(siptrace, trace_to_database, 0)
modparam(siptrace, trace_flag, 22)
modparam(siptrace, trace_on, 1)
#HEPv2 == timestamp will be included to HEP header
modparam(siptrace, hep_version, 1)
modparam(siptrace, db_url, mysql://opensips:xx@localhost
/opensips)
...
...
route{
setflag(22);
sip_trace();
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Re: [OpenSIPS-Users] URGENT: ERROR:siptrace:pipport2su: bad protocol udp
Hey Razvan,
Can i take following patch and directly apply to my existing install branch
instead of downloading new Master?
https://github.com/OpenSIPS/opensips/commit/178b0cc26b05a81947de150fe1c2df36d600ccaa
Currently i am running:
[root@sip ]# opensips -V
version: opensips 2.1.1dev-tls (x86_64/linux)
flags: STATS: On, USE_IPV6, USE_TCP, USE_TLS, DISABLE_NAGLE, USE_MCAST,
SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
git revision: b3beb20
main.c compiled on 11:44:56 Dec 31 2014 with gcc 4.4.7
On Mon, Mar 9, 2015 at 10:39 AM, Satish Patel satish@gmail.com wrote:
Thanks Razvan,
It is working great!! you guys are awesome!
On Mon, Mar 9, 2015 at 9:43 AM, Satish Patel satish@gmail.com wrote:
Sorry It was branch
My iPhone is over smart :(
--
Sent from my iPhone
On Mar 9, 2015, at 9:12 AM, Satish Patel satish@gmail.com wrote:
Superb, definitely going to give a try, I have a silly question. Can I
apply that patch manually on my beach because if I try new master then
fires it require any change in mysql?
Reason we have couple of box running 2.1.x so want to keep it same. But
any way let me test code first. And get back to you.
--
Sent from my iPhone
On Mar 9, 2015, at 5:03 AM, Răzvan Crainea raz...@opensips.org wrote:
Hi, Satish!
Indeed, there was an issue with the protocol check. I fixed it in the
master branch. Please give it another try and let me know if you still have
problems.
Best regards,
Răzvan Crainea
OpenSIPS Solutionswww.opensips-solutions.com
On 03/08/2015 09:31 PM, Satish Patel wrote:
I got your point, but our plan is to use 2.1.x and we are already using
it since last 6 month without issue.
But it should work in 2.1.x right?
On Sun, Mar 8, 2015 at 3:20 PM, shah...@voip-demos.com wrote:
OpenSIPS v2.x is much different then v1.x. So, if you are not familiar
with it, you should better use 1.x
Thank you.
On 2015-03-08 19:52, Satish Patel wrote:
I tried same configuration on 1.11 version and it works! so look like
something wrong in 2.1.x version please fix that bug as soon as possible
On Sun, Mar 8, 2015 at 2:04 PM, Satish Patel satish@gmail.com
wrote:
sorry for push but it wired error!
I have configure siptrace to send packet to Homer but getting
following error in logs
ERROR:siptrace:pipport2su: bad protocol udp
ERROR:siptrace:pipport2su: bad protocol udp
ERROR:siptrace:pipport2su: bad protocol udp
Opensips 2.1.x
SIP Capture agent
loadmodule siptrace.so
modparam(siptrace, duplicate_uri, sip:192.168.1.200:9060)
modparam(siptrace, duplicate_with_hep, 1)
modparam(siptrace, trace_to_database, 0)
modparam(siptrace, trace_flag, 22)
modparam(siptrace, trace_on, 1)
#HEPv2 == timestamp will be included to HEP header
modparam(siptrace, hep_version, 1)
modparam(siptrace, db_url, mysql://opensips:xx@localhost
/opensips)
...
...
route{
setflag(22);
sip_trace();
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Re: [OpenSIPS-Users] execute lb_status remotely to disable a node
I guess further searching I need to install xmlrpc module, but I read on doc that it needs 0.9.10 but I tried to do apt-get install libxmlrpc-c3=0.9.10 it says it can not find that version. If I simply install, it will install version 1.16, will that be ok? -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/execute-lb-status-remotely-to-disable-a-node-tp7595707p7595711.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] Destination IP in the Branch and Reply Routes
Hello Răzvan, Thank you for your response. Testing against the $dd and local ip would work perfectly however, I have attempted to place some test messages `xlog(L_INFO,Destiantion IP1: $dd\n);` in the main,branch,relay and $dd is always null. I am not sure how this is possible? Do I need to add a modparam for the variable to be populated? Terrance. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] URGENT: ERROR:siptrace:pipport2su: bad protocol udp
Thanks, I did exactly whatever you told me, i just patch 5 line in
siptrace.c file in my current branch (2.1.1dev-tls git revision: b3beb20)
Now it is working but still i am getting following error in logs. ( it is
not saying UDP this time)
ERROR:siptrace:pipport2su: bad protocol
ERROR:siptrace:pipport2su: bad protocol
And Interesting thing, I have Homer running on other box it is getting
following error. Look like it is related to above issue. could you please
take a look
ERROR: sipcapture [hep.c:139]: hepv2_received(): ERROR:
sipcapture:hep_msg_received: unknow protocol [1]
ERROR: sipcapture [hep.c:139]: hepv2_received(): ERROR:
sipcapture:hep_msg_received: unknow protocol [1]
On Mon, Mar 9, 2015 at 1:04 PM, Răzvan Crainea raz...@opensips.org wrote:
Hi, Satish!
Not sure what's the order of your messages, but yes, you can apply the
patch directly on your branch, without cloning the entire Master.
Best regards,
Răzvan Crainea
OpenSIPS Solutionswww.opensips-solutions.com
On 03/09/2015 05:43 PM, Satish Patel wrote:
Hey Razvan,
Can i take following patch and directly apply to my existing install
branch instead of downloading new Master?
https://github.com/OpenSIPS/opensips/commit/178b0cc26b05a81947de150fe1c2df36d600ccaa
Currently i am running:
[root@sip ]# opensips -V
version: opensips 2.1.1dev-tls (x86_64/linux)
flags: STATS: On, USE_IPV6, USE_TCP, USE_TLS, DISABLE_NAGLE, USE_MCAST,
SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
git revision: b3beb20
main.c compiled on 11:44:56 Dec 31 2014 with gcc 4.4.7
On Mon, Mar 9, 2015 at 10:39 AM, Satish Patel satish@gmail.com
wrote:
Thanks Razvan,
It is working great!! you guys are awesome!
On Mon, Mar 9, 2015 at 9:43 AM, Satish Patel satish@gmail.com
wrote:
Sorry It was branch
My iPhone is over smart :(
--
Sent from my iPhone
On Mar 9, 2015, at 9:12 AM, Satish Patel satish@gmail.com wrote:
Superb, definitely going to give a try, I have a silly question. Can
I apply that patch manually on my beach because if I try new master then
fires it require any change in mysql?
Reason we have couple of box running 2.1.x so want to keep it same.
But any way let me test code first. And get back to you.
--
Sent from my iPhone
On Mar 9, 2015, at 5:03 AM, Răzvan Crainea raz...@opensips.org wrote:
Hi, Satish!
Indeed, there was an issue with the protocol check. I fixed it in the
master branch. Please give it another try and let me know if you still have
problems.
Best regards,
Răzvan Crainea
OpenSIPS Solutionswww.opensips-solutions.com
On 03/08/2015 09:31 PM, Satish Patel wrote:
I got your point, but our plan is to use 2.1.x and we are already using
it since last 6 month without issue.
But it should work in 2.1.x right?
On Sun, Mar 8, 2015 at 3:20 PM, shah...@voip-demos.com wrote:
OpenSIPS v2.x is much different then v1.x. So, if you are not
familiar with it, you should better use 1.x
Thank you.
On 2015-03-08 19:52, Satish Patel wrote:
I tried same configuration on 1.11 version and it works! so look like
something wrong in 2.1.x version please fix that bug as soon as possible
On Sun, Mar 8, 2015 at 2:04 PM, Satish Patel satish@gmail.com
wrote:
sorry for push but it wired error!
I have configure siptrace to send packet to Homer but getting
following error in logs
ERROR:siptrace:pipport2su: bad protocol udp
ERROR:siptrace:pipport2su: bad protocol udp
ERROR:siptrace:pipport2su: bad protocol udp
Opensips 2.1.x
SIP Capture agent
loadmodule siptrace.so
modparam(siptrace, duplicate_uri, sip:192.168.1.200:9060)
modparam(siptrace, duplicate_with_hep, 1)
modparam(siptrace, trace_to_database, 0)
modparam(siptrace, trace_flag, 22)
modparam(siptrace, trace_on, 1)
#HEPv2 == timestamp will be included to HEP header
modparam(siptrace, hep_version, 1)
modparam(siptrace, db_url, mysql://opensips:xx@localhost
/opensips)
...
...
route{
setflag(22);
sip_trace();
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Re: [OpenSIPS-Users] URGENT: ERROR:siptrace:pipport2su: bad protocol udp
Hi, Satish!
Not sure what's the order of your messages, but yes, you can apply the
patch directly on your branch, without cloning the entire Master.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 03/09/2015 05:43 PM, Satish Patel wrote:
Hey Razvan,
Can i take following patch and directly apply to my existing install
branch instead of downloading new Master?
https://github.com/OpenSIPS/opensips/commit/178b0cc26b05a81947de150fe1c2df36d600ccaa
Currently i am running:
[root@sip ]# opensips -V
version: opensips 2.1.1dev-tls (x86_64/linux)
flags: STATS: On, USE_IPV6, USE_TCP, USE_TLS, DISABLE_NAGLE,
USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC,
FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
git revision: b3beb20
main.c compiled on 11:44:56 Dec 31 2014 with gcc 4.4.7
On Mon, Mar 9, 2015 at 10:39 AM, Satish Patel satish@gmail.com
mailto:satish@gmail.com wrote:
Thanks Razvan,
It is working great!! you guys are awesome!
On Mon, Mar 9, 2015 at 9:43 AM, Satish Patel satish@gmail.com
mailto:satish@gmail.com wrote:
Sorry It was branch
My iPhone is over smart :(
--
Sent from my iPhone
On Mar 9, 2015, at 9:12 AM, Satish Patel satish@gmail.com
mailto:satish@gmail.com wrote:
Superb, definitely going to give a try, I have a silly
question. Can I apply that patch manually on my beach because
if I try new master then fires it require any change in mysql?
Reason we have couple of box running 2.1.x so want to keep it
same. But any way let me test code first. And get back to you.
--
Sent from my iPhone
On Mar 9, 2015, at 5:03 AM, Răzvan Crainea
raz...@opensips.org mailto:raz...@opensips.org wrote:
Hi, Satish!
Indeed, there was an issue with the protocol check. I fixed
it in the master branch. Please give it another try and let
me know if you still have problems.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com http://www.opensips-solutions.com
On 03/08/2015 09:31 PM, Satish Patel wrote:
I got your point, but our plan is to use 2.1.x and we are
already using it since last 6 month without issue.
But it should work in 2.1.x right?
On Sun, Mar 8, 2015 at 3:20 PM, shah...@voip-demos.com
mailto:shah...@voip-demos.com wrote:
OpenSIPS v2.x is much different then v1.x. So, if you
are not familiar with it, you should better use 1.x
Thank you.
On 2015-03-08 19:52, Satish Patel wrote:
I tried same configuration on 1.11 version and it
works! so look like something wrong in 2.1.x version
please fix that bug as soon as possible
On Sun, Mar 8, 2015 at 2:04 PM, Satish Patel
satish@gmail.com mailto:satish@gmail.com
wrote:
sorry for push but it wired error!
I have configure siptrace to send packet to
Homer but getting following error in logs
ERROR:siptrace:pipport2su: bad protocol udp
ERROR:siptrace:pipport2su: bad protocol udp
ERROR:siptrace:pipport2su: bad protocol udp
Opensips 2.1.x
SIP Capture agent
loadmodule siptrace.so
modparam(siptrace, duplicate_uri,
sip:192.168.1.200:9060 http://192.168.1.200:9060)
modparam(siptrace, duplicate_with_hep, 1)
modparam(siptrace, trace_to_database, 0)
modparam(siptrace, trace_flag, 22)
modparam(siptrace, trace_on, 1)
#HEPv2 == timestamp will be included to HEP header
modparam(siptrace, hep_version, 1)
modparam(siptrace, db_url,
mysql://opensips:xx@localhost/opensips)
...
...
route{
setflag(22);
sip_trace();
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[OpenSIPS-Users] opensips 1.9 missing mi_xmlrpc.so file
Is there a place I can download the mi_xmlrpc.so? I looked under my /usr/local/lib64/opensips/modules directory, the file is not there. Thx! -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/opensips-1-9-missing-mi-xmlrpc-so-file-tp7595712.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] calling opensipsctl fifo lb_status 2 1 by xmlrpc
Ok, so I figured that you can use xmlrpc to call fifo function remotely, but the issue is I am no linux expert, so I've been reading things here and there. I guess the thing I need to do is to write a xxx.php file on Asterisk and then use the php -h xxx.php to execute it to send a command to opensips server? If that is the case can someone provide me with a simple php script to just call lbl_status 2 1 with php script? I would greatly appreciate any help I can get. thank you! -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/calling-opensipsctl-fifo-lb-status-2-1-by-xmlrpc-tp7595718.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] Call transfer problem
I have a OpenSIPS server that acts as a load balancing server for a couple of asterisk servers, and lately I'm having a issue with asterisk responding with a 488 Not acceptable here for transfer requests when it comes to port 5060 on the Opensips server, but if I send it to another port it works fine, I realized that when using port 5060 the sdp has port 0 ie (m=audio 0 RTP/AVP 0 8 101.) but when using another port it does have a port number (ie m=audio 16422 RTP/AVP 0 8 101.) Im attaching both traces, I changed ips for security 212.212.212.212. is the opensips server, 212.212.212.213 is the asterisk server, and 91.176.221.245 is the end user. Can anyone please help me make this work on port 5060 too, and can you please explain why it would act differently, the NAT is also handeled differently as can be seen in the VIA header (maybe router is trying to use some ALG?) Here is the trace using port 5060 U 91.176.221.245:49242 - 212.212.212.212:5060 INVITE sip:1917425@212.212.212.213:5060;nat=yes SIP/2.0. Via: SIP/2.0/UDP 91.176.221.245:32781;branch=z9hG4bK-a6635dda. From: EXT 101 sip:windo...@sip.myserver.com;tag=3189bbd5cf53a984o0. To: sip:19174250...@sip.myserver.com;tag=as7f636018. Call-ID: f8a392e9-f1139d80@192.168.0.101. CSeq: 103 INVITE. Max-Forwards: 70. Route: sip:212.212.212.212:5060 ;lr;ftag=3189bbd5cf53a984o0;did=d97.6c4d2422. Proxy-Authorization: Digest username=WindowP1,realm=myserver,nonce=54f468a5000180cd88bd6cead37ab9d1920cc4c54a0ecdd3,uri=sip:1917425@212.212. 212.213:5060,algorithm=MD5,response=0a5a118e2b318621f32b5d60b600229c. Contact: EXT 101 sip:WindowP1@91.176.221.245:32781. Expires: 30. User-Agent: Cisco/SPA525G2-7.5.6. Content-Length: 226. Content-Type: application/sdp. . v=0. o=- 33523926 33523927 IN IP4 192.168.0.101. s=-. c=IN IP4 0.0.0.0. t=0 0. m=audio 0 RTP/AVP 0 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendonly. U 212.212.212.212:5060 - 91.176.221.245:49242 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP 91.176.221.245:32781 ;received=91.176.221.245;rport=49242;branch=z9hG4bK-a6635dda. From: EXT 101 sip:windo...@sip.myserver.com;tag=3189bbd5cf53a984o0. To: sip:19174250...@sip.myserver.com;tag=as7f636018. Call-ID: f8a392e9-f1139d80@192.168.0.101. CSeq: 103 INVITE. Server: OpenSIPS (1.7.2-notls (x86_64/linux)). Content-Length: 0. . U 212.212.212.212:5060 - 212.212.212.213:5060 INVITE sip:1917425@212.212.212.213:5060 SIP/2.0. Record-Route: sip:212.212.212.212;lr;ftag=3189bbd5cf53a984o0. Via: SIP/2.0/UDP 212.212.212.212;branch=z9hG4bKe66e.0702ec92.0. Via: SIP/2.0/UDP 91.176.221.245:32781 ;rport=49242;received=91.176.221.245;branch=z9hG4bK-a6635dda. From: EXT 101 sip:windo...@sip.myserver.com;tag=3189bbd5cf53a984o0. To: sip:19174250...@sip.myserver.com;tag=as7f636018. Call-ID: f8a392e9-f1139d80@192.168.0.101. CSeq: 103 INVITE. Max-Forwards: 69. Proxy-Authorization: Digest username=WindowP1,realm=myserver,nonce=54f468a5000180cd88bd6cead37ab9d1920cc4c54a0ecdd3,uri=sip:1917425@212.212. 212.213:5060,algorithm=MD5,response=0a5a118e2b318621f32b5d60b600229c. Contact: EXT 101 sip:WindowP1@91.176.221.245:49242;nat=yes. Expires: 30. User-Agent: Cisco/SPA525G2-7.5.6. Content-Length: 226. Content-Type: application/sdp. . v=0. o=- 33523926 33523927 IN IP4 192.168.0.101. s=-. c=IN IP4 0.0.0.0. t=0 0. m=audio 0 RTP/AVP 0 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendonly. U 212.212.212.213:5060 - 212.212.212.212:5060 SIP/2.0 488 Not acceptable here. Via: SIP/2.0/UDP 212.212.212.212;branch=z9hG4bKe66e.0702ec92.0;received=212.212.212.212;rport=5060. Via: SIP/2.0/UDP 91.176.221.245:32781 ;rport=49242;received=91.176.221.245;branch=z9hG4bK-a6635dda. From: EXT 101 sip:windo...@sip.myserver.com;tag=3189bbd5cf53a984o0. To: sip:19174250...@sip.myserver.com;tag=as7f636018. Call-ID: f8a392e9-f1139d80@192.168.0.101. CSeq: 103 INVITE. Server: SIP Server 9.21/CS. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. X-Asterisk-HangupCause: Normal Clearing. X-Asterisk-HangupCauseCode: 16. Content-Length: 0. . here is the trace using port 5744 U 91.176.221.245:63719 - 212.212.212.212:5744 INVITE sip:1917425@212.212.212.213:5060;nat=yes SIP/2.0. Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-6f0161f0. From: EXT 101 sip:windo...@sip.myserver.com;tag=f8c2b14a55549ac2o0. To: sip:19174250...@sip.myserver.com;tag=as1fcb82f4. Call-ID: 5832b9ea-a4a32aa2@192.168.0.101. CSeq: 103 INVITE. Max-Forwards: 70. Route: sip:212.212.212.212:5744 ;lr;ftag=f8c2b14a55549ac2o0;did=9ae.775cc5c5. Proxy-Authorization: Digest username=WindowP1,realm=myserver,nonce=54f46d7a081272baa39f9e34a0ee81f471dec7b034ba,uri= sip:1917425@212.212.212.213:5060 ,algorithm=MD5,response=4096f485853bf6a209424c36dac2342b. Contact: EXT 101 sip:WindowP1@192.168.0.101:5060. Expires: 30. User-Agent:
Re: [OpenSIPS-Users] TLS handling
Vlad, community, still no idea about that?
I really need your advice with that.
06.03.2015, 14:14, Чалков Артём achal...@ya.ru:
Yes, i sure. I have only 1 branch because i have only one instance to receive
sent MESSAGE and i dont add branches manually. Actually, that is all MESSAGE
routing:
...
if (is_method(MESSAGE)) route(MESSAGE);
...
route[MESSAGE] {
...
if (!lookup(location)) {
...
} else {
t_on_reply(ON_REPLY);
if ($si != 127.0.0.1) t_on_failure(ON_FAIL_MESSAGE);
t_relay(0x01);
$var(retcode) = $retcode;
xlog(L_INFO, [!!!MESSAGE_DEBUG!!!] t_relay returns
$var(retcode) LOGHDR);
ifdef(`LOGS', `xlog(L_INFO, [MESSAGE] Request is leaving
server LOGHDR);')
exit;
}
}
I have sent to your e-mail (vladp...@opensips.org) debug=4 log from moment,
when i killed softphone (destination of MESSAGE) without sip unregistering,
but with removing tcp/tls connection.
06.03.2015, 12:43, Vlad Paiu vladp...@opensips.org:
Hello,
OpenSIPS complains that there is an error when connecting via TCP to
that endpoint.
Now, are you sure you do not have multple branches when relaying that
SIP MESSAGE,out of which only one fails ? In t_relay(), if you have
multiple branches and at least one succceeds, you will get a 1 return code.
Please post the complete debug=4 logs for the processing. In the
previous email, you've truncated the logs to the moment OpenSIPS gets
blocked in trying to connect to the endpoint - what happens afterwards (
after connet timeout ) would also be helpfull.
Best Regards,
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com
On 06.03.2015 11:06, Чалков Артём wrote:
do anyone have any idea about how to handle that?
05.03.2015, 16:22, Чалков Артём achal...@ya.ru:
debug=4
Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]:
DBG:core:tcp_read_req: We're releasing the connection in state 3
Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]:
DBG:core:io_watch_del: io_watch_del op on index -1 36 (0x77dee0, 36, -1,
0x10,0x1) fd_no=2 called
Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]:
DBG:core:release_tcpconn: releasing con 0x7f2be91663a8, state 0, fd=36,
id=1
Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]:
DBG:core:release_tcpconn: extra_data (nil)
Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_msg:
SIP Request:
Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_msg:
method: MESSAGE
Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19028]:
DBG:core:handle_tcp_child: reader response= 7f2be91663a8, 0 from 0
Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_msg:
uri: sip:achalkov1@x.x.x.x:3631;transport=TCP
Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19028]:
DBG:core:io_watch_add: io_watch_add op on 52 (0x77dd80, 52, 2,
0x7f2be91663a8,1), fd_no=38
Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_msg:
version: SIP/2.0
Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]:
DBG:core:parse_headers: flags=2
Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]:
DBG:core:parse_via_param: found param type 232, branch =
z9hG4bK-d8754z-668ef50b1a4c0a31-1---d8754z-; state=6
Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]:
DBG:core:parse_via_param: found param type 235, rport = n/a; state=17
Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_via:
end of header reached, state=5
Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]:
DBG:core:parse_headers: via found, flags=2
Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]:
DBG:core:parse_headers: this is the first via
Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:receive_msg:
After parse_msg...
Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:receive_msg:
preparing to run routing scripts...
Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]:
DBG:core:parse_headers: flags=800
Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_to:
end of header reached, state=10
Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_to:
display={}, ruri={sip:achalkov1@x.x.x.x:3631;transport=TCP}
Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]:
DBG:core:get_hdr_field: To [51];
uri=[sip:achalkov1@x.x.x.x:3631;transport=TCP]
Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]:
DBG:core:get_hdr_field: to body
[sip:achalkov1@x.x.x.x:3631;transport=TCP#015#012]
Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]:
DBG:core:get_hdr_field: cseq CSeq: 3 MESSAGE
Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]:
DBG:maxfwd:is_maxfwd_present: value = 70
Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]:
DBG:core:parse_headers: flags=
Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]:
DBG:core:get_hdr_field:
Re: [OpenSIPS-Users] Safe Place/Method to Fix Contact
Hi, Terrance!
Can you make a trace and check whether the Contact header is correct in
the 200OK? Also, make sure that all the URI parameters are preserved
over this change.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 03/07/2015 08:32 AM, Terrance Devor wrote:
Hello Everyone,
Our environment is in an EC2 instance, and would like to fix the
Contact when
signalling internally (ie, private IP), and externally (ie, public
IP). I have added
the following code:
In branch:
if(!isflagset(5)) {
remove_hf(Contact:);
append_hf(Contact: sip:$fU@private ip\r\n);
}
In on reply:
if(!isflagset(5)) {
remove_hf(Contact:);
append_hf(Contact: sip:$tU@public ip:5060\r\n);
}
Code works fine and changes the contact as needed however, OpenSIPS
stops 200OK on OK.
We have the following error message:
Failed to match the sequential request to a known dialog
Attempt to route with preloaded Route's [sip:5147398047@unerline
carrier ip/sip:5193403221@public ip/sip:public
ip;lr;did=2ad.791c28e4/74f3fb2e-3f34-1233-8f94-000c29d9b8bf]
Your help is greatly appreciated,
Terrance.
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Re: [OpenSIPS-Users] dispatcher wired behavior
Hi, Satish!
If you want to stop OpenSIPS from failing over to next FS2, you should
not call t_relay() in failure_route for the 404 status.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 03/08/2015 03:57 PM, Satish Patel wrote:
I have two Freeswitch in dispatcher, everything works great but i have
notice in sip trace if FS1 receive 404 SIP code then it sending it to
next FS2, i think it should stop there instead of forwarding next FS2
Following is my config
Dispatcher
loadmodule dispatcher.so
modparam(dispatcher, dst_avp, $avp(271))
modparam(dispatcher, grp_avp, $avp(272))
modparam(dispatcher, cnt_avp, $avp(273))
modparam(dispatcher, ds_ping_interval, 5)
modparam(dispatcher, ds_probing_threshhold, 5)
modparam(dispatcher, ds_probing_mode, 0)
modparam(dispatcher, options_reply_codes, 501, 403, 200)
modparam(dispatcher, db_url,
mysql://opensips:@localhost/opensips)
...
...
route[to_dispatcher] {
# Dispatch to FS
if ( !ds_select_dst(1, 4, FM10)) {
send_reply(500,Unable to dispatch call to Freeswitch);
exit;
} else {
xlog(L_WARN, dispatcher: Attempting to dispatch call to
$du\n);
}
t_on_failure(dispatcher_rollover);
t_relay();
}
failure_route[dispatcher_rollover] {
if (t_was_cancelled()) {
exit;
}
if (t_check_status(408) t_local_replied(all)) {
xlog(L_NOTICE, dispatcher: connection timeout: $rd\n);
ds_mark_dst(p);
}
if(!ds_next_dst()) {
xlog(L_ERR, dispatcher: No more dispatcher in route
set\n);
t_reply(500, Temporary failure);
exit;
}
xlog(L_INFO, dispatcher: attempting relay to new
dispatcher: $du\n);
t_on_failure(dispatcher_rollover);
t_relay();
}
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Re: [OpenSIPS-Users] Destination IP in the Branch and Reply Routes
Hi, Terrance! No, OpenSIPS does not provide such function or variable afaik. However, probably you need to test against a limited range of private addresses, so you can simply check a subclass, something like $dd =~ 192\.168\. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 03/08/2015 03:25 AM, Terrance Devor wrote: Hello Everyone, Our opensips is in a nat'ed environment , and we need a way to test if the destination ip (ie, where the next message is going to be sent) is public or private. Is there a variable, or even better a function that can test for this. Kind Regards, Terrance ___ 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] URGENT: ERROR:siptrace:pipport2su: bad protocol udp
Hi, Satish!
Indeed, there was an issue with the protocol check. I fixed it in the
master branch. Please give it another try and let me know if you still
have problems.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 03/08/2015 09:31 PM, Satish Patel wrote:
I got your point, but our plan is to use 2.1.x and we are already
using it since last 6 month without issue.
But it should work in 2.1.x right?
On Sun, Mar 8, 2015 at 3:20 PM, shah...@voip-demos.com
mailto:shah...@voip-demos.com wrote:
OpenSIPS v2.x is much different then v1.x. So, if you are not
familiar with it, you should better use 1.x
Thank you.
On 2015-03-08 19:52, Satish Patel wrote:
I tried same configuration on 1.11 version and it works! so look
like something wrong in 2.1.x version please fix that bug as soon
as possible
On Sun, Mar 8, 2015 at 2:04 PM, Satish Patel
satish@gmail.com mailto:satish@gmail.com wrote:
sorry for push but it wired error!
I have configure siptrace to send packet to Homer but
getting following error in logs
ERROR:siptrace:pipport2su: bad protocol udp
ERROR:siptrace:pipport2su: bad protocol udp
ERROR:siptrace:pipport2su: bad protocol udp
Opensips 2.1.x
SIP Capture agent
loadmodule siptrace.so
modparam(siptrace, duplicate_uri, sip:192.168.1.200:9060
http://192.168.1.200:9060)
modparam(siptrace, duplicate_with_hep, 1)
modparam(siptrace, trace_to_database, 0)
modparam(siptrace, trace_flag, 22)
modparam(siptrace, trace_on, 1)
#HEPv2 == timestamp will be included to HEP header
modparam(siptrace, hep_version, 1)
modparam(siptrace, db_url,
mysql://opensips:xx@localhost/opensips)
...
...
route{
setflag(22);
sip_trace();
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[OpenSIPS-Users] Logwatch-rules for OpenSIPS?
Hi, I'm using version 1.11 from git - are there some predefined logwatch rules available I can use on Ubuntu 12.04? Thanks! ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Logwatch-rules for OpenSIPS?
Hello Karl, We haven't added any logwatch files/scripts yet, but this looks like an excellent suggestion for a Feature Request on GitHub [1] [1] https://github.com/OpenSIPS/opensips/issues?q=is%3Aopen+is%3Aissue+label%3A%22feature+request%22 Best regards, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 09.03.2015 11:10, Karl Karpfen wrote: Hi, I'm using version 1.11 from git - are there some predefined logwatch rules available I can use on Ubuntu 12.04? Thanks! ___ 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] incoming DID will not pass to asterisk
Hi,
I need some help please.
I’m trying to pass all sip packets to asterisk via dispatcher module.
The problem I’m having if UA signed in the incoming DID will not pass to
asterisk. The error message was “SIP/2.0 401 Unauthorized”
How can I solve this problem?
preshiead
Sathees
This is my scripts.
route{
if ( !mf_process_maxfwd_header(10) )
{
sl_send_reply(483,To Many Hops);
drop();
}
ds_select_dst(1, 0);
forward();
#t_relay();
}
--
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