[OpenSIPS-Users] opensips B2BUA configuration in top hiding
Hi,
I am new to Opensips and also new on SIP as well. I am trying to configure
Opensips as B2BUA top hiding mode.
I used the below configuration for that:
route {
if(is_method(INVITE) !(src_ip == my IP src_port ==5060)) /* skip
Invite messages generated by the B2BUA*/
{
b2b_init_request(top hiding);
exit; # do not forward this request, another one will be generated
};
With such configuration, I am able to establish a call between my client and
server machine.
During the call establishment, Opensips B2BUA translated the Call ID with its
own call ID [B2B.###.##]
However, when I tried to terminate the call, Opensips B2BUA does not translate
the call ID and its causes the error at the server.
Looking into the looks, it seemed to me that the Opensips B2BUA does not go to
search the dialog context while handling BYE message.
The termination works fine for me when I changed the small code in
b2b_prescript_f so that the code go to search the dialog when BYE message
arrives.
if((method_value == METHOD_ACK) ||
(method_value == METHOD_BYE)) //BYE Method added here
{
goto search_dialog;
}
Let me know if I am missing something here?
Regards
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Re: [OpenSIPS-Users] Share database between opensips instances
Hi Aron, For dialog module, you should use different tables for different servers. Otherwise, if an opensips restarts, it will have no idea which dialogs belong to it or to another instance. Simply create different tables in the same DB. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 09.06.2015 17:32, Podrigal, Aron wrote: Nothing specific, its just that I don't want to have another database, so using the same database with the default tables. However if it is not recommended to use the same table, than I'll have to specify a different table. On Jun 9, 2015 9:31 AM, Bogdan-Andrei Iancu bog...@opensips.org mailto:bog...@opensips.org wrote: Hello Aron, What are you trying to achieve by using the same table ? keep in mind the dialog table is more for restart persistence, the primary copy of the dialog data is in memory. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 09.06.2015 03:21, Podrigal, Aron wrote: Can I use the same dialog tables for 2 seperate opensips instances one proxying in front of the other? -- Aron Podrigal - //Be happy :-) ___ 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
Re: [OpenSIPS-Users] Manipulate Opensips generated SIP headers
Hi Ping, You should not manually change the cseq in the messages as you will break the whole dialog (the sequential requests). The 2.1 version does cseq increasing (with dialog consistency) when performing uac_auth(). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 05.06.2015 09:39, Ping Han wrote: Hi Bogdan, Thanks for your reply. I am trying to manipulate the CSeq header. I have attached a packet trace. In the example 10.216.235.38 sends a SIP Invite to 10.216.235.37 via Opensips. Then 10.216.235.37 challenges authentication. In the second SIP Invite that contains the authentication info (frame number 6), I have to increment the CSeq header otherwise 10.216.235.37 does not like it. The new CSeq is now set to 2. If 10.216.235.37 is not able to accept the call, it returns something other than 200 OK (606 in this example). Now it is the problem. Opensips sends back an ACK (frame number 12) with CSeq set to 1 (not 2). 10.216.235.37 is expecting an ACK with a CSeq set to 2. The consequence is that 10.216.235.37 keeps sending 606 until times out. The ACK is internally generated by Opensips and as you said we can not manipulate it. I am wondering if there is any way to solve this problem. PS: 10.216.235.97 is the internal IP of Opensips server and 10.216.235.74 is the external IP. Thanks, Ping On Thu, Jun 4, 2015 at 12:55 AM, Bogdan-Andrei Iancu bog...@opensips.org mailto:bog...@opensips.org wrote: Hi Ping, There is no access to internally generated ACK and CANCEL requests. What are you trying to do ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 03.06.2015 02:53, Ping Han wrote: Hi, With Opensips is there a way to manipulate the SIP headers of a messages that are generated locally by Opensips. See the example below. ACK sip:0370103204@192.168.100.37:5060 http://sip:0370103204@192.168.100.37:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.74:5060;branch=z9hG4bK5094.3ee05403.1 From:sip:0370103...@xxxvoip.com mailto:sip%3a0370103...@xxxvoip.com;tag=1740643510 Call-ID: 641613075@192.168.100.38 mailto:641613075@192.168.100.38 To:sip:0370103204@192.168.100.37:5060 http://sip:0370103204@192.168.100.37:5060;tag=1994410995-1432625097305; CSeq: 1 ACK Max-Forwards: 70 User-Agent: OpenSIPS (2.1.0 (x86_64/linux)) Content-Length: 0 Thanks, Ping Han ___ 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
Re: [OpenSIPS-Users] Failed to replicate Dialog
Hello Kneeoh, Finally managed to replicate these errors on my own setup. In my case, the cause was insufficient shared memory for the _primary_ OpenSIPS instance, which MAY end up with some missing data within the dialog module structures, and unfortunately it gets replicated that way. Recommendation: Please make sure you always have enough shared memory (-m and -M command line parameters, or variables from /etc/default/opensips). For each 1K calls/sec with tm+dialog and 60s duration you need roughly 640MB of shared memory. Regarding pkg memory (-M parameter), just use -M16 and you should be fine. Best regards, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 04.06.2015 22:03, Kneeoh wrote: I just popped up to 1.11.5 and am still getting a stream of dialog replication failure even though the non-active host IS listening on the same socket as the primary host. I'm banging my head on the desk, I can't figure out what this isn't working. Host 2 (passive host) Jun 4 18:34:50 /usr/local/sbin/opensips[27448]: ERROR:dialog:receive_binary_packet: Failed to process a binary packet! Jun 4 18:34:50 /usr/local/sbin/opensips[27445]: ERROR:dialog:dlg_replicated_update: dialog not found, building new Jun 4 18:34:50 /usr/local/sbin/opensips[27445]: ERROR:dialog:dlg_replicated_create: Dialog in DB doesn't match any listening sockets Jun 4 18:34:50 /usr/local/sbin/opensips[27445]: ERROR:dialog:receive_binary_packet: Failed to process a binary packet! Netstat on Host 1 netstat -nlp | grep opensips udp0 0 192.168.30.40:5060 0.0.0.0:* 7304/opensips ---virtual ip udp0 0 192.168.30.39:5060 0.0.0.0:* 7304/opensips ---virtual ip udp0 0 10.1.0.41:5092 0.0.0.0:* 7304/opensips ---binary replication binding (bin_listen) Netstat on Host 2 netstat -nlp | grep opensips udp0 0 192.168.30.40:5060 0.0.0.0:* 27441/opensips ---virtual ip udp0 0 192.168.30.39:5060 0.0.0.0:* 27441/opensips ---virtual ip udp 2176 0 10.1.0.42:5092 0.0.0.0:* 27441/opensips ---binary replication binding (bin_listen) On Thursday, May 7, 2015 1:36 PM, Kneeoh kne...@yahoo.com wrote: Hi Bogdan, Both Opensips hosts are set to use corosync/heartbeat to failover the two IPs in our config. Both hosts are set to non-localbind and opensips is explicitly listening on both of the VIPs. This is why I'm confused. It seems that everything is configured correctly yet I'm getting these errors on the inactive opensips instance. On Thursday, May 7, 2015 1:05 PM, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hi Kneeoh, The dialog replication is done assuming that both opensips servers do share the listening interface (via vrrp, heartbeat, etc). Do you different listening IPs on the 2 opensips instances ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com http://www.opensips-solutions.com/ On 29.04.2015 20:35, Kneeoh wrote: Hello, I've got two VIPs on two instances of opensips and am doing dialog replication. I'm getting a steady stream of failed to replicate dialog errors in my opensips log. 192.168.30.39 192.168.30.40 are the two VIPs. Both have a listen = on both opensips configs. I'm not sure if this line in the log is the problem but it looks like it: DBG:core:bin_pop_str: Popped: '' [0] I'm not sure how the receive IP could be an empty string. debug: DBG:dialog:dlg_replicated_create: Received replicated dialog! DBG:core:bin_pop_str: Popped: 'udp:192.168.30.40:5060' [22] DBG:core:grep_sock_info: checking if host==us: 13==13 [192.168.30.40] == [192.168.30.39] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 DBG:core:grep_sock_info: checking if host==us: 13==13 [192.168.30.40] == [192.168.30.40] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 DBG:core:bin_pop_str: Popped: '' [0] ERROR:dialog:dlg_replicated_create: Dialog in DB doesn't match any listening sockets DBG:dialog:destroy_dlg: destroing dialog 0x7f09ddd9f958 DBG:dialog:destroy_dlg: dlg expired or not in list - dlg 0x7f09ddd9f958 [2225:721583693] with clid 'f4f2446c-6937-1233-f798-0024e869f1eb' and tags 'NULL' 'NULL' ERROR:dialog:receive_binary_packet: Failed to process a binary packet! ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Double via in MI generated NOTIFY
nice trick ))
Well, lookup works with this trick, but t_relay does not. I have:
CRITICAL:tm:w_t_relay: unsupported route type: 32
When I replaced t_relay with forward, I have:
ERROR:core:forward_reply: no 2nd via found in reply
Well, since you can not reproduce it on both OpenSIPS version, I conciser
it as my configuration problem.
I will check again and may be will find something.
Thank you for your help!
On Thu, Jun 4, 2015 at 11:45 AM, Bogdan-Andrei Iancu bog...@opensips.org
wrote:
Trick - put the lookup in a sub-route and call it from local route :)
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
On 04.06.2015 16:20, Stas Kobzar wrote:
Yes, I had the same idea first, but lookup function is not available in
local_route.
Thanks,
On Thu, Jun 4, 2015 at 9:15 AM, Newlin, Ben ben.new...@inin.com wrote:
Can you not perform the lookup in local_route? Then you could send it
where it is supposed to go without the need for the loop.
Ben Newlin
From: Stas Kobzar
Reply-To: OpenSIPS users mailling list
Date: Wednesday, June 3, 2015 at 8:45 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Double via in MI generated NOTIFY
Yes, it resolves to itself. The domain is the one from OpenSIPS
domains table.
And this is actually what I want. I want OpenSIPS to find the contact
from its locations table using function lookup(locations);
And it works, OpenSIPS finds contact IP and sends NOTIFY to the phone.
The only thing is that the packet has 2 Via headers with the same IP and
port of OpenSIPS server.
As I said, it works, but looks weird.
On Wed, Jun 3, 2015 at 7:00 PM, Newlin, Ben ben.new...@inin.com wrote:
It sounds like you may be sending the NOTIFY to yourself when you use
the domain name instead of the IP. Have you verified the address that the
domain resolves to? Is it the same as the OpenSIPS instance?
Ben Newlin
From: Stas Kobzar
Reply-To: OpenSIPS users mailling list
Date: Wednesday, June 3, 2015 at 6:00 PM
To: Bogdan-Andrei Iancu
Cc: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Double via in MI generated NOTIFY
Hi Bogdan,
No, I do not use local_route at all.
My code for these kind of notifies is in the beginning of the initial
part or main route after t_check_trans and looks like this:
t_check_trans();
# RFC3265: NOTIFY can not be outside SIP dialog.
# So we should drop the packet
if(is_method(NOTIFY)) {
# There is an exception: if this is a check-sync packet
# for phone configuration reload request
if($hdr(Event) =~ check-sync){
lookup(locations, m);
xlog(L_INFO, $ci|$rm| Send reboot request notify packet to
destination $ru);
t_relay();
exit();
}
send_reply(481,Dialog does not exists);
exit;
}
I have tried different scenarios:
- Put this code before t_check_trans
- do not use lookup function
- replaced t_relay with forward
Nothing helped.
However, when I run fifo command using IP address in sip URI, like this:
opensipsctl fifo t_uac_dlg NOTIFY sip:7037@10.130.8.225 . . 'From:
sip:7...@voip.etsmtl.ca;tag=8755a8d01aa27e903a6f4ccaf393f04\r\nTo:
sip:7...@voip.etsmtl.ca\r\nEvent: check-sync\r\n'
then, the packet seems to send directly from local_route. Because, in
this case, I do not even see it in the logs.
Thank you!
Stas
On Wed, Jun 3, 2015 at 11:05 AM, Bogdan-Andrei Iancu
bog...@opensips.org wrote:
Hi Stas,
Do you do any local_route stuff ? If yes, do you modify the RURI/DURI
or other parts of the requests?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
On 01.06.2015 17:42, Stas Kobzar wrote:
Hello,
I am sending NOTIFY packet with event check-sync to reload phone
configuration.
I am doing it with opensips FIFO mi.
My command looks like this:
opensipsctl fifo t_uac_dlg NOTIFY sip:7037@10.130.8.225 . . 'From:
sip:7...@voip.etsmtl.ca;tag=8755a8d01aa27e903a6f4ccaf393f04\r\nTo:
sip:7...@voip.etsmtl.ca\r\nEvent: check-sync\r\n'
When I use IP address in RURI (sip:7037@10.130.8.225) it works as
expected.
However, when I use domain name in RURI (like this:
sip:7...@campus.voip.etsmtl.ca) and my command looks like this:
opensipsctl fifo t_uac_dlg NOTIFY sip:7...@campus.voip.etsmtl.ca . .
'From:
sip:7...@voip.etsmtl.ca;tag=8755a8d01aa27e903a6f4ccaf393f04\r\nTo:
sip:7...@voip.etsmtl.ca\r\nEvent: check-sync\r\n'
I have two Via headers in my resulting NOTIFY packet with different
branche tags:
NOTIFY sip:7037@10.130.8.225 SIP/2.0.
Via: SIP/2.0/UDP 10.130.8.20:5060;branch=z9hG4bK0872.598957f2.0.
Via: SIP/2.0/UDP 10.130.8.20:5060;branch=z9hG4bK0872.498957f2.0.
To: sip:7...@campus.voip.etsmtl.ca.
From:
Re: [OpenSIPS-Users] Issues using memcache auth
Hi Tito,
OK, so you have a plain text pwd in DB. You also load it to the script
during DB auth and push it into the cache. What I was asking is to do
some xlog from script to double check that whatever is stored and later
fetched from script is correct - have you checked that ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 03.06.2015 20:01, Tito Cumpen wrote:
Bogdan,
The password is provided in plaintext by the db. The working scenario
looks likes this :
loadmodule auth.so
loadmodule auth_db.so
modparam(auth_db, calculate_ha1, yes)
modparam(auth_db, password_column, password)
#modparam(auth_db, db_url,
modparam(auth_db, db_url,
http://myauthdb;)
modparam(auth_db, load_credentials, )
On Wed, Jun 3, 2015 at 11:59 AM, Bogdan-Andrei Iancu
bog...@opensips.org mailto:bog...@opensips.org wrote:
Tito,
In DB, what do you have - the plain text passwd or the HA1 ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 03.06.2015 18:56, Tito Cumpen wrote:
Bogdan,
The password is hashed into a numeric value it would seem. Though
my http db provides the password in raw unhashed string when
queried for the subscriber password. The debug shows that the md5
hashing is not being matched matching but I am not sure why since
the save function is only called if (!www_authorize(,
subscriber)) is succeeded. Maybe something is being left out?
Thanks,
Tito
On Wed, Jun 3, 2015 at 11:12 AM, Bogdan-Andrei Iancu
bog...@opensips.org mailto:bog...@opensips.org wrote:
Hi Tito,
Have you double checked if the passwd you push to
pv_www_authorize() (from cache) is the correct one ?
Best Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 02.06.2015 01:58, Tito Cumpen wrote:
my db http returns the password in plain string by the way.
On Mon, Jun 1, 2015 at 6:57 PM, Tito Cumpen t...@xsvoce.com
mailto:t...@xsvoce.com wrote:
Hello group,
I am attempting to add memcache auth validation in
opensips 2.1. I was using http db which returns a string
of the user password password. This was working prior to
utilizing pv_www_authorize. I used this document as a
guideline
http://www.opensips.org/Documentation/Tutorials-MemoryCaching
Here is my auth mod param config
loadmodule cachedb_local.so
loadmodule auth.so
loadmodule auth_db.so
modparam(auth,username_spec,$avp(i:54))
modparam(auth,password_spec,$avp(i:55))
modparam(auth,calculate_ha1,1)
modparam(auth_db, calculate_ha1, yes)
modparam(auth_db, password_column, password)
#modparam(auth_db, db_url,
modparam(auth_db, db_url,
http://mysubscriberdatabase.com;)
modparam(auth_db, load_credentials,
$avp(i:55)=password)
if (is_method(REGISTER)) {
# indicate that the client supports DTLS
# so we know when he is called
if (isflagset(SRC_WS))
setbflag(DST_WS);
if ( isflagset(uac_ws) ) {
xlog(setting avp attribute in register for
websocket \n);
$avp(attr)=websocket;
}
if(cache_fetch(local,passwd_$tu,$avp(i:55))) {
xlog($tU 's credentials are stored in local cache using
it for this register request \n);
$avp(i:54) = $tU;
xlog(SCRIPT: stored password is $avp(i:55)\n);
# perform auth from variables
# $avp(i:54) contains the username
# $avp(i:55) contains the password
if (!pv_www_authorize()) {
$var(rc2) = pv_www_authorize();
# $var(rc2) = www_authorize(, subscriber);
xlog(Return code is $var(rc2) \n);
switch ( $var(rc2) ) {
case 1 :
# if ( proto==TCP || 0 ) {
# setflag(TCP_PERSISTENT);
#setflag(6);
# }
if (!save(location,f))
sl_reply_error();
exit;
# success
break;
case -1:
sl_send_reply(404,User not found);
exit;
break;
case -2:
sl_send_reply(403,Forbidden (Bad auth));
exit;
break;
case -3:
www_challenge(, 0);
exit;
#sl_send_reply(403,Forbidden auth ID);
Re: [OpenSIPS-Users] Manipulate Opensips generated SIP headers
Thanks, Bogdan, I have just upgraded Opensips from 1.9 to 2.1 and did not notice the new features related to uac_auth. I was looking a way to dynamically match the authentication credentials and I believe the following new parameters will solve the problem. auth_realm_avp auth_username_avp auth_password_avp Thanks, Ping On Wed, Jun 10, 2015 at 2:01 AM, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hi Ping, You should not manually change the cseq in the messages as you will break the whole dialog (the sequential requests). The 2.1 version does cseq increasing (with dialog consistency) when performing uac_auth(). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developerhttp://www.opensips-solutions.com On 05.06.2015 09:39, Ping Han wrote: Hi Bogdan, Thanks for your reply. I am trying to manipulate the CSeq header. I have attached a packet trace. In the example 10.216.235.38 sends a SIP Invite to 10.216.235.37 via Opensips. Then 10.216.235.37 challenges authentication. In the second SIP Invite that contains the authentication info (frame number 6), I have to increment the CSeq header otherwise 10.216.235.37 does not like it. The new CSeq is now set to 2. If 10.216.235.37 is not able to accept the call, it returns something other than 200 OK (606 in this example). Now it is the problem. Opensips sends back an ACK (frame number 12) with CSeq set to 1 (not 2). 10.216.235.37 is expecting an ACK with a CSeq set to 2. The consequence is that 10.216.235.37 keeps sending 606 until times out. The ACK is internally generated by Opensips and as you said we can not manipulate it. I am wondering if there is any way to solve this problem. PS: 10.216.235.97 is the internal IP of Opensips server and 10.216.235.74 is the external IP. Thanks, Ping On Thu, Jun 4, 2015 at 12:55 AM, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hi Ping, There is no access to internally generated ACK and CANCEL requests. What are you trying to do ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developerhttp://www.opensips-solutions.com On 03.06.2015 02:53, Ping Han wrote: Hi, With Opensips is there a way to manipulate the SIP headers of a messages that are generated locally by Opensips. See the example below. ACK sip:0370103204@192.168.100.37:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.74:5060;branch=z9hG4bK5094.3ee05403.1 From:sip:0370103...@xxxvoip.com;tag=1740643510 Call-ID: 641613075@192.168.100.38 To:sip:0370103204@192.168.100.37:5060;tag=1994410995-1432625097305; CSeq: 1 ACK Max-Forwards: 70 User-Agent: OpenSIPS (2.1.0 (x86_64/linux)) Content-Length: 0 Thanks, Ping Han ___ 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
[OpenSIPS-Users] Handling of re-invite message from SIP callee on Opensips B2BUA top hiding mode
Hi, I would like to handle the re-invite message from a SIP callee when Opensips is configured as B2BUA and in top hiding mode. Scenario: 1. Caller and Callee established call with Opensips B2BUA configured in top hiding mode 2. Callee put the caller on hold which initiates the re-invite message from Callee 3. Caller send 200 OK however Opensips B2BUA does not seem to forward the 200 OK message Is it possible to have such scenario in top hiding mode or specific scenario would be needed. Please suggest. Thanks DISCLAIMER: This message is proprietary to Aricent and is intended solely for the use of the individual to whom it is addressed. It may contain privileged or confidential information and should not be circulated or used for any purpose other than for what it is intended. If you have received this message in error, please notify the originator immediately. If you are not the intended recipient, you are notified that you are strictly prohibited from using, copying, altering, or disclosing the contents of this message. Aricent accepts no responsibility for loss or damage arising from the use of the information transmitted by this email including damage from virus. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Manipulate Opensips generated SIP headers
Hi Bogdan, I have implemented the new features and they work perfectly. Thanks, Chris On Wed, Jun 10, 2015 at 10:13 AM, Ping Han pinghan...@gmail.com wrote: Thanks, Bogdan, I have just upgraded Opensips from 1.9 to 2.1 and did not notice the new features related to uac_auth. I was looking a way to dynamically match the authentication credentials and I believe the following new parameters will solve the problem. auth_realm_avp auth_username_avp auth_password_avp Thanks, Ping On Wed, Jun 10, 2015 at 2:01 AM, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hi Ping, You should not manually change the cseq in the messages as you will break the whole dialog (the sequential requests). The 2.1 version does cseq increasing (with dialog consistency) when performing uac_auth(). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developerhttp://www.opensips-solutions.com On 05.06.2015 09:39, Ping Han wrote: Hi Bogdan, Thanks for your reply. I am trying to manipulate the CSeq header. I have attached a packet trace. In the example 10.216.235.38 sends a SIP Invite to 10.216.235.37 via Opensips. Then 10.216.235.37 challenges authentication. In the second SIP Invite that contains the authentication info (frame number 6), I have to increment the CSeq header otherwise 10.216.235.37 does not like it. The new CSeq is now set to 2. If 10.216.235.37 is not able to accept the call, it returns something other than 200 OK (606 in this example). Now it is the problem. Opensips sends back an ACK (frame number 12) with CSeq set to 1 (not 2). 10.216.235.37 is expecting an ACK with a CSeq set to 2. The consequence is that 10.216.235.37 keeps sending 606 until times out. The ACK is internally generated by Opensips and as you said we can not manipulate it. I am wondering if there is any way to solve this problem. PS: 10.216.235.97 is the internal IP of Opensips server and 10.216.235.74 is the external IP. Thanks, Ping On Thu, Jun 4, 2015 at 12:55 AM, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hi Ping, There is no access to internally generated ACK and CANCEL requests. What are you trying to do ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developerhttp://www.opensips-solutions.com On 03.06.2015 02:53, Ping Han wrote: Hi, With Opensips is there a way to manipulate the SIP headers of a messages that are generated locally by Opensips. See the example below. ACK sip:0370103204@192.168.100.37:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.74:5060;branch=z9hG4bK5094.3ee05403.1 From:sip:0370103...@xxxvoip.com;tag=1740643510 Call-ID: 641613075@192.168.100.38 To:sip:0370103204@192.168.100.37:5060;tag=1994410995-1432625097305; CSeq: 1 ACK Max-Forwards: 70 User-Agent: OpenSIPS (2.1.0 (x86_64/linux)) Content-Length: 0 Thanks, Ping Han ___ 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
[OpenSIPS-Users] redis with shared dialogs question
Dear all, I have a master redis instance and two opensips connected to it with shared dialogs. Everything work as expected. I can see the same number of profiles in both Opensips servers. When I have a Redis failure I see in both Opensips servers that the profiles are showing 0 calls which is not true. Now the question is: 1. Is there any way for the Opensips to remember the profiles(calls) in memory after the redis dies ? 2. Does OpenSips support redis sentinel for HA ? (from older email I see that doesn't support it) 3. Any idea on how to implement best this ? Thanks a lot for taking the time to reply, Alex ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Share database between opensips instances
Hello Aron, What are you trying to achieve by using the same table ? keep in mind the dialog table is more for restart persistence, the primary copy of the dialog data is in memory. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 09.06.2015 03:21, Podrigal, Aron wrote: Can I use the same dialog tables for 2 seperate opensips instances one proxying in front of the other? -- Aron Podrigal - //Be happy :-) ___ 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
