Hello,
I am using Textops Module, but i have one problem. I am able to change sdp parameters to force the use of 1 codec, below an exemple : SDP from caller : m=audio 9999 RTP/AVP 107 119 98 101 SDP to callee : m=audio 9999 RTP/AVP 98 101
My problem is, i want to make sdp parameters cleaner :) I want to remove a=rtpmap:107 and a=rtmap:119 from the body..
I am using textops function for that: subst_body('/a=rtpmap:107 BV32/16000 //');
it removed it well from the sdp parameters but leave an empty line :( like this :
a=alt:2 1 : AWxJfaLh kK2my/RE 10.10.10.3 8588 a=fmtp:101 0-15
a=rtpmap:98 iLBC/8000= a=rtpmap:101 telephone-event/8000 a=sendrecv a=nortpproxy:yes
it seems that the callee can't read the sdp session and the call is rejected.
Any idea how to remove these empty lines ?
cheers,
karhu
2009/5/27 karhu yoannks@gmail.com:
Hello,
I am using Textops Module, but i have one problem. I am able to change sdp parameters to force the use of 1 codec, below an exemple : SDP from caller : m=audio 9999 RTP/AVP 107 119 98 101 SDP to callee : m=audio 9999 RTP/AVP 98 101
My problem is, i want to make sdp parameters cleaner :) I want to remove a=rtpmap:107 and a=rtmap:119 from the body..
Theorically it's not required at all. These lines "a=rtpmap" are just a textual description. The could appear or not, the only important line is "m=audio...".
I am using textops function for that: subst_body('/a=rtpmap:107 BV32/16000 //');
it removed it well from the sdp parameters but leave an empty line
Not sure if it's possible to delet a line. Perhaps you could play with the "s" flag (match newline don't treat it as end of line).
Theorically it's not required at all. These lines "a=rtpmap" are just a textual description. The could appear or not, the only important line is "m=audio...".
Sure it is useless to delete it but i have to ;) Btw it is possible to add a line in the body ? i don't find any functions doing that in Textops. I need to manipulate SDP parameters.
Not sure if it's possible to delet a line. Perhaps you could play with the "s" flag (match newline don't treat it as end of line).
I tried to use "s" flag at the end of subst_body functions but doesn't change anything. I used also replace_body function but it is the same result as subst_body function.
Cheers,
Karhu
2009/5/27 karhu yoannks@gmail.com:
Theorically it's not required at all. These lines "a=rtpmap" are just a textual description. The could appear or not, the only important line is "m=audio...".
Sure it is useless to delete it but i have to ;)
Could I know why? (just curiosity).
Btw it is possible to add a line in the body ? i don't find any functions doing that in Textops. I need to manipulate SDP parameters.
Not sure, but perhaps this could work:
subst_body('/a=rtpmap:107 BV32/16000 /SAME_LINE\r\nNEW_LINE/');
In SAME_LINE you should use the typical variable to copy the match (maybe $1)?
Not sure, but perhaps this could work:
subst_body('/a=rtpmap:107 BV32/16000 /SAME_LINE\r\nNEW_LINE/');
In SAME_LINE you should use the typical variable to copy the match (maybe
$1)?
subst_body('/(a=rtpmap:107 BV32/16000) /\1\r\nNEW_LINE/'); like this it works :) thanks.
It should be possible to delete empty line with perl substitution, have no clue about how to do that.
Cheers
Do you have any ideas how to remove blank line with per substitution and textops functions ?
Cheers, Karhu
2009/5/28 karhu yoannks@gmail.com:
Do you have any ideas how to remove blank line with per substitution and textops functions ?
Never tryed it.
Hello,
I'm still on removing SDP parameters from the SIP Invite method. So here is the initial SDP from caller :
v=0 o=- 5 2 IN IP4 10.10.10.3 s=CounterPath X-Lite 3.0 c=IN IP4 10.10.10.3 t=0 0 m=audio 38028 RTP/AVP 107 119 0 98 8 3 101 a=alt:1 2 : D/EiVcOe ZNbDTRu7 10.193.15.206 38028 a=alt:2 1 : 4oNCIzwV oJwrCrfw 10.10.10.3 38028 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:119 BV32-FEC/16000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv m=video 17166 RTP/AVP 115 34 a=alt:1 2 : UQl6viQz cTcTrMFc 10.193.15.206 17166 a=alt:2 1 : US97TDnv HVVaJHFF 10.10.10.3 17166 a=fmtp:115 QCIF=1 I=1 J=1 K=1 MAXBR=1960 a=fmtp:34 QCIF=1 MAXBR=1960 a=rtpmap:115 H263-1998/90000 a=rtpmap:34 H263/90000 a=sendrecv
I just want to force the use of payload 98 on the audio part, so in kamailio.cfg i added that :
##### INVITE ###### if (is_method("INVITE")){ route(4); record_route();
# route(10); route(20); route(21); }
######### Route Rtpproxy Bridge Mode for INVITE methods ########### route[4] { if (lookup("location_internal")) { if (dst_ip == 10.10.10.2) if (force_rtp_proxy("FAII")) xlog("L_INFO","appel_rtpproxyFAII"); t_on_reply("1"); if (dst_ip == 192.168.0.2) if (force_rtp_proxy("FAEI")) xlog("L_INFO","appel_rtpproxyFAEI"); t_on_reply("1"); } else if (lookup("location_external")) { if (dst_ip == 10.10.10.2) if (force_rtp_proxy("FAIE")) xlog("L_INFO","appel_rtpproxyFAIE"); t_on_reply("1"); if (dst_ip == 192.168.0.2) if (force_rtp_proxy("FAEE")) xlog("L_INFO","appel_rtpproxyFAEE"); t_on_reply("1"); } else { sl_send_reply("403", "Call cannot be served here"); exit; }; }
route[20] { if(has_body("application/sdp") && search_body("m=audio")) xlog("L_INFO", "application/sdp_present_avec_audio"); subst_body('/AVP (.*) /AVP 98 /'); }
route[21] { if(has_body("application/sdp") && search_body("a=rtpmap:[0-9]+")) xlog("L_INFO", "application/sdp_present_avec_description_audio"); subst_body('/a=rtpmap:107 BV32/16000//'); subst_body('/a=rtpmap:119 BV32-FEC/16000//'); # replace_body("a=rtpmap:107 BV32/16000",""); # subst_body('/a=rtpmap:[0-9]+ (.*)//'); }
and i get from wireshark capture the following sdp :
v=0 o=- 5 2 IN IP4 10.10.10.3 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.0.2 t=0 0 m=audio 35044 RTP/AVP 98 101 a=alt:1 2 : D/EiVcOe ZNbDTRu7 10.193.15.206 38028 a=alt:2 1 : 4oNCIzwV oJwrCrfw 10.10.10.3 38028 a=fmtp:101 0-15
a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv m=video 35046 RTP/AVP 115 34 a=alt:1 2 : UQl6viQz cTcTrMFc 10.193.15.206 17166 a=alt:2 1 : US97TDnv HVVaJHFF 10.10.10.3 17166 a=fmtp:115 QCIF=1 I=1 J=1 K=1 MAXBR=1960 a=fmtp:34 QCIF=1 MAXBR=1960 a=rtpmap:115 H263-1998/90000 a=rtpmap:34 H263/90000 a=sendrecv a=nortpproxy:yes
First, the 98 payload is well forced. The problem is i get "Call failed: Not acceptable Here" on the caller sip phone. Obviously the callee phone doesn't ring.. It seems that the blanck lines in the sdp message shouldn't prevent the callee phone to ring. Do you have any ideas ? or configuration modifications ? i am stick on that problem.
Cheers,
Karhu
Hello,
you should remove end of lines as well.
Cheers, Daniel
On 06/03/2009 11:20 AM, karhu wrote:
Hello,
I'm still on removing SDP parameters from the SIP Invite method. So here is the initial SDP from caller :
v=0 o=- 5 2 IN IP4 10.10.10.3 s=CounterPath X-Lite 3.0 c=IN IP4 10.10.10.3 t=0 0 m=audio 38028 RTP/AVP 107 119 0 98 8 3 101 a=alt:1 2 : D/EiVcOe ZNbDTRu7 10.193.15.206 38028 a=alt:2 1 : 4oNCIzwV oJwrCrfw 10.10.10.3 38028 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:119 BV32-FEC/16000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv m=video 17166 RTP/AVP 115 34 a=alt:1 2 : UQl6viQz cTcTrMFc 10.193.15.206 17166 a=alt:2 1 : US97TDnv HVVaJHFF 10.10.10.3 17166 a=fmtp:115 QCIF=1 I=1 J=1 K=1 MAXBR=1960 a=fmtp:34 QCIF=1 MAXBR=1960 a=rtpmap:115 H263-1998/90000 a=rtpmap:34 H263/90000 a=sendrecv
I just want to force the use of payload 98 on the audio part, so in kamailio.cfg i added that :
##### INVITE ###### if (is_method("INVITE")){ route(4); record_route();
# route(10); route(20); route(21); }
######### Route Rtpproxy Bridge Mode for INVITE methods ########### route[4] { if (lookup("location_internal")) { if (dst_ip == 10.10.10.2) if (force_rtp_proxy("FAII")) xlog("L_INFO","appel_rtpproxyFAII"); t_on_reply("1"); if (dst_ip == 192.168.0.2) if (force_rtp_proxy("FAEI")) xlog("L_INFO","appel_rtpproxyFAEI"); t_on_reply("1"); } else if (lookup("location_external")) { if (dst_ip == 10.10.10.2) if (force_rtp_proxy("FAIE")) xlog("L_INFO","appel_rtpproxyFAIE"); t_on_reply("1"); if (dst_ip == 192.168.0.2) if (force_rtp_proxy("FAEE")) xlog("L_INFO","appel_rtpproxyFAEE"); t_on_reply("1"); } else { sl_send_reply("403", "Call cannot be served here"); exit; }; }
route[20] { if(has_body("application/sdp") && search_body("m=audio")) xlog("L_INFO", "application/sdp_present_avec_audio"); subst_body('/AVP (.*) /AVP 98 /'); }
route[21] { if(has_body("application/sdp") && search_body("a=rtpmap:[0-9]+")) xlog("L_INFO", "application/sdp_present_avec_description_audio"); subst_body('/a=rtpmap:107 BV32/16000//'); subst_body('/a=rtpmap:119 BV32-FEC/16000//'); # replace_body("a=rtpmap:107 BV32/16000",""); # subst_body('/a=rtpmap:[0-9]+ (.*)//'); }
and i get from wireshark capture the following sdp :
v=0 o=- 5 2 IN IP4 10.10.10.3 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.0.2 t=0 0 m=audio 35044 RTP/AVP 98 101 a=alt:1 2 : D/EiVcOe ZNbDTRu7 10.193.15.206 38028 a=alt:2 1 : 4oNCIzwV oJwrCrfw 10.10.10.3 38028 a=fmtp:101 0-15
a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv m=video 35046 RTP/AVP 115 34 a=alt:1 2 : UQl6viQz cTcTrMFc 10.193.15.206 17166 a=alt:2 1 : US97TDnv HVVaJHFF 10.10.10.3 17166 a=fmtp:115 QCIF=1 I=1 J=1 K=1 MAXBR=1960 a=fmtp:34 QCIF=1 MAXBR=1960 a=rtpmap:115 H263-1998/90000 a=rtpmap:34 H263/90000 a=sendrecv a=nortpproxy:yes
First, the 98 payload is well forced. The problem is i get "Call failed: Not acceptable Here" on the caller sip phone. Obviously the callee phone doesn't ring.. It seems that the blanck lines in the sdp message shouldn't prevent the callee phone to ring. Do you have any ideas ? or configuration modifications ? i am stick on that problem.
Cheers,
Karhu
you should remove end of lines as well.
What do you mean by that ?
On 06/03/2009 02:09 PM, karhu wrote:
you should remove end of lines as well.
What do you mean by that ?
you removed only content of the lines, not including the end of it (e.g., \r, \n).
Daniel
Ok thanks :)
I've tried but doesn't delete the line anymore, i tried with : subst_body('/a=rtpmap:107 BV32/16000\r//'); subst_body('/a=rtpmap:107 BV32/16000\n//'); subst_body('/a=rtpmap:107 BV32/16000\r\n//'); subst_body('/a=rtpmap:107 BV32/16000$\r\n//');
but nothing :s
On Wednesday 03 June 2009 12:41:52 karhu wrote:
Ok thanks :)
I've tried but doesn't delete the line anymore, i tried with : subst_body('/a=rtpmap:107 BV32/16000\r//'); subst_body('/a=rtpmap:107 BV32/16000\n//'); subst_body('/a=rtpmap:107 BV32/16000\r\n//'); subst_body('/a=rtpmap:107 BV32/16000$\r\n//');
It is just '$' for "end of line", not '$\r\n' you should read some regex docs ... just and advice.
Hello,
Well i bother you with that, i am pretty sure, but i have new problem with sdp modification. When i try to add a new payload in "m=video ..." line, the video port from the proxy is inserted at the end of "m=video.." line as follow :
m=video 21840 RTP/AVP 99 115 34 35186 a=alt:1 2 : TsApLPhC /VstqkH6 10.193.15.206 21840 a=alt:2 1 : 3p37pqju 6bkvXakZ 10.10.10.3 21840 a=fmtp:115 QCIF=1 I=1 J=1 K=1 MAXBR=1960 a=fmtp:34 QCIF=1 MAXBR=1960 a=rtpmap:115 H263-1998/90000 a=rtpmap:34 H263/90000 a=sendrecv a=nortpproxy:yes
21840 is the original port (from caller), the good proxy port uses for this session should be 35186. I called my function to modify the sdp body after force_rtp_proxy(). So everything should be ok. I have seen that textops functions are applied to the original request..is that why i can't solve my problem?
Below is my .conf file :
if (is_method("INVITE")){ if (route(4)) xlog("L_INFO","Route4_Ok_à[$Tf]"); setflag(10); record_route(); }
if (isflagset(10) && is_method("INVITE")){ route(30); }
route[4] { if (lookup("location_internal")) { if (dst_ip == 10.10.10.2) if (force_rtp_proxy("FAII")) xlog("L_INFO","appel_rtpproxyFAII"); t_on_reply("1"); if (dst_ip == 192.168.0.2) if (force_rtp_proxy("FAEI")) xlog("L_INFO","appel_rtpproxyFAEI"); t_on_reply("1"); } else if (lookup("location_external")) { if (dst_ip == 10.10.10.2) if (force_rtp_proxy("FAIE")) xlog("L_INFO","appel_rtpproxyFAIE"); t_on_reply("1"); return; if (dst_ip == 192.168.0.2) if (force_rtp_proxy("FAEE")) xlog("L_INFO","appel_rtpproxyFAEE"); t_on_reply("1"); } else { sl_send_reply("403", "Call cannot be served here"); exit; }; }
route[30] { if(has_body("application/sdp") && search_body("m=video")){ subst_body('/m=video ([0-9]+) RTP/AVP (.*)/m=video \1 RTP/AVP 99 \2 /'); } }
Cheers,
Karhu
On Thursday 04 June 2009 16:22:01 karhu wrote:
Hello,
Well i bother you with that, i am pretty sure, but i have new problem with sdp modification. When i try to add a new payload in "m=video ..." line, the video port from the proxy is inserted at the end of "m=video.." line as follow :
m=video 21840 RTP/AVP 99 115 34 35186 a=alt:1 2 : TsApLPhC /VstqkH6 10.193.15.206 21840 a=alt:2 1 : 3p37pqju 6bkvXakZ 10.10.10.3 21840 a=fmtp:115 QCIF=1 I=1 J=1 K=1 MAXBR=1960 a=fmtp:34 QCIF=1 MAXBR=1960 a=rtpmap:115 H263-1998/90000 a=rtpmap:34 H263/90000 a=sendrecv a=nortpproxy:yes
21840 is the original port (from caller), the good proxy port uses for this session should be 35186. I called my function to modify the sdp body after force_rtp_proxy(). So everything should be ok. I have seen that textops functions are applied to the original request..is that why i can't solve my problem?
Umm, what I guest is that you don't know that all modifications to a request take place AT THE SAME TIME just before you call t_relay, they are not executed one after the other during the script flow, all of them are called TOGETHER before t_relay, thats why if you modify the "m=video..." line by hand during the script flow and after that you call force_rtpproxy, what you get is not what you desire
Hello,
Thx Raúl for your answer. What i guess is, i need to develop my own module to play with these SDP parameters. Hard it will be :)
Cheers
Hello,
On 06/05/2009 10:37 AM, karhu wrote:
Hello,
Thx Raúl for your answer. What i guess is, i need to develop my own module to play with these SDP parameters. Hard it will be :)
maybe is not that hard as some expect. Here some docs that can help: http://www.kamailio.org/dokuwiki/doku.php/development:sdp-parsing http://www.asipto.com/pub/kamailio-devel-guide/
Cheers, Daniel
Hello,
On 06/04/2009 06:22 PM, karhu wrote:
[...]if(has_body("application/sdp") && search_body("m=video")){ subst_body('/m=video ([0-9]+) RTP/AVP (.*)/m=video \1 RTP/AVP 99 \2 /');
just a tip, if you want to get rid of escaping / you can use another character as delimiter in subst expression, e.g., '#':
subst_body('#m=video ([0-9]+) RTP/AVP (.*)#m=video \1 RTP/AVP 99 \2 #');
Cheers, Daniel
Hello,
My question is maybe stupid, but i need to know. If i develop my own module using sdp.c sdp_helpr_funcs.c functions, will it be like the use of textops module i did before? I mean the modifications will all take place before t_relay, meaning that i might have the same problem as i had (see few posts before when i wanted to add specific payload in the "m=video" part, i get the rtpproxy port at the end of the line)..
Cheers,
Karhu
[2009-06-08 14:34:52 (GMT)] An exception occurred while executing this script: Error message: #0, SQLSTATE[HY000] [2003] Can't connect to MySQL server on 'sql.sipit.be' (13) Script name and line number of error: /var/www/siremis/openbiz/others/zendfrwk/library/Zend/Db/Adapter/Pdo/Abstract.php:131
Hello Ramona,
I get this error with Siremis . [2009-06-08 14:34:52 (GMT)] An exception occurred while executing this script: Error message: #0, SQLSTATE[HY000] [2003] Can't connect to MySQL server on 'sql.sipit.be' (13) Script name and line number of error: /var/www/siremis/openbiz/others/zendfrwk/library/Zend/Db/Adapter/Pdo/Abstract.php:131
In fact, in the xml, there are 2 DB entries.. * edit file siremis/metadata/Config.xml and set MySQL server address, username, password and Kamailio (OpenSER) database name. <Database Name="Default" Driver="Pdo_Mysql" Server="xx.sipit.xx" DBName="openserx" User="zzz" Password="xxx" /> <Database Name="Siremis" Driver="Pdo_Mysql" Server="xx.sipit.xx" DBName="openserx" User="zzz" Password="xxx" />
I have a default entry and a siremis entry... Wich one has to be modified? Do I have to change the names? (even when changed, it doesn't work)
ob_usershas been created.
Kamailio version is 1.5.1 Siremis is 0.9.2
what do I wrong?
Kind regards,
Olivier (The guy who where on the Fosdem restaurant in Brussels)
hh174 wrote:
[2009-06-08 14:34:52 (GMT)] An exception occurred while executing this script: Error message: #0, SQLSTATE[HY000] [2003] Can't connect to MySQL server on 'sql.sipit.be' (13) Script name and line number of error: /var/www/siremis/openbiz/others/zendfrwk/library/Zend/Db/Adapter/Pdo/Abstract.php:131
Hello Ramona,
I get this error with Siremis . [2009-06-08 14:34:52 (GMT)] An exception occurred while executing this script: Error message: #0, SQLSTATE[HY000] [2003] Can't connect to MySQL server on 'sql.sipit.be' (13) Script name and line number of error: /var/www/siremis/openbiz/others/zendfrwk/library/Zend/Db/Adapter/Pdo/Abstract.php:131
In fact, in the xml, there are 2 DB entries..
- edit file siremis/metadata/Config.xml and set MySQL server address,
username, password and Kamailio (OpenSER) database name. <Database Name="Default" Driver="Pdo_Mysql" Server="xx.sipit.xx" DBName="openserx" User="zzz" Password="xxx" /> <Database Name="Siremis" Driver="Pdo_Mysql" Server="xx.sipit.xx" DBName="openserx" User="zzz" Password="xxx" />
I have a default entry and a siremis entry... Wich one has to be modified? Do I have to change the names? (even when changed, it doesn't work)
ob_usershas been created.
Kamailio version is 1.5.1 Siremis is 0.9.2
what do I wrong?
Kind regards,
Olivier (The guy who where on the Fosdem restaurant in Brussels)
Check the mysql permissions for the user on the database.
On the machine with siremis type:
mysql -h xx.sipit.xx -u zzz -p openserx
enter the password you set when prompted. If you get access denied, log into mysql as the admin used (usually root, but it could be something else) and type:
grant all on openserx.* to 'zzz' identified by 'xxx';
you may have to use the form of 'zzz'@'machine' to get the grant right.
when the mysql line works, siremis should work as well, or at least THAT error will go away.
Bruce
[2009-06-08 14:34:52 (GMT)] An exception occurred while executing this script: Error message: #0, SQLSTATE[HY000] [2003] Can't connect to MySQL server on 'sql.sipit.be' (13) Script name and line number of error: /var/www/siremis/openbiz/others/zendfrwk/library/Zend/Db/Adapter/Pdo/Abstract.php:131
Hello Ramona,
I get this error with Siremis . [2009-06-08 14:34:52 (GMT)] An exception occurred while executing this script: Error message: #0, SQLSTATE[HY000] [2003] Can't connect to MySQL server on 'sql.sipit.be' (13) Script name and line number of error: /var/www/siremis/openbiz/others/zendfrwk/library/Zend/Db/Adapter/Pdo/Abstract.php:131
In fact, in the xml, there are 2 DB entries.. * edit file siremis/metadata/Config.xml and set MySQL server address, username, password and Kamailio (OpenSER) database name. <Database Name="Default" Driver="Pdo_Mysql" Server="xx.sipit.xx" DBName="openserx" User="zzz" Password="xxx" /> <Database Name="Siremis" Driver="Pdo_Mysql" Server="xx.sipit.xx" DBName="openserx" User="zzz" Password="xxx" />
I have a default entry and a siremis entry... Wich one has to be modified? Do I have to change the names? (even when changed, it doesn't work)
ob_usershas been created.
Kamailio version is 1.5.1 Siremis is 0.9.2
what do I wrong?
Kind regards,
Olivier (The guy who where on the Fosdem restaurant in Brussels)
[2009-06-08 14:34:52 (GMT)] An exception occurred while executing this script: Error message: #0, SQLSTATE[HY000] [2003] Can't connect to MySQL server on 'sql.sipit.be' (13) Script name and line number of error: /var/www/siremis/openbiz/others/zendfrwk/library/Zend/Db/Adapter/Pdo/Abstract.php:131
Hello Ramona,
I get this error with Siremis . [2009-06-08 14:34:52 (GMT)] An exception occurred while executing this script: Error message: #0, SQLSTATE[HY000] [2003] Can't connect to MySQL server on 'sql.sipit.be' (13) Script name and line number of error: /var/www/siremis/openbiz/others/zendfrwk/library/Zend/Db/Adapter/Pdo/Abstract.php:131
In fact, in the xml, there are 2 DB entries.. * edit file siremis/metadata/Config.xml and set MySQL server address, username, password and Kamailio (OpenSER) database name. <Database Name="Default" Driver="Pdo_Mysql" Server="xx.sipit.xx" DBName="openserx" User="zzz" Password="xxx" /> <Database Name="Siremis" Driver="Pdo_Mysql" Server="xx.sipit.xx" DBName="openserx" User="zzz" Password="xxx" />
I have a default entry and a siremis entry... Wich one has to be modified? Do I have to change the names? (even when changed, it doesn't work)
ob_usershas been created.
Kamailio version is 1.5.1 Siremis is 0.9.2
what do I wrong?
Kind regards,
Olivier (The guy who where on the Fosdem restaurant in Brussels)
Hi Olivier,
is it working now? It is about rights as Bruce already suggested.
olivier.taylor@gmail.com wrote:
[2009-06-08 14:34:52 (GMT)] An exception occurred while executing this script: Error message: #0, SQLSTATE[HY000] [2003] Can't connect to MySQL server on 'sql.sipit.be' (13) Script name and line number of error: /var/www/siremis/openbiz/others/zendfrwk/library/Zend/Db/Adapter/Pdo/Abstract.php:131
Hello Ramona,
I get this error with Siremis . [2009-06-08 14:34:52 (GMT)] An exception occurred while executing this script: Error message: #0, SQLSTATE[HY000] [2003] Can't connect to MySQL server on 'sql.sipit.be' (13) Script name and line number of error: /var/www/siremis/openbiz/others/zendfrwk/library/Zend/Db/Adapter/Pdo/Abstract.php:131
In fact, in the xml, there are 2 DB entries..
- edit file siremis/metadata/Config.xml and set MySQL server
address, username, password and Kamailio (OpenSER) database name. <Database Name="Default" Driver="Pdo_Mysql" Server="xx.sipit.xx" DBName="openserx" User="zzz" Password="xxx" /> <Database Name="Siremis" Driver="Pdo_Mysql" Server="xx.sipit.xx" DBName="openserx" User="zzz" Password="xxx" />
I have a default entry and a siremis entry... Wich one has to be modified?
Both entries with same values, there are two connectors used.
Best regards, Ramona
Do I have to change the names? (even when changed, it doesn't work)
ob_usershas been created.
Kamailio version is 1.5.1 Siremis is 0.9.2
what do I wrong?
Kind regards,
Olivier (The guy who where on the Fosdem restaurant in Brussels)
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Hi Elena,
Yes, I got it, just now :)
I did disable selinux in the config file and didn't restart the service :( That's all.
Thanks anyway to Bruce and you.
Olivier
Elena-Ramona Modroiu a écrit :
Hi Olivier,
is it working now? It is about rights as Bruce already suggested.
olivier.taylor@gmail.com wrote:
[2009-06-08 14:34:52 (GMT)] An exception occurred while executing this script: Error message: #0, SQLSTATE[HY000] [2003] Can't connect to MySQL server on 'sql.sipit.be' (13) Script name and line number of error: /var/www/siremis/openbiz/others/zendfrwk/library/Zend/Db/Adapter/Pdo/Abstract.php:131
Hello Ramona,
I get this error with Siremis . [2009-06-08 14:34:52 (GMT)] An exception occurred while executing this script: Error message: #0, SQLSTATE[HY000] [2003] Can't connect to MySQL server on 'sql.sipit.be' (13) Script name and line number of error: /var/www/siremis/openbiz/others/zendfrwk/library/Zend/Db/Adapter/Pdo/Abstract.php:131
In fact, in the xml, there are 2 DB entries..
- edit file siremis/metadata/Config.xml and set MySQL server
address, username, password and Kamailio (OpenSER) database name. <Database Name="Default" Driver="Pdo_Mysql" Server="xx.sipit.xx" DBName="openserx" User="zzz" Password="xxx" /> <Database Name="Siremis" Driver="Pdo_Mysql" Server="xx.sipit.xx" DBName="openserx" User="zzz" Password="xxx" />
I have a default entry and a siremis entry... Wich one has to be modified?
Both entries with same values, there are two connectors used.
Best regards, Ramona
Do I have to change the names? (even when changed, it doesn't work)
ob_usershas been created.
Kamailio version is 1.5.1 Siremis is 0.9.2
what do I wrong?
Kind regards,
Olivier (The guy who where on the Fosdem restaurant in Brussels)
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
-
Bruce Ferrell -
Daniel-Constantin Mierla -
Elena-Ramona Modroiu -
hh174 -
Iñaki Baz Castillo -
karhu -
olivier.taylor＠gmail.com -
Raúl Alexis Betancor Santana