Re: [OpenSIPS-Users] SRV failover results in double call
Brent Thomson wrote: Stanisław Pitucha wrote: 2009/8/24 Brent Thomson bthom...@getjive.com: Bogdan-Andrei Iancu wrote: Hi Brent, This problem was reported last week by another person and fixed on SVN (including in 1.6 branch). What you have to do is to upgrade from SVN and hopefully the problem will be solved. Cool. Thanks. Or if you're ok with applying custom patches, just pull the trunk change - rev 6007. It applies to opensips 1.5.2 just fine. Thanks for the tip. Patching the release version is definitely preferred. I ran (all on one line): svn diff -r 6006:6007 https://opensips.svn.sourceforge.net/svnroot/opensips/trunk opensips.diff and got about 8 lines of changes in modules/tm/tm_reply.c. Does this seem about right? Yes, that is correct. Regards, Bogdan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Dialog information tracing in opensips Issue
Post the SIP capture of the call you are testing with. Use ngrep -d any . port 5060 to get the capture - this will solve the mystery. Regards, Bogdan urmi lakkad wrote: Hello, Can u please suggest me some solution of my problem of DIALLOG module ? Thank you for your attention. -Urmi 2009/8/21 urmi lakkad urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com Hello Brett, Thank you very much for quick response. My calls are working fine. I have checked through SIPp and also with Grandstream Phones. The call is working fine with out failure. At the time of call, I have started the wireshark to capture the packets, but there also I m not getting any negative reply like 400 or 300. See, my call is working fine, call dialog created successfully, but after that it destroyed, again new dialog is created n that too destroy. For a single call it creates 2 dialogs. But that dialog entry is not going to DB. Please suggest me the right thing to do. Thanks a lot for your attention. -Urmi On Thu, Aug 20, 2009 at 7:38 PM, Brett Nemeroff br...@nemeroff.com mailto:br...@nemeroff.com wrote: Urmi, You log shows the call having failed. I'm not sure why you think it runs for the proper duration. But as far as OpenSIPs is concerned, the call failed. It's likely a problem in your sipp scenario. It's very possible that sipp thinks the call is up, but the proxy does not. In any case, OpenSIPs is behaving as expected, the call fails, the dialog is destroyed. Aug 19 17:46:27 [6060] DBG:dialog:next_state_dlg: dialog 0x2d55af90 changed from state 1 to state 5, due event 1 Aug 19 17:46:27 [6060] DBG:dialog:dlg_onreply: dialog 0x2d55af90 failed (negative reply) BTW, a negative reply is =400 (or may also include = 300, can't remember). Check your traces, see where that comes from. -Brett On Thu, Aug 20, 2009 at 9:02 AM, urmi lakkad urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com wrote: Hello Stanisław Pitucha, Thank you for support. No, My call is established perfectly and is running for the specified duration without fail. I m firing the call using SIPp. Also, the dialog state gives me 1. Thanks for ur attention. -Urmi 2009/8/20 Stanisław Pitucha virap...@gmail.com mailto:virap...@gmail.com 2009/8/20 urmi lakkad urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com: Am I doing right or not ? If not, please tell me the correct way. One more thing, Is my configuration is correct or not ?? It looks like your call doesn't even get accepted: Aug 19 17:46:27 [6060] DBG:dialog:next_state_dlg: dialog 0x2d55af90 changed from state 1 to state 5, due event 1 Aug 19 17:46:27 [6060] DBG:dialog:dlg_onreply: dialog 0x2d55af90 failed (negative reply) Maybe you require authentication, or something else? Just take care of the call not failing first. So far it's rejected before an OK answer (state 1 is after sending an INVITE, state 5 is deleted - more or less). Capture the traffic and see what's going on. -- Kind regards, Stanisław Pitucha, Gradwell Voip Engineer T: 01225 800 831 | F: 01225 800 801 | E: s...@gradwell.net mailto:s...@gradwell.net | www.gradwell.com http://www.gradwell.com Gradwell – Internet for Business People Phone Services | Business Broadband | Email Website Hosting Can switching to VoIP today put some change in your pocket? Registered Address: 26 Cheltenham Street, Bath, BA2 3EX, UK. Company Number: 3673235 ___ 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 mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org
Re: [OpenSIPS-Users] Dialog information tracing in opensips Issue
Hello Bogdan, Thank you for ur response. Here with this mail I have *attached my SIP call capture* using ngrep. So, please find the attachment. and do needful. -Thanks Urmi On Tue, Aug 25, 2009 at 12:41 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Post the SIP capture of the call you are testing with. Use ngrep -d any . port 5060 to get the capture - this will solve the mystery. Regards, Bogdan urmi lakkad wrote: Hello, Can u please suggest me some solution of my problem of DIALLOG module ? Thank you for your attention. -Urmi 2009/8/21 urmi lakkad urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com Hello Brett, Thank you very much for quick response. My calls are working fine. I have checked through SIPp and also with Grandstream Phones. The call is working fine with out failure. At the time of call, I have started the wireshark to capture the packets, but there also I m not getting any negative reply like 400 or 300. See, my call is working fine, call dialog created successfully, but after that it destroyed, again new dialog is created n that too destroy. For a single call it creates 2 dialogs. But that dialog entry is not going to DB. Please suggest me the right thing to do. Thanks a lot for your attention. -Urmi On Thu, Aug 20, 2009 at 7:38 PM, Brett Nemeroff br...@nemeroff.com mailto:br...@nemeroff.com wrote: Urmi, You log shows the call having failed. I'm not sure why you think it runs for the proper duration. But as far as OpenSIPs is concerned, the call failed. It's likely a problem in your sipp scenario. It's very possible that sipp thinks the call is up, but the proxy does not. In any case, OpenSIPs is behaving as expected, the call fails, the dialog is destroyed. Aug 19 17:46:27 [6060] DBG:dialog:next_state_dlg: dialog 0x2d55af90 changed from state 1 to state 5, due event 1 Aug 19 17:46:27 [6060] DBG:dialog:dlg_onreply: dialog 0x2d55af90 failed (negative reply) BTW, a negative reply is =400 (or may also include = 300, can't remember). Check your traces, see where that comes from. -Brett On Thu, Aug 20, 2009 at 9:02 AM, urmi lakkad urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com wrote: Hello Stanisław Pitucha, Thank you for support. No, My call is established perfectly and is running for the specified duration without fail. I m firing the call using SIPp. Also, the dialog state gives me 1. Thanks for ur attention. -Urmi 2009/8/20 Stanisław Pitucha virap...@gmail.com mailto:virap...@gmail.com 2009/8/20 urmi lakkad urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com: Am I doing right or not ? If not, please tell me the correct way. One more thing, Is my configuration is correct or not ?? It looks like your call doesn't even get accepted: Aug 19 17:46:27 [6060] DBG:dialog:next_state_dlg: dialog 0x2d55af90 changed from state 1 to state 5, due event 1 Aug 19 17:46:27 [6060] DBG:dialog:dlg_onreply: dialog 0x2d55af90 failed (negative reply) Maybe you require authentication, or something else? Just take care of the call not failing first. So far it's rejected before an OK answer (state 1 is after sending an INVITE, state 5 is deleted - more or less). Capture the traffic and see what's going on. -- Kind regards, Stanisław Pitucha, Gradwell Voip Engineer T: 01225 800 831 | F: 01225 800 801 | E: s...@gradwell.net mailto:s...@gradwell.net | www.gradwell.com http://www.gradwell.com Gradwell – Internet for Business People Phone Services | Business Broadband | Email Website Hosting Can switching to VoIP today put some change in your pocket? Registered Address: 26 Cheltenham Street, Bath, BA2 3EX, UK. Company Number: 3673235 ___ Users mailing list Users@lists.opensips.org mailto: Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Dialog information tracing in opensips Issue
Looks good - can you post the opensips logs (in debug=6) for this single call ? just to verify. Regards, Bogdan urmi lakkad wrote: Hello Bogdan, Thank you for ur response. Here with this mail I have _attached my SIP call capture_ using ngrep. So, please find the attachment. and do needful. -Thanks Urmi On Tue, Aug 25, 2009 at 12:41 PM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Post the SIP capture of the call you are testing with. Use ngrep -d any . port 5060 to get the capture - this will solve the mystery. Regards, Bogdan urmi lakkad wrote: Hello, Can u please suggest me some solution of my problem of DIALLOG module ? Thank you for your attention. -Urmi 2009/8/21 urmi lakkad urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com Hello Brett, Thank you very much for quick response. My calls are working fine. I have checked through SIPp and also with Grandstream Phones. The call is working fine with out failure. At the time of call, I have started the wireshark to capture the packets, but there also I m not getting any negative reply like 400 or 300. See, my call is working fine, call dialog created successfully, but after that it destroyed, again new dialog is created n that too destroy. For a single call it creates 2 dialogs. But that dialog entry is not going to DB. Please suggest me the right thing to do. Thanks a lot for your attention. -Urmi On Thu, Aug 20, 2009 at 7:38 PM, Brett Nemeroff br...@nemeroff.com mailto:br...@nemeroff.com mailto:br...@nemeroff.com mailto:br...@nemeroff.com wrote: Urmi, You log shows the call having failed. I'm not sure why you think it runs for the proper duration. But as far as OpenSIPs is concerned, the call failed. It's likely a problem in your sipp scenario. It's very possible that sipp thinks the call is up, but the proxy does not. In any case, OpenSIPs is behaving as expected, the call fails, the dialog is destroyed. Aug 19 17:46:27 [6060] DBG:dialog:next_state_dlg: dialog 0x2d55af90 changed from state 1 to state 5, due event 1 Aug 19 17:46:27 [6060] DBG:dialog:dlg_onreply: dialog 0x2d55af90 failed (negative reply) BTW, a negative reply is =400 (or may also include = 300, can't remember). Check your traces, see where that comes from. -Brett On Thu, Aug 20, 2009 at 9:02 AM, urmi lakkad urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com wrote: Hello Stanisław Pitucha, Thank you for support. No, My call is established perfectly and is running for the specified duration without fail. I m firing the call using SIPp. Also, the dialog state gives me 1. Thanks for ur attention. -Urmi 2009/8/20 Stanisław Pitucha virap...@gmail.com mailto:virap...@gmail.com mailto:virap...@gmail.com mailto:virap...@gmail.com 2009/8/20 urmi lakkad urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com: Am I doing right or not ? If not, please tell me the correct way. One more thing, Is my configuration is correct or not ?? It looks like your call doesn't even get accepted: Aug 19 17:46:27 [6060] DBG:dialog:next_state_dlg: dialog 0x2d55af90 changed from state 1 to state 5, due event 1 Aug 19 17:46:27 [6060] DBG:dialog:dlg_onreply: dialog 0x2d55af90 failed (negative reply) Maybe you require authentication, or something else? Just take care of the call not failing first. So far it's rejected before an OK answer (state 1 is after sending an INVITE, state 5 is deleted - more or less). Capture the traffic and see what's going on. -- Kind regards, Stanisław Pitucha, Gradwell Voip Engineer T: 01225 800 831 | F: 01225 800 801 | E: s...@gradwell.net mailto:s...@gradwell.net mailto:s...@gradwell.net mailto:s...@gradwell.net | www.gradwell.com http://www.gradwell.com http://www.gradwell.com Gradwell – Internet for Business People Phone Services | Business Broadband | Email Website Hosting Can switching to VoIP today put some change in your pocket? Registered Address: 26 Cheltenham Street, Bath, BA2 3EX, UK. Company Number: 3673235 ___ Users mailing list
[OpenSIPS-Users] Dialog information tracing in opensips Issue
Hello Bogdan, Thank you very much for your quick response. Here I have attached OpenSIPs call log. -Urmi On Tue, Aug 25, 2009 at 3:54 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Looks good - can you post the opensips logs (in debug=6) for this single call ? just to verify. Regards, Bogdan urmi lakkad wrote: Hello Bogdan, Thank you for ur response. Here with this mail I have _attached my SIP call capture_ using ngrep. So, please find the attachment. and do needful. -Thanks Urmi On Tue, Aug 25, 2009 at 12:41 PM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Post the SIP capture of the call you are testing with. Use ngrep -d any . port 5060 to get the capture - this will solve the mystery. Regards, Bogdan urmi lakkad wrote: Hello, Can u please suggest me some solution of my problem of DIALLOG module ? Thank you for your attention. -Urmi 2009/8/21 urmi lakkad urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com Hello Brett, Thank you very much for quick response. My calls are working fine. I have checked through SIPp and also with Grandstream Phones. The call is working fine with out failure. At the time of call, I have started the wireshark to capture the packets, but there also I m not getting any negative reply like 400 or 300. See, my call is working fine, call dialog created successfully, but after that it destroyed, again new dialog is created n that too destroy. For a single call it creates 2 dialogs. But that dialog entry is not going to DB. Please suggest me the right thing to do. Thanks a lot for your attention. -Urmi On Thu, Aug 20, 2009 at 7:38 PM, Brett Nemeroff br...@nemeroff.com mailto:br...@nemeroff.com mailto:br...@nemeroff.com mailto:br...@nemeroff.com wrote: Urmi, You log shows the call having failed. I'm not sure why you think it runs for the proper duration. But as far as OpenSIPs is concerned, the call failed. It's likely a problem in your sipp scenario. It's very possible that sipp thinks the call is up, but the proxy does not. In any case, OpenSIPs is behaving as expected, the call fails, the dialog is destroyed. Aug 19 17:46:27 [6060] DBG:dialog:next_state_dlg: dialog 0x2d55af90 changed from state 1 to state 5, due event 1 Aug 19 17:46:27 [6060] DBG:dialog:dlg_onreply: dialog 0x2d55af90 failed (negative reply) BTW, a negative reply is =400 (or may also include = 300, can't remember). Check your traces, see where that comes from. -Brett On Thu, Aug 20, 2009 at 9:02 AM, urmi lakkad urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com wrote: Hello Stanisław Pitucha, Thank you for support. No, My call is established perfectly and is running for the specified duration without fail. I m firing the call using SIPp. Also, the dialog state gives me 1. Thanks for ur attention. -Urmi 2009/8/20 Stanisław Pitucha virap...@gmail.com mailto:virap...@gmail.com mailto:virap...@gmail.com mailto:virap...@gmail.com 2009/8/20 urmi lakkad urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com: Am I doing right or not ? If not, please tell me the correct way. One more thing, Is my configuration is correct or not ?? It looks like your call doesn't even get accepted: Aug 19 17:46:27 [6060] DBG:dialog:next_state_dlg: dialog 0x2d55af90 changed from state 1 to state 5, due event 1 Aug 19 17:46:27 [6060] DBG:dialog:dlg_onreply: dialog 0x2d55af90 failed (negative reply) Maybe you require authentication, or something else? Just take care of the call not failing first. So far it's rejected before an OK answer (state 1 is after sending an INVITE, state 5 is deleted - more or less). Capture the traffic and see what's going on. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Dialog information tracing in opensips Issue
Hello Bogdan, Thank you very much. Here I have attached my OpenSIPs Log. so please find the attachment. -Urmi Looks good - can you post the opensips logs (in debug=6) for this single call ? just to verify. Regards, Bogdan urmi lakkad wrote: Hello Bogdan, Thank you for ur response. Here with this mail I have _attached my SIP call capture_ using ngrep. So, please find the attachment. and do needful. -Thanks Urmi On Tue, Aug 25, 2009 at 12:41 PM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Post the SIP capture of the call you are testing with. Use ngrep -d any . port 5060 to get the capture - this will solve the mystery. Regards, Bogdan urmi lakkad wrote: Hello, Can u please suggest me some solution of my problem of DIALLOG module ? Thank you for your attention. -Urmi 2009/8/21 urmi lakkad urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com Hello Brett, Thank you very much for quick response. My calls are working fine. I have checked through SIPp and also with Grandstream Phones. The call is working fine with out failure. At the time of call, I have started the wireshark to capture the packets, but there also I m not getting any negative reply like 400 or 300. See, my call is working fine, call dialog created successfully, but after that it destroyed, again new dialog is created n that too destroy. For a single call it creates 2 dialogs. But that dialog entry is not going to DB. Please suggest me the right thing to do. Thanks a lot for your attention. -Urmi On Thu, Aug 20, 2009 at 7:38 PM, Brett Nemeroff br...@nemeroff.com mailto:br...@nemeroff.com mailto:br...@nemeroff.com mailto:br...@nemeroff.com wrote: Urmi, You log shows the call having failed. I'm not sure why you think it runs for the proper duration. But as far as OpenSIPs is concerned, the call failed. It's likely a problem in your sipp scenario. It's very possible that sipp thinks the call is up, but the proxy does not. In any case, OpenSIPs is behaving as expected, the call fails, the dialog is destroyed. Aug 19 17:46:27 [6060] DBG:dialog:next_state_dlg: dialog 0x2d55af90 changed from state 1 to state 5, due event 1 Aug 19 17:46:27 [6060] DBG:dialog:dlg_onreply: dialog 0x2d55af90 failed (negative reply) BTW, a negative reply is =400 (or may also include = 300, can't remember). Check your traces, see where that comes from. -Brett On Thu, Aug 20, 2009 at 9:02 AM, urmi lakkad urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com wrote: Hello Stanisław Pitucha, Thank you for support. No, My call is established perfectly and is running for the specified duration without fail. I m firing the call using SIPp. Also, the dialog state gives me 1. Thanks for ur attention. -Urmi 2009/8/20 Stanisław Pitucha virap...@gmail.com mailto:virap...@gmail.com mailto:virap...@gmail.com mailto:virap...@gmail.com 2009/8/20 urmi lakkad urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com mailto:urmi.lak...@gmail.com: Am I doing right or not ? If not, please tell me the correct way. One more thing, Is my configuration is correct or not ?? It looks like your call doesn't even get accepted: Aug 19 17:46:27 [6060] DBG:dialog:next_state_dlg: dialog 0x2d55af90 changed from state 1 to state 5, due event 1 Aug 19 17:46:27 [6060] DBG:dialog:dlg_onreply: dialog 0x2d55af90 failed (negative reply) Maybe you require authentication, or something else? Just take care of the call not failing first. So far it's rejected before an OK answer (state 1 is after sending an INVITE, state 5 is deleted - more or less). Capture the traffic and see what's going on. -- Kind regards, Stanisław Pitucha, Gradwell Voip Engineer T: 01225 800 831 | F: 01225 800 801 | E: s...@gradwell.net mailto:s...@gradwell.net mailto:s...@gradwell.net mailto:s...@gradwell.net | www.gradwell.com http://www.gradwell.com http://www.gradwell.com Gradwell – Internet for Business People Phone Services | Business Broadband | Email Website Hosting Can switching to VoIP today put some change in your pocket? Registered Address: 26 Cheltenham Street, Bath, BA2 3EX, UK.
Re: [OpenSIPS-Users] Regrarding is_user_in problem in opensips-1.5
Hi Bogdan,
Thank You for reply. It was network mapping issue and not opensips
issue. I solved it.
On Mon, Aug 24, 2009 at 4:09 PM, Bogdan-Andrei Iancu bog...@voice-system.ro
wrote:
Hi Ashwini,
As your script shows, you do either IP auth (allow trusted) or digest
auth , but credentials are present only after digest auth.
So, if it is a trusted peer, there will be no digest auth, no
credentials and is_user_in() will fail.
My advice is to replace is_user_in(credentials); with
is_user_in(from); - anyhow you required both FROM USERNAME and AUTH
USERNAME to be the same when doing check_from().
Regards,
Bogdan
ASHWINI NAIDU wrote:
Hi Bogdan,
Authen tication is done
*# - auth_db params -
/* uncomment the following lines if you want to enable the DB based
authentication */
modparam(auth_db, calculate_ha1, yes)
modparam(auth_db, password_column, password)
modparam(auth_db, db_url,
mysql://opensips:opensip...@localhost/opensips)
modparam(auth_db, load_credentials, )
*
*if (is_from_local()){
# From an internal domain - check the credentials and the FROM
if (method==MESSAGE) {
log(1,\n-- ROUTE
3 MESSAGE Looop---\n);
route(17);
};
if(!allow_trusted()){
if (!proxy_authorize(,subscriber)) {
proxy_challenge(,0);
exit;
} else if(!check_from()) {
sl_send_reply(403, Forbidden, use From=ID);
exit;
};
};
if (client_nat_test(3)) {
append_hf(P-hint: setflag7|forcerport|fix_contact\r\n);
setbflag(7);
force_rport();
fix_contact();
};
#unconditional call forward
if(avp_db_load($ru/username,$avp(s:callfwd))) {
avp_pushto($ru, $avp(s:callfwd));
route(1);
exit;
}
* *consume_credentials();*
* if (uri=~^sip:00[0-9]{6,20}@) {
if (is_user_in(Credentials,local)) {
route(6);
log(1,\n*** I AM GOING TO ENTER ROUTE
4);
route(4);
exit;
} else {
sl_send_reply(403, No permissions for local calls);
exit;
};
};*
Can you tell me where i may be going wrong
This is the piece of script
On Fri, Aug 21, 2009 at 5:58 PM, Bogdan-Andrei Iancu
bog...@voice-system.ro mailto:bog...@voice-system.ro wrote:
HI Ashwini,
If you wan to used the Credentials, then you need to be sure you did
authentication before (in script).
Regards,
Bogdan
ASHWINI NAIDU wrote:
Hi all,
I have installed opensips-1.5. I have applied the required
patch
for group. When i use
* is_user_in(Credentials, local) { *
I get the following error
*ERROR:auth:consume_credentials: no authorized credentials found
(error in scripts)
Aug 21 17:09:30 debian /sbin/opensips[18916]:
ERROR:group:get_username_domain: no authorized credentials found
(error in scripts)
Aug 21 17:09:30 debian /sbin/opensips[18916]:
ERROR:group:is_user_in:
failed to get usern...@domain*
Can anyone say what may be the problem.
--
Thanking You,
Ashwini BR Naidu
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--
Thanking You,
Ashwini BR Naidu
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--
Thanking You,
Ashwini BR Naidu
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Re: [OpenSIPS-Users] 1.5.2 dispatcher module behaviour
It's working now as expected. Thanks On Thu, Aug 13, 2009 at 12:59 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Taner, Taner Sener wrote: Hi Bogdan, On Wed, Aug 12, 2009 at 3:56 PM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Hi Taner, Taner Sener wrote: Hi, I'm using Opensips 1.5.2 to distribute incoming calls to my clients using dispatcher module. I'm keeping my gateway list in db_mysql and use ds_select_dst(1, 4); to select a gateway using round-robin algorithm. I have a few issues about the module behaviour. - The first one is about pinging. I've configured dispatcher to send ping requests every 20 seconds. But if destination is not available, ping requests are repeated every 4 seconds. I guess there is another module which repeats the unresponded sip messages. How can I prevent this and change the repeat timeout about this? there should be no second module to do the pinging, and there is no way the module can dynamically change the pinging interval. try enabling full debug (debug=6) and look for the log messages like: probing set #n, URI I looked inside logs and found DBG:dispatcher:ds_check_timer: probing set #1, URI sip: lines there. So i guess it means that timer has expired and dispatcher is sending SIP OPTIONS at that time. But later found that TM module was enabled in my configuration and it was TM retransmitting SIP OPTIONS to dead destinations (with T2_timer which is 4 seconds). I can increase T2_timer but it will effect other messages, so I will leave it as is. AhaThe dispatcher module uses TM support for sending the pings in a statefull manner - so, if there is no reply at all, the TM will do retransmission of the original request it send. It was not clear from your original email if new OPTIONS are fire (at each 2 secs) or what you are simply retransmissions (copies) of the pings that were already sent out. You not control the retransmission interval via T1 and T2 params in TM (see http://www.opensips.org/html/docs/modules/1.5.x/tm.html#id228598), but note that this will have a global impact. Also you can configure how long the retransmission will be done via the fr_timer (see http://www.opensips.org/html/docs/modules/1.5.x/tm.html#id271112). - The second issue is about selecting gateways. When I receive busy from one of the destinations I'm calling ds_next_dst() and this returns me a destination which is not alive and does not respond to ping requests. I'm expecting to have only destinations which are alive, and don't understand why it is returned. Another issue here is: I'm sending INVITE request to this dead destination and dead host is not responding as expected. After that, every 4 seconds INVITE request is repeated for this dead destination. you should call the ds_mark_dst() function from failure route, when you detect a destination as failed (and before the ds_next_dst() ). See: http://www.opensips.org/html/docs/modules/1.5.x/dispatcher.html#id271344 I thought that if a destination is not alive and not responding to PING requests (in my case Destination Unreachable ICMP messages are received), it is marked as failure route automatically, but it looks like I must mark it by myself. At this point I want to ask if I can listen for results of PING resuls. So if I receive REPLY I will mark it as healthy and if PING timeout occurs I can mark it as dead. BTW are Destination Unreachable ICMP messages identified by opensips? There are two ways to mark (as failed) a destination: 1) from script, via ds_mark_dst() function, based on the negative replies you get when routing traffic to your destination. 2) automatically, based on 408 replies received. You should see in logs debug like: OPTIONS-Request was finished with code XX (to xx, group ) Setting the probing state failed (xx, group XX) Regards, Bogdan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Next OpenSIPS releases
Hi Bogdan, speaking about the futur, what do you think about the B2B_ENTITIES and B2B_LOGIC module, to be realistic/strategic, you will delay this to a futur version maybe even 2.x, will this require modification to the core ? Personnaly, I want to take some time to test them in a lab, I am curious to see how it is implemented. I beleive that a lot of users could benefit from this, in term of : - security : hide interconnection information - interoperability : handle/fix possible compatibility problem between other SIP UA Opensips could then become a one peice solution for SIP carrier, as now they may require an SBC as well for such concern Thank you for your continuous effort and accomplishement. From: users-boun...@lists.opensips.org on behalf of Bogdan-Andrei Iancu Sent: Tue 25/08/2009 9:59 AM To: users@lists.opensips.org; de...@lists.opensips.org Subject: [OpenSIPS-Users] Next OpenSIPS releases Hi, Here are the plans for the next OpenSIPS releases (minor and major). This is an initial draft (content and dates), so please comment and contribute (if necessary): 1) Minor release 1.5.3 --- Why: This is needed as more than 50 fixes were done on the 1.5 branch since 1.5.2. Date: during this week Pending: personally I'm hunting an memory leak in SNMP module (mainly design issues). If someone is aware of any other issues that requires fixing in 1.5 branch, please speak up. 2) Major release 1.6.0 --- Code freeze: estimated for mid September (depending of how fast the pending work is completed) Release date: estimated for October What we have so far: http://www.opensips.org/Main/Ver160 Pending work: - adding context for PVs (like reply, request) - route types - init, onreply per branch, timer based - dialog - early dialog support to be finished; new functions to check dialog consistency (cseq numbers, route set, contacts); dialog direction function - pike enhancement for catching more events (replies, non-SIP traffic attacks) - json support If there is something missing or if somebody is working some (new) code and needs time and support, please let me know. Regards, Bogdan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] B2BUA module question
All,Question about the direction of the B2BUA module. I know one of the key feature is topology hiding. Does this also occur in the SDP? I would expect that it would need to still be paired with something like mediaproxy or rtpproxy to achieve topology hiding with SDP as well, is this correct? Do you expect the B2BUA module will ever integrate into any of the media proxying solutions? Also, what's the possibility of doing things like changing headers, removing headers and such. For example, internally, I may have an X-Account-Number: field that is used between servers and I never want an request from the outside to ever come in with one of those and likewise I don't ever want a request to go out with one of those. I know a lot of that can be done in the script already, but I'm wondering if the B2BUA portions have any special handling for that kind of thing (ie: remove all non-standard headers). Also, there are a lot of non-rfc-ish things that I have to do on a regular basis that a B2BUA always performs better. For example, I have partners that insist on specific formatting of the From or To headers (like adding or removing prefixes to from/to headers.. yes.. I know..). Thanks! -Brett ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] B2BUA module question
Brett Nemeroff wrote: Question about the direction of the B2BUA module. I know one of the key feature is topology hiding. Does this also occur in the SDP? I would expect that it would need to still be paired with something like mediaproxy or rtpproxy to achieve topology hiding with SDP as well, is this correct? Yes. -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] B2BUA module question
Hi Brett, Brett Nemeroff wrote: All, Question about the direction of the B2BUA module. I know one of the key feature is topology hiding. Does this also occur in the SDP? I would expect that it would need to still be paired with something like mediaproxy or rtpproxy to achieve topology hiding with SDP as well, is this correct? Do you expect the B2BUA module will ever integrate into any of the media proxying solutions? As you correctly assumed, the B2BUA implementation in OpenSIPS is only a signaling B2BUA and it does not deal with sdp. The media will still go end to end and you need to use something like rtpproxy for a full b2b. Also, what's the possibility of doing things like changing headers, removing headers and such. For example, internally, I may have an X-Account-Number: field that is used between servers and I never want an request from the outside to ever come in with one of those and likewise I don't ever want a request to go out with one of those. I know a lot of that can be done in the script already, but I'm wondering if the B2BUA portions have any special handling for that kind of thing (ie: remove all non-standard headers). Also, there are a lot of non-rfc-ish things that I have to do on a regular basis that a B2BUA always performs better. For example, I have partners that insist on specific formatting of the From or To headers (like adding or removing prefixes to from/to headers.. yes.. I know..). The headers that are now taken from the initial message and inserted in the message sent on the other side are: Supported, Require, Proxy-Require, Accept and Content-Type. We can extend the rules action part to include this one of requesting a certain header to be added since it can indeed be useful. But the one with formatting the to or from header in a certain way is quite hard to express as a rule.. regards, Anca Thanks! -Brett ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] B2BUA module question
Anca,What I was imaging was something like the dialplan module to perform the rewrite, and the B2BUA module to track was it was originally so the reverse direction doesn't include the translation. That kind of thing. inside - b2bua + translation --- outside see's translated TO URI outside with translated TOURI -b2bua - inside see's restored TO URI (original TO URI) Forgive me for not entirely understanding the B2BUA scenarios and rules quite yet. ;) -Brett On Tue, Aug 25, 2009 at 10:28 AM, Anca Vamanu a...@opensips.org wrote: Hi Brett, Brett Nemeroff wrote: All, Question about the direction of the B2BUA module. I know one of the key feature is topology hiding. Does this also occur in the SDP? I would expect that it would need to still be paired with something like mediaproxy or rtpproxy to achieve topology hiding with SDP as well, is this correct? Do you expect the B2BUA module will ever integrate into any of the media proxying solutions? As you correctly assumed, the B2BUA implementation in OpenSIPS is only a signaling B2BUA and it does not deal with sdp. The media will still go end to end and you need to use something like rtpproxy for a full b2b. Also, what's the possibility of doing things like changing headers, removing headers and such. For example, internally, I may have an X-Account-Number: field that is used between servers and I never want an request from the outside to ever come in with one of those and likewise I don't ever want a request to go out with one of those. I know a lot of that can be done in the script already, but I'm wondering if the B2BUA portions have any special handling for that kind of thing (ie: remove all non-standard headers). Also, there are a lot of non-rfc-ish things that I have to do on a regular basis that a B2BUA always performs better. For example, I have partners that insist on specific formatting of the From or To headers (like adding or removing prefixes to from/to headers.. yes.. I know..). The headers that are now taken from the initial message and inserted in the message sent on the other side are: Supported, Require, Proxy-Require, Accept and Content-Type. We can extend the rules action part to include this one of requesting a certain header to be added since it can indeed be useful. But the one with formatting the to or from header in a certain way is quite hard to express as a rule.. regards, Anca Thanks! -Brett ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] B2BUA module question
The headers that are now taken from the initial message and inserted in the message sent on the other side are: Supported, Require, Proxy-Require, Accept and Content-Type. We can extend the rules action part to include this one of requesting a certain header to be added since it can indeed be useful. I'd be really happy to see this, as I stated on the dev list. It would be nice if we had a configuration parameter so that all custom headers are passed to the other leg, so OpenSIPS could act as a transparent b2bua :) -- /Saúl http://www.saghul.net | http://www.sipdoc.net ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Next OpenSIPS releases
Hi Julien, Julien Chavanton wrote: Hi Bogdan, speaking about the futur, what do you think about the B2B_ENTITIES and B2B_LOGIC module, to be realistic/strategic, you will delay this to a futur version maybe even 2.x, will this require modification to the core ? Not sure how you see this, but the B2BUA related modules are part of the 1.6 release. I admit they are in alpha-stage and probably it will take some more time (in the next releases) to (1) become stable and (2) functionality consistent. Personnaly, I want to take some time to test them in a lab, I am curious to see how it is implemented. I beleive that a lot of users could benefit from this, in term of : - security : hide interconnection information - interoperability : handle/fix possible compatibility problem between other SIP UA and also there is a third case, quite large, when comes to implementing complex SIP scenarios (not possible by a simple proxy) which does not need any media manipulation (we still have a signalling b2bua) - like the examples we have in the tutorial, with inserting announcements in the call, tele-marketing, etc...More or less better integration of calls with media services. Opensips could then become a one peice solution for SIP carrier, as now they may require an SBC as well for such concern Indeed. Thank you for your continuous effort and accomplishement. Thank you :) Regards, Bogdan *From:* users-boun...@lists.opensips.org on behalf of Bogdan-Andrei Iancu *Sent:* Tue 25/08/2009 9:59 AM *To:* users@lists.opensips.org; de...@lists.opensips.org *Subject:* [OpenSIPS-Users] Next OpenSIPS releases Hi, Here are the plans for the next OpenSIPS releases (minor and major). This is an initial draft (content and dates), so please comment and contribute (if necessary): 1) Minor release 1.5.3 --- Why: This is needed as more than 50 fixes were done on the 1.5 branch since 1.5.2. Date: during this week Pending: personally I'm hunting an memory leak in SNMP module (mainly design issues). If someone is aware of any other issues that requires fixing in 1.5 branch, please speak up. 2) Major release 1.6.0 --- Code freeze: estimated for mid September (depending of how fast the pending work is completed) Release date: estimated for October What we have so far: http://www.opensips.org/Main/Ver160 Pending work: - adding context for PVs (like reply, request) - route types - init, onreply per branch, timer based - dialog - early dialog support to be finished; new functions to check dialog consistency (cseq numbers, route set, contacts); dialog direction function - pike enhancement for catching more events (replies, non-SIP traffic attacks) - json support If there is something missing or if somebody is working some (new) code and needs time and support, please let me know. Regards, Bogdan ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] init.d file?
Look at packaging/debian directory, file is opensips.init. -- /Saúl http://www.saghul.net | http://www.sipdoc.net ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] init.d file?
so i changed the name of the file to opensips and moved it into init.d, but it does not respond to any commands. any hints you could give me? On Tue, Aug 25, 2009 at 3:38 PM, Saúl Ibarra sag...@gmail.com wrote: Look at packaging/debian directory, file is opensips.init. -- /Saúl http://www.saghul.net | http://www.sipdoc.net ___ 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] init.d file?
can anyone point me to a current init.d script for starting/stopping opensips? am trying use in conjunction with monit and have an older init.d from openser days thanks in advance! ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] drouting module
Hi, When applying the do_routing function without specifying the group id, what are the matching patterns that it will look for? In other words If I want ALL the users behind domain test.com how would I do it? Leaving the username field empty and putting test.com in the domain did not work. Thanks Sebastian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] init.d file?
Did you make it executable? You may also need to adjust the script to point to the proper binary location. Open up the script, it's not too tricky. I'm pretty sure that this line: test -f $DAEMON || exit 0 Says, If the binary isn't there, just quietly die, without giving the user a useful error message /some sarcasm -Brett On Tue, Aug 25, 2009 at 3:57 PM, Alex G greekman0...@gmail.com wrote: so i changed the name of the file to opensips and moved it into init.d, but it does not respond to any commands. any hints you could give me? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] drouting module
stop asking so much! ;) D El 25/08/2009, a las 18:05, Sebastian Sastre sebast...@next-ip.com escribió: Hi, When applying the do_routing function without specifying the group id, what are the matching patterns that it will look for? In other words If I want ALL the users behind domain test.com how would I do it? Leaving the username field empty and putting test.com in the domain did not work. Thanks Sebastian ___ 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] XCAP scalability: Integrated xcap server VS xcap_client mode
Hi, I've read somewhere (but cannot find it right now) that integrated xcap mode (presence module getting XCAP documents vía SQL) is suitable for small environments. I know that the other way, using xcap_client module, is not really suitable for now as the HTTP request is blocking (the opensips process gets blocked until the XCAP server replies). But let's imagine this issue is solved. So we have two options: 1) Integrated server: OpenSIPS presence module gets the documents from the xcap table. 2) XCAP client mode: OpenSIPS acts as a xcap client to get the documents from the XCAP server. In the option 1: - OpenSIPS does a SQL query to get the document (faster than a HTTP request). - So the XCAP server is not queried by the presence server (less work for the XCAP server). - There could be various XCAP servers running at the same time (perhaps DNS random or a http proxy between clients and XCAP servers), all of them storing the documents in same DB. And the presence document uses directly that DB. In the option 2: - OpenSIPS must perform a HTTP request which takes more time than a SQL request (even if the DB is in other host), right? - The XCAP server receives a XCAP request from the presence server, so the XCAP server must work. - The xcap client would contact just an unique XCAP server (it would learn the IP after the first DNS resolution, so ramdom DNS is not valid here). - Solution: Using a HTTP proxy between OpenSIPS and various XCAP servers. Is it really the option 2 more suitable in order to favour scalability and big environments? -- Iñaki Baz Castillo i...@aliax.net ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OPenSIPS with SQLite3 database
Hello Everybody, This is my first post to this list, so greeting to all of you. I would like to know if anybody has successfully use OpenSIPS with SQLite3 database. I suppose we could use it through unixodbc, but is there any plan to support native interface like for MySQL? Thanks a lot in advance for your response. Kind regards, Anto ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] init.d file?
not using the dpkg because i have all my opensips servers built already. for the sake of using monit, i require an init.d file to be there to start/stop opensips. i've changed the pointer for DAEMON and added the default file. I'm getting Starting opensips: opensips already running. but its obviously not running. I can see I'm getting closer, but still missing something... any more tips? :) On Tue, Aug 25, 2009 at 5:14 PM, Saúl Ibarra sag...@gmail.com wrote: You'll need to make it executable as well as copy the opensips.default file to /etc/default. That should do the job :) Anyway, why don't you make the debian packages so that you install opensips by just doing dpkg -i and everything is done automagically? -- /Saúl http://www.saghul.net | http://www.sipdoc.net ___ 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] Multiple Area Codes in Customer Area
Got it working! Man OpenSIPS sure can do anything with SIP
So here is what I did for future searchers
So the users account is a 7 digit DID number XXX at blah.com
I set up an AVP called areacode for the whole domain blah.com (this assumes
that the whole domain blah.com is only in one areacode)
opensipsctl avp add -T usr_preferences 0 at blah.com areacode 1 201
opensipsctl avp add -T usr_preferences 0 at foo.com areacode 1 339
In the opensips.cfg file I do the following (it depends on your config as to
where you want to put this)
if (uri=~^sip:[2-9][0-9]{6}@) {
avp_db_load($ru/domain,$avp(s:areacode));
subst_uri('/^sip:([0-9]+)@(.*)$/sip:$avp(s:areacode)\...@\2/i');
};
So when someone calls a 7 digit number the avp_db_load() loads the variable
for areacode and the subst_uri adds the areacode at the beginning of the
Request-URI.
Bogdan-Andrei Iancu wrote:
Hi Duane,
You can correlate AVPs you a USER, a DOAMAIN, etc - it is up to you,
from the script, when loading the AVP - is a pure logical mapping.
Regards,
Bogdan
osiris123d wrote:
I was reading Flavio's Building Telephony Systems with OpenSER chapter
about AVPOPs and he mentions that AVP's can be used for a whole domain.
I
was thinking that I might be able to configure a area code for Company
A's
domain and then route calls that way. If not that then I can set the AVP
on
the fly within the transaction by looking at the callers Request URI's
first
3 digits and route it appropriately.
Bogdan-Andrei Iancu wrote:
Hi,
Requirements on the format of CONTACT and TO headers are nonsense as
they are not used for routing at all. Only FROM (which provides info on
the caller) and RURI (request URI) (which provide info on callee).
So, bottom line, only the normalization of the RURI should be required
on the system.
Regards,
Bogdan
osiris123d wrote:
Thanks for the info.
I will look into this and work up a config.
I also got this direct email about my post from someone else who lives
in
the US. I figured I would go ahead and post below what he sent just so
its
out there.
Hello Duane --
You have hit on one of the more difficult areas in SIP and telephony in
general -- especially here in the North American Numbering Plan. Below
I
will address the problem in general, and not particularly related to
the
OpenSIPs question, because IMO you need a solution that will work in
any
architecture, not just OpenSIPs.
In a nutshell, I recommend that for your USA users you:
1.) Require From: and Contact: headers to be in NANPA National (10
digit)
format. This is method is standard in the telephone industry, and will
allow easy integration with North American ANI or Caller ID format,
especially when a call may eventually be handed off to the PSTN.
2.) Require incoming To: headers to be in e.164 International format,
i.e.
NANPA-destination numbers all begin with the 1 digit, followed by the
10
digit National number. Any incoming call to 612xxx goes to
Sydney,
Austrailia, and not Minneapolis, MN. This requirement should be
enforced
at
the perimeter of your network, where Customer Equipment can enforce the
local digit normalization policy.
3.) If you can't enforce #2 above, you will need to Normalize
incoming
calls to the e.164 International format prior to routing. The
unfortunate
reality here in the USA is that the requirements for how many digits to
dial
for a given destination (the dialing plan) depends on where the call
comes
from. Here in the Chicago area, residents of the 847 area code must
today
dial all calls as 11 digits. Calls within the 312 or 773 area code may
today be dialed as 7 digits, however beginning on 07 November, all
calls
originating in 312 and 773 must be dialed as 1+10 digits, due to the
new
872
overlay area code.For more information, see
http://www.nanpa.com/reports/NPA_Dialing_Plans_05_09.pdf
4.) Finally, if you have any termination carriers who need special
prefixes, append them after you have made your route selection.
If you would like further information or discussion, please feel free
to
contact me.
John S. Robi
j...@communxxx.com
Bogdan-Andrei Iancu wrote:
Hi there,
When you have to deal with local dialling you need consider the amount
of information yon need to keep in order to translate to national
format
and the complexity of the processing you have to do.
A compromise solution will be to keep in user profile some information
about the location (like for US, the 2 digits Id of the state) - this
will reduce the amount of data you need to keep per user. Also, this
info can be loaded at auth time, using load_credentials parameter
(just an example).
Now, using the location information, you can use dialplan to do the
actual transformation. Like, if location is NJ (use a separate plan):
if 7 digits - put 011-201 prefix
if 10 digits -
[OpenSIPS-Users] how to uninstall
Dear all, I have installed OpenSIPS-1.5.2 from source with make prefix=/usr/local . But I need to change prefix=/ or uninstall opensips completely. Sorry for my body language. Sincerely, Tseveen. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
