[OpenSIPS-Users] Add User From Opensips-Cp

Hi all, I setted up a OpenSIPS-Cp and almost everything it's working good except for User--->User Managment--> Add New. If I try to add a new User, when I press OK, nothing happen. User is not added and I can't see anything into opensips.log. I created some user by using "opensipsctl add " and it w

Re: [OpenSIPS-Users] Add User From Opensips-Cp

Hi Bogdan, thanks for reply. It was my mistake. I was creating a user, whitout compiling all fields. ( i was missing aliases ). Now it works perfectly. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Add-User-From-Opensips-Cp-tp7051225p7052539.html Sent fr

[OpenSIPS-Users] Siptraces not shown on OpenSIPS-CP

Hi folk, i am facing this issue: I set, just for learing purpose, my opensips.cfg as follow: loadmodule "siptrace.so" # - sip_trace -- modparam("siptrace", "db_url", "mysql://opensips:opensipsrw@localhost/opensips") modparam("siptrace", "trace_on", 1) modparam("siptrace", "enable_

Re: [OpenSIPS-Users] Siptraces not shown on OpenSIPS-CP

No one has idea? -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Siptraces-not-shown-on-OpenSIPS-CP-tp7052741p7054099.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list

Re: [OpenSIPS-Users] Modify To:'s field and forward

Hi List, I searched into Forum and i am reading about AVP module and UAC module but it's hard to write my first OpenSIPS's script :-( As I sayd i am trying to modify the following INVITE message: INVITE sip:00522375568@10.9.101.166:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.16.52.7:5061 From:

[OpenSIPS-Users] Add Menu to OpenSIPS-CP

Hi List, I was wondering is there is a way, and/or is possible, to add a menu to OpenSIPS-CP under System Tab. I would like to populate AVP Table by Control Panel. It would be easier then do it by opensipsctl command. Regards. -- View this message in context: http://opensips-open-sip-server.1449

Re: [OpenSIPS-Users] Modify To:'s field and forward

Hi Again, i am tryng to do, as sayd, my first script. I setted up the following: if (avp_db_load("$fu/username","$avp(Linea-LW)")) { xlog("L_DBG", "AVP_DB_LOAD Invocato!!\n"); if (avp_check("$avp(Linea-LW)","eq/y/i"));

Re: [OpenSIPS-Users] Siptraces not shown on OpenSIPS-CP

Yes, sure: $config->results_per_page = 25; $config->results_page_range = 10; // highlighting $config->from_color="black"; $config->from_bgcolor="yellow"; $config->to_color="white"; $config->to_bgcolor="blue"; $config->callid_color="black"; $config->callid_bgcolor="orange"; $con

Re: [OpenSIPS-Users] Add Menu to OpenSIPS-CP

Thanksss Alex!!! It works perfectly! -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Add-Menu-to-OpenSIPS-CP-tp7057646p7063312.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. _

Re: [OpenSIPS-Users] Siptraces not shown on OpenSIPS-CP

I done: mysql> SELECT DISTINCT callid FROM sip_trace WHERE status='' AND direction='in' ORDER BY id DESC ; Empty set (0.00 sec) mysql> mysql> -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Siptraces-not-shown-on-OpenSIPS-CP-tp7052741p7063340.html Sent

Re: [OpenSIPS-Users] Siptraces not shown on OpenSIPS-CP

Yes, sure... 3104 rows in set (0.14 sec) mysql> select * from sip_trace ; I did not pasted the 3104 rows but "sip_trace" table is full of sip traces -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Siptraces-not-shown-on-OpenSIPS-CP-tp7052741p706339

Re: [OpenSIPS-Users] Siptraces not shown on OpenSIPS-CP

Hi, i think it'l lastone: opensips -V version: opensips 1.7.1-notls (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 102

Re: [OpenSIPS-Users] Modify To:'s field and forward

Hi, i am gonna step by step closer but..it still not working. This a log from opensips.log: tail -f /var/log/opensips.log Dec 5 16:42:06 opensips /sbin/opensips[2]: DBG:tm:timer_routine: timer routine:2,tl=0xaf3edca0 next=(nil), timeout=237 Dec 5 16:42:06 opensips /sbin/opensips[2]: DB

Re: [OpenSIPS-Users] Add Menu to OpenSIPS-CP

Alex, just to be sure, Can I add more than one mysql's table at the time?? I tried to put also "address" tabel into local's file but seems it 's not working. Can you just confirm that? Regards -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Add-Menu-to-OpenS

Re: [OpenSIPS-Users] Siptraces not shown on OpenSIPS-CP

No one has idea? -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Siptraces-not-shown-on-OpenSIPS-CP-tp7052741p7066578.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list

Re: [OpenSIPS-Users] Modify To:'s field and forward

Hi Bogdan, thank you for your reply and your time. I did as you sayd and it worked in part. I had to modify this statement *if ($rU=~"^sip:0[0-9]{11}@") * with *if ($rU=~"^0[0-9]+")* I don't really understood why. I really thought that regex " *~"^sip:0[0-9]{11}@* " was correct. Anyway, thank

Re: [OpenSIPS-Users] Modify To:'s field and forward

I am trying to improve my script: Now I would like that if avp_check fails, Opensips simply rewrite host and port and keep ruri as recived. I wrote the following but seems it's wrong if (avp_db_load("$fu/username","$avp(Linea-LW)")) { xlog("L_DBG", "A

Re: [OpenSIPS-Users] Modify To:'s field and forward

Hi list, I would share with you all what i reached about my script. Hope this would be helpfull for someone of you. if (avp_db_load("$fu/username","$avp(ARS-OFS)")) { if (avp_check("$avp(ARS-OFS)","eq/lw-re/i")) { if ($rU=~"^0[0-9]+")

[OpenSIPS-Users] Help with " uac_replace_to"

Hi all, I need to rewrite a To header, after send INVITE. Here is my part of script: if (avp_db_load("$fu/username","$avp(ARS-OFS)")) { if (avp_check("$avp(ARS-OFS)","eq/lw-re/i")) { if ($rU=~"^0[0-9]+") {

Re: [OpenSIPS-Users] Help with " uac_replace_to"

Hi Vlad, thank you for your reply. I found this solution and seems a good one. Can you confirm it? if ($rU=~"^0[0-9]+") { strip(1); prefix("87019"); rewritehostport("10.9.6.3:5060");

[OpenSIPS-Users] how does OpenSIPS manage 183's message??

Hi all, I am still testing my solution to provide some additional features to own Fax server, thanks to OpenSIPS. IP-PBX<> OpenSIPS <> Fax Server I am using OpenSIPS in stateless mode ( so without record-route ) and this is the sip trace at OpenSIPS level U 2011/12/13 15:46:04.075195 172

Re: [OpenSIPS-Users] how does OpenSIPS manage 183's message??

Hi, Can someone explain/help me on this I really will appreciate Best Regards -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/how-does-OpenSIPS-manage-183-s-message-tp7090462p7097041.html Sent from the OpenSIPS - Users mailing list archive at Nabble.co

Re: [OpenSIPS-Users] how does OpenSIPS manage 183's message??

Hi Denis, I know. it's wanted. Is changed to 87019. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/how-does-OpenSIPS-manage-183-s-message-tp7090462p7097102.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. __

Re: [OpenSIPS-Users] how does OpenSIPS manage 183's message??

Hi Denis, that was it!!! It was setted to "auto" . I set it to "none" and now it works as aspected Perfet. Thank you very much for your hint ;-) Best regards -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/how-does-OpenSIPS-manage-183-s-message-tp70

Re: [OpenSIPS-Users] Siptraces not shown on OpenSIPS-CP

Can someone help me with this? I checked again config and seems ok but form CP nothing yet. Regards -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Siptraces-not-shown-on-OpenSIPS-CP-tp7052741p7097614.html Sent from the OpenSIPS - Users mailing list archiv

[OpenSIPS-Users] Query on external source ( execel or DB )

Hi folk, I need to know if with OpenSIPS i am able to have the following scenario: INITIAL SCENARIO: Incoming external call goes into "SIP PBX" , this one routes call toward another "SIP PBX". WANTED SCENARIO: Incoming external call goes into "SIP PBX,this one routes call toward OpeSIPS ( OpenS

Re: [OpenSIPS-Users] Query on external source ( execel or DB )

Thanks Vlad for your reply. Query has to be done into AS400's db. I think is not a standard db. So I thought about excel or access file ( I can export data from AS400 ). If there is not another solution, I will have a try as you suggested me. Thnaks a lot. -- View this message in context: http:/

Re: [OpenSIPS-Users] Forward NOTIFY msg,how to do that?

Hi Chris, for the moment I am non working on this case. I am working with a fax server and pbx-ip, placing opensips in the middle. Btw, the code you can see on this thread should work. Where are you stuck? -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Fo

Re: [OpenSIPS-Users] Forward NOTIFY msg,how to do that?

Yes, very similar to mine. Tell me if you will be able to achive your target. Btw, using google, there are some good blogs that they explain integration of opensips into asterisk box very well. If you need help, ask me. No problem. Have a nice day -- View this message in context: http://opensip

[OpenSIPS-Users] Add or modify "grp" table

Hi all, by default ( i thought ) there are 3 groups that i can use whit grp table: local,int and ld. I would like to use my own ( i.e: nazionali,locali and so on.. ) Is there a way to add them to database? Now, if a try command ./opensipsctl acl grant lifesize1@172.16.55.100 nazionali i get this

[OpenSIPS-Users] External SIP require AUTH

Hi all, maybe is a dummy question but I don't know how to achieve solution. I am connecting a External SIP LINE provided by a Pubblic ISP. It require authentication for each incoming call ( outgoing call by opensips point of view ). How can i solve it? Is there a module or a db table where store cr

Re: [OpenSIPS-Users] Add or modify "grp" table

Thanks Bogdan, that was the solution!! Best regards. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Add-or-modify-grp-table-tp7581378p7581428.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. __

[OpenSIPS-Users] RTPPROXY shutdown at each call

Hi all, I am implementing my first OpenSIPS with RTPPROXY so, for sure, any issue is related to my NOT-Knowledge!! :-) Issue is follow: each time I make a call ( from USERA to USERB, for instance ) RTPPROXY shuts down immediatly. Here there are logs and config: STARTING SERVER: Sep 4 16:12:20

Re: [OpenSIPS-Users] RTPPROXY shutdown at each call

Hi, can someone help me? -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/RTPPROXY-shutdown-at-each-call-tp7581647p7581676.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing

Re: [OpenSIPS-Users] RTPPROXY shutdown at each call

Hi Bogdan, NO, after made call rtpproxy is no more running!!! -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/RTPPROXY-shutdown-at-each-call-tp7581647p7581679.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com.

Re: [OpenSIPS-Users] RTPPROXY shutdown at each call

Hi Sam, i've not created a /etc/init.d/rtpproxy file!! Have I to create manually or is inside rtpproxy's installation dir? Btw, as wrote above, I start RTPP. by using this command: rtpproxy -F -s udp:127.0.0.1:7890 -l 10.9.23.41/151.x.x.201 -d DBUG:LOG_LOCAL2 and also Yes, it crash when i make a

Re: [OpenSIPS-Users] RTPPROXY shutdown at each call

Hi, I am making some test and also changing configuration at opensips.cfg, i have always same issue. The problem is related to this ERROR: ERROR:rtpproxy:send_rtpp_command: can't send command to a RTP proxy Connection refused But i cannot understand why RTPPROXY refuses connection. Any idea?

Re: [OpenSIPS-Users] RTPPROXY shutdown at each call

Hi Sam, and thanks for reply. OS is Debian 6.03 As you can see above i already posted log either from opnesips and rtpproxy. I post them again. STARTING RTPPROXY: ep 6 18:05:22 opensips rtpproxy[24996]: DBUG:handle_command: received command "25020_0 V" Sep 6 18:05:22 opensips rtpproxy[24996]:

Re: [OpenSIPS-Users] RTPPROXY shutdown at each call

Hi, I have a news regarding logs. Under /var/log/messages I found this error: Sep 7 12:52:52 opensips kernel: [584059.568944] rtpproxy[2256]: segfault at 0 ip 08051334 sp b6ce60f0 error 4 in rtpproxy[8048000+e000] Any idea about what that means?? -- View this message in context: http://open

Re: [OpenSIPS-Users] RTPPROXY shutdown at each call

I installed it by git clone git://sippy.git.sourceforge.net/gitroot/sippy/rtpproxy then ./configure and then make, make install. Sorry if i cannot understand but i'am not so skilled with opensips. I believe that code, where rtpproxy is called, is this: route[1] { # for INVITEs enable

Re: [OpenSIPS-Users] RTPPROXY shutdown at each call

Hi again and sorry for big delay but i was out of office for long time. Today I started to have some other test and always get same issue!!! I also tried with a new installation ( debian+opensips 1.8.1+new IPs+rtpproxy). As Sam suggested a tried using engage_rtpproxy but always same issue. I really

[OpenSIPS-Users] Rtpproxy connection

Hi all, I thought to open a new threat about my issue because is not ( i suppose ) related to scripting. My issue is that any time rtpproxy has to be invoked by opensips script i get the following error: *ERROR:rtpproxy:send_rtpp_command: can't send command to a RTP proxy Connection refused* and

Re: [OpenSIPS-Users] Rtpproxy connection

Hi Binan, thanks for your reply. I read links you posted me but i cannot understand how to send command to rtpproxy to test it. I tried from CLI like: *root@opensips:~# rtpproxy VF 20050322* but i get no response. Probably is not the way like it should. Can you explain me better? Thanks again.

Re: [OpenSIPS-Users] Rtpproxy connection

- No!, after opensips module sends command to rtpproxy this one shut down ( i cannot see anymore with ps command ) - in rtpproxy logs i cannot see any specific error. Below example of code that I get from rtpproxy: *Sep 27 00:31:01 opensips rtpproxy[19991]: DBUG:handle_command: received command "2

Re: [OpenSIPS-Users] Rtpproxy connection

Hi Binan, don't care about it. I made a lot of changing to figure out the issue so traces could be different to one another. Configuration of opensips and rtpproxy are always correct ( I mean the two part was always matched ). Thanks anyway :-) -- View this message in context: http://opensips

Re: [OpenSIPS-Users] Rtpproxy connection

I have some news but issue still present. I used unix socket, instead of udp, but RTPPROXY crashes at each call. Below traces of opensips side and rtpproxy side: OPENSIPS: *Sep 27 13:01:57 opensips /usr/local/opensips_proxy/sbin/opensips[26026]: ERROR:rtpproxy:send_rtpp_command: can't read reply

Re: [OpenSIPS-Users] Rtpproxy connection

I tried but i get the following error: *Sep 27 15:09:49 opensips opensips: ERROR:core:set_mod_param_regex: parameter not found in module * Seems that parameter you suggested me is not more avaible in nathelper module. I started rtpproxy like below *root 27019 1 0 15:05 ?00:00:

Re: [OpenSIPS-Users] Rtpproxy connection

I tried to use RTPPROXY_CLIENT but i get an error: *root@opensips:/tmp# java rtpproxy-client-api-0.2.jar Exception in thread "main" java.lang.NoClassDefFoundError: rtpproxy-client-api-0/2/jar Caused by: java.lang.ClassNotFoundException: rtpproxy-client-api-0.2.jar at java.net.URLClassLoade

Re: [OpenSIPS-Users] Rtpproxy connection

Yes, it is! I already tried with different users, both opensips and rtpproxy. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Rtpproxy-connection-tp7581935p7581959.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. __

Re: [OpenSIPS-Users] Rtpproxy connection

Hi Max, *"No available proxies"* and so on is because after first error ( opensips can't connect to rtpproxy and then it disable rtpproxy momentaly ). btw the port is listening: *root@opensips:/tmp# netstat -anp|grep rtpproxy udp0 0 127.0.0.1:7890 0.0.0.0:*

Re: [OpenSIPS-Users] Rtpproxy connection

Hi Binan, i checked the debian and i really think that SELinux is not implemented and is obviuosly NOT RUNNING!! Maybe i did not understand you. Should be running? -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Rtpproxy-connection-tp7581935p7581969.html

[OpenSIPS-Users] Need Hint for Opensips as UDP/TCP Proxy

Hi all, i am trying to realize a TCP/UDP gateway, using Opensips 1.8.1, to connect a IPPBX to Lync Server 2010. IPPBX<---(Trunk SIP)--->OPENSIPS<---(Trunk SIP)--->LYNC Ip pbx does not support TCP protocol and Lync does support only TCP and TLS so i need a proxy to do this job. For

Re: [OpenSIPS-Users] Need Hint for Opensips as UDP/TCP Proxy

Hi Binan, thanks for reply. I understood your idea and I am going to try but i don't know how to create a condition. let me explain: I could do something like: if (is_method(INVITE)) ... ... set_dlg_flag("3") .. but how to tell to Opensips to check a=inactive / a=sendrecv ??? if((a=inactive)

Re: [OpenSIPS-Users] Need Hint for Opensips as UDP/TCP Proxy

Any idea? -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Need-Hint-for-Opensips-as-UDP-TCP-Proxy-tp7582137p7582175.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list

Re: [OpenSIPS-Users] Need Hint for Opensips as UDP/TCP Proxy

Hi Binan, seems your idea does not work. I post ngrep traces. Look at last 200OK from IP 172.16.55.100 (opensips) to IP 10.9.101.163 (lync). SDP still have "a=inactive". Snippet of code: branch_route[2] { if (is_method("INVITE") && is_audio_on_hold()) { xlog( "L_ERR", "#

Re: [OpenSIPS-Users] Need Hint for Opensips as UDP/TCP Proxy

SOLVED!! i was not reseted flag!!! :-( This is the right code: branch_route[2] { if (is_method("INVITE") && is_audio_on_hold()) { set_dlg_flag("7"); } else { if (is_method("INVITE")){ reset_dlg_flag("

Re: [OpenSIPS-Users] Need Hint for Opensips as UDP/TCP Proxy

I talked to early !! :-( It's happening a very strange thing. Seems that this part of code is no more taken into account: *if (is_method("INVITE") || is_audio_on_hold()){ set_dlg_flag("7"); } else { if (is_method("INVITE")){

Re: [OpenSIPS-Users] Need Hint for Opensips as UDP/TCP Proxy

Hi Binan, as I said, i tried both methods but seems nothing is changing. How can I test if seted flag is really seted? How can i show it on logs?? -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Need-Hint-for-Opensips-as-UDP-TCP-Proxy-tp7582137p7582199.ht

[OpenSIPS-Users] dlg_validate_dialog Error, help

Hi, while i was making some test for my lab ( place on hold and retrive it ) i noticed on opensips's log the following error: *ERROR:dialog:dlg_validate_dialog: failed to validate remote contact: dlg=[sip:5100@172.16.52.51;transport=UDP;user=phone] , req=[sip:5100@172.16.52.51;user=phone] * I can

Re: [OpenSIPS-Users] dlg_validate_dialog Error, help

Hi Vlad, thanks for reply. I am trying to follow your hint but I get this error when I start Opensips: *Oct 11 10:29:45 opensips opensips: DBG:core:find_cmd_export_t: not found Oct 11 10:29:45 opensips opensips: DBG:core:find_cmd_export_t: found (0) in module uri [/usr/local/opensips_proxy/lib/op

Re: [OpenSIPS-Users] dlg_validate_dialog Error, help

Hi Vlad. Ok now it's a bit clear. What i am trying to do is to solve the following issue: PBX<--->Opensips<>Lync server Opensips acts as UDP/TCP proxy. Call can go from PBX to LYNC and viceversa. The problem comes when from LYNC I put on hold the call. Everytime i get the following error, ju

Re: [OpenSIPS-Users] Need Hint for Opensips as UDP/TCP Proxy

SOLVED!! It was because i have an error on dialog behaviour. For now I solved in this way, hope could be interesting for someone else: *branch_route[2] { if (is_method("INVITE") && is_audio_on_hold()){ if ( search_body("a=sendonly")){ set_dlg_

Re: [OpenSIPS-Users] dlg_validate_dialog Error, help

Any idea? -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/dlg-validate-dialog-Error-help-tp7582200p7582253.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lis

[OpenSIPS-Users] Catch 183 Session Progress

Hi all, in a trace like this, how can I catch the 183 message? I tried with this code if (t_check_status("183")) { # no support for early media xlog("LOG: Individuo il 183\n" ); } or if (status=="183") { # no support for early media xlog("#

[OpenSIPS-Users] Perl Script

Hi all, I am trying to use the perl script that is present under tutorial section of opensips web site but i have and error on opensips logs and i don't know why, here is the log: *Oct 15 15:11:14 opensips /usr/local/opensips_proxy/sbin/opensips[28780]: ERROR:core:XS_OpenSIPS__Message_log: perl er

Re: [OpenSIPS-Users] Catch 183 Session Progress

Ok Sam, solved. The t_relay was called before, in a different part of script. Now it's working as you suggested me. Thanks -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Catch-183-Session-Progress-tp7582272p7582293.html Sent from the OpenSIPS - Users mai

Re: [OpenSIPS-Users] Perl Script

Hi Brett, here is the code: use OpenSIPS qw ( log ); use OpenSIPS::Constants; ### # Create a hashref out of ab=123;bc=45 ## sub splitKeyValue { my @parts = split /\;/, shift; my $avp; my $key; my $val; while (my $part = shift(@parts)) { ($key, $val) = split /=/, $part,

Re: [OpenSIPS-Users] Perl Script

Tried but when restart Opensips I get this error: *Oct 15 15:46:31 opensips /usr/local/opensips_proxy/sbin/opensips[29672]: ERROR:core:XS_OpenSIPS__Message_log: perl error: Can't locate object method "Use" via package "IO::Socket" (perhaps you forgot to load "IO::Socket"?) at /usr/local/opensips_p

Re: [OpenSIPS-Users] Perl Script

Ok Brett!! That's was the problem. Thanks a lot -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Perl-Script-tp7582291p7582299.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Use

Re: [OpenSIPS-Users] Perl Script

Hi Brett and all, after some tests i can use the perl script but it has to be adjusted to fit my enviroment. Now i see on opensips log that the $ip is missing, infact i have this error: *Oct 15 16:48:17 opensips /usr/local/opensips_proxy/sbin/opensips[31821]: INFO:core:XS_OpenSIPS_log: Sending rep

Re: [OpenSIPS-Users] Perl Script

Is there a way to test "offline" the perl script to check what's wrong? Seems that script can't extract IP from VIA header of 183 message. Am I wrong? -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Perl-Script-tp7582291p7582311.html Sent from the OpenSIP

Re: [OpenSIPS-Users] Perl Script

No one as idea? Just in case, is there another way to convert 183 into 180? Thanks -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Perl-Script-tp7582291p7582359.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. __

Re: [OpenSIPS-Users] Perl Script

Thanks Muhammad, i will wait for your reply. Btw, I am using OpenSIPS v. 1.8.0 -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Perl-Script-tp7582291p7582362.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. __

Re: [OpenSIPS-Users] how to set $var(name) in opensips.cfg?

Hi Bogdan, I am trying to do the same thing but what if in my script I have different destination uri? Ex. if ... route(2); if route(3); ... the pv " $du " could be only one, for instance, $du = $var(new_uri); and NOT *$du1 = $var(new_uri); $du2 = $var(new_uri);*

Re: [OpenSIPS-Users] Perl Script

Hi Binan, thanks for your hint. I resolved my problem, without using perl script. Thanks to everybody. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Perl-Script-tp7582291p7582397.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. _

Re: [OpenSIPS-Users] Rtpproxy connection

Hi all, i am still stuck on this problem and I don't really know how to solve it. Is there somebody that can help me understand what's wrong? -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Rtpproxy-connection-tp7581935p7582634.html Sent from the OpenSIPS

Re: [OpenSIPS-Users] Rtpproxy connection

Hi Nick, you mean something like this? rtpproxy -F -s udp:127.0.0.1:10177 -l 10.9.23.41/151.x.x.201 -u root *-n unix:/var/run/rtpproxy_timeout.sock* -d DBUG:LOG_LOCAL2 The rtpproxy man page ( http://linux.die.net/man/8/rtpproxy ) says following: *-n timeout_socket This parameter configures the o

Re: [OpenSIPS-Users] Rtpproxy connection

Hi Nick, you SOLVED my issue!!! Thank you very much. So, seems rtpproxy's bug is not resolved yet!! Hope this long treat can help someone else. regards -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Rtpproxy-connection-tp7581935p7582656.html Se

Re: [OpenSIPS-Users] Help to Understand Loop

Him nobody has an idea? I searched for similar issue in the forum and I found something similar in this treat: http://opensips-open-sip-server.1449251.n2.nabble.com/loose-route-loop-on-ACK-requests-td2462835.html I checked but in the domain table there is only opensips ip. Thanks -- View this

Re: [OpenSIPS-Users] Help to Understand Loop

I found in the last 200 OK, before ACK, that contact is like this: *Contact: sip:10.9.6.3* but, comparing whit another call, that should be something like: *Contact: * Should be this the issue? For some reason Opensips does not like that format and inserts into ACK itself instead of the content

Re: [OpenSIPS-Users] Help to Understand Loop

Hi, I found this ERRORS in the log, but i cannot understand why. What Am i missing? ov 2 14:30:21 opensips /usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]: DBG:core:parse_msg: SIP Reply (status): Nov 2 14:30:21 opensips /usr/local/opensips_proxy_1.8.1/sbin/opensips[15969]: DBG:core:parse_m

Re: [OpenSIPS-Users] Help to Understand Loop

Hi, can someone help me understand this issue? Thanks -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Help-to-Understand-Loop-tp7582655p7582785.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. _

[OpenSIPS-Users] Help: Understanding ACK loop

Hi, i had to rewrite post because the preview one was too big. Sorry for this mistake!! Here is the new, short, one: Hi, thanks for reply. Here the SIP dump and after it, configuration. IP 172.16.55.100 = Opensips Gw IP 10.9.6.3 = PBX IP 10.9.6.40 = Gateway GSM The problem is when i receiv

Re: [OpenSIPS-Users] Help: Understanding ACK loop

Hi Qasim, i tried but same issue. SIP trace is changed a bit,like below, but issue not. i also get this errors into opensips logs but i cannot uderstand them. U 2012/11/06 20:33:25.046107 10.9.6.3:5060 -> 172.16.55.100:5060 SIP/2.0 200 OK. Record-Route: . Record-Route: . Allow: INVITE, ACK, CANCE

Re: [OpenSIPS-Users] Help: Understanding ACK loop

Hi Bogdan, thanks for your time. Log, in debug 6, is attached as file. Regards Opnesips_log.txt -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Help-Understanding-A

Re: [OpenSIPS-Users] Help: Understanding ACK loop

SOLVED!!! Thanks Bogdan for your hint ;-) That was the problem. PBX was sending a Contact header NOT compliant to RFC!!! Just to help someone else, I added this snippet of code: *if ((status=="200") && (src_ip == IP_PBX)) { replace("Contact: sip:x.x.x.x.","Contact: .");

Re: [OpenSIPS-Users] textops and Variable

Hi, i really suppose you should use REGEX. As per manual: 1.3.8. replace_body_all(re, txt) Replaces all occurrence of re in the body of the message with txt. Matching is done on a per-line basis. Meaning of the parameters is as follows: *re - Regular expression.* txt - String. Regards -

Re: [OpenSIPS-Users] RTPProxy Problem

Hi, take a look at my post. Probably you will find a solution for your issue. http://opensips-open-sip-server.1449251.n2.nabble.com/Rtpproxy-connection-td7581935.html -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/RTPProxy-Problem-tp7582930p7582943.html

Re: [OpenSIPS-Users] RTPProxy Problem

Please, post entire opensips log ( set debug to 6 ). Why are you using same port for sock and notify_sock ??? Have you tried with different ports? Bye -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/RTPProxy-Problem-tp7582930p7582974.html Sent from the O

[OpenSIPS-Users] Question about Parallel Forking

Hi all, i need to do parallel forking towards 2 different systems. When one of them had picked up the call, opensips has to be able to knows that until call is terminated. What i need is, if a second call comes again, opensips has to reply to the caller the busy tone. Is this possible? SCENARIO:

Re: [OpenSIPS-Users] Question about Parallel Forking

Any Idea?? -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Question-about-Parallel-Forking-tp7583079p7583111.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@l

[OpenSIPS-Users] Help with "t_relay"

Hi list, i need some hints by you experts to get working my idea. What I am trying to do is parallel forking toward 2 different IP PBXs whose have different carateristics. Let me explain: IP PBX #1: accept UDP connections on port 5060 IP PBX #2: accept TCP connections on port 5068 What i tried t

Re: [OpenSIPS-Users] Help with "t_relay"

Hi Bogdan, you said: $ru = "sip:user@domain"; # set initial URI But the initial URI, in my case, could be different, because is an incoming call from PSTN towards several DDI users. I can't define ($ru) as i unique URI. Can I set is as a variable? Am I wrong? Thanks -- View this message in

Re: [OpenSIPS-Users] Help with "t_relay"

Hi all, solved with following solution: if (src_ip == IP_MEDIANT || (method=="INVITE")) { rewritehostport("FQDN_IPPBX1:PORT_IPPBX1"); route(10); exit; } .. ..

Re: [OpenSIPS-Users] check_source_address not working with upgrade

Hi all, i would arise this post because it's happening same thing to me. Opensips v. : OpenSIPS (1.9.0-notls (i386/linux)) Database entries: mysql> select * from address; ++-+---+--+--+---+-+--+ | id | grp | ip| mask | port | proto

[OpenSIPS-Users] Modify FROM header with REG EXP

Hi all, I am trying to do a weird thing regarding modify FROM header of INVITE. Let me explain: This is a snippet of INVITE that i am trying to modify: *FROM: "Name Surname";epid=6BEB47B1B9;tag=898892e6e7.* What I want to achieve is: FROM: "Name Surname"532542*@domain.local;user=phone>;epid=6BE

[OpenSIPS-Users] check_source_address not working in 1.9

Hi all, i am facing issue on using "check_source_address" Opensips v. : OpenSIPS (1.9.0-notls (i386/linux)) Database entries: mysql> select * from address; ++-+---+--+--+---+-+--+ | id | grp | ip| mask | port | proto | pattern

Re: [OpenSIPS-Users] check_source_address not working in 1.9

Hi Bogdan, here snippet of used code: *if(!check_source_address("0")){ xlog("LOG: Controllo dell'IP sorgente!\n" ); if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0");

[OpenSIPS-Users] Inbound call not works

Hi all, I am tryng to understand how OpenSIPS works and i am playng with OpenSIPS liveCD. I am able to register to it user and make outbound call toward asterisk machine ( is a separated server with different IP ). The problem is when an asterisk user try to call a OpenSIPS user. I always have t

[OpenSIPS-Users] Forward NOTIFY msg,how to do that?

Hi all, I am pretty new on OpenSIPS world so first of all sorry for dummy questions. My configuration is listed below: IP-PBX<>Asterisk<>ExchangeUM 2010 Ip-PBX use Exchange voice mail. Asterisk acts as Proxy SIP because IP-PBX is not able to modify SIP HEADER, needed for integration wi

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