- sr-users - kamailio.org

[Serusers] How to set the query into database and get answers with exec_msg?
by 邱磊 26 Feb '09

26 Feb '09

Hi ,I am a student studying in VoIP and implementing my theory in SER. I try to access mysql database to get some information, but have a problem aboout paramaters of exe_msg. My goal is :count the number of contact addresses that certain user registered, and take this is my solution: exec_msg(' Count="select count(contact) from location where username=\"$SIP_HF_CONTACT\" "' CountResult=`mysql -user -pheslo -e "$Count" ser`; ') What is the result of "CountResult" ? I suppose originally that it will be the the number of the query, but it seems not......... and i try the exec_msg('rm -r test'); but it failed; Dose the function wored? and how to use it,Hhw can I get my answer what I want to get ? Hoping someone can give me some advises ^^. Thannks in advance. 2009-02-26 邱磊

1 0

[Kamailio-Users] Handling CANCEL and 487
by Juan Perez 25 Feb '09

25 Feb '09

Hello I am having a prob handling the CANCEL in this scenario Kamailio 1.4 doing basic load balancing with LCR module. Calling party makes a call, sends it to Kamailio, Kamailio looks up the destination gw and sends the call to the dest IP. I am using statefull mode and loose routing. The calling party makes the call and while it is ringing it hangs up. Kamailio receives the CANCEL and forward it to the dest IP, it receives the 487 from dest IP and pass it back to the calling party. This is when it gets weird. The calling party sends the ACK to the 487 and it looks like the proxy [kamailio] ignores it and keep sending the 487 to the calling party. Calling party IP: 192.168.1.200 kamailio[proxy]: 192.168.1.50 dest IP: 192.168.11 PLease see below a tethereal output and my config file. I would appreciate any help to help me fix this prob. [root@proxy1 ~]# tethereal -i eth1 -S -w /root/call-with-cancel3.pcap udp port 5060 Running as user "root" and group "root". This could be dangerous. Capturing on eth1 0.000000 192.168.1.200 -> 192.168.1.50 SIP/SDP Request: INVITE sip:664050495179280@192.168.1.50, with session description 0.000702 192.168.1.50 -> 192.168.1.200 SIP Status: 100 Giving a try 0.000774 192.168.1.50 -> 192.168.1.11 SIP/SDP Request: INVITE sip:664050495179280@192.168.1.11:5060;transport=udp, with session description 0.044219 192.168.1.11 -> 192.168.1.50 SIP Status: 100 Trying 0.731527 192.168.1.11 -> 192.168.1.50 SIP/SDP Status: 183 Session Progress, with session description 0.731585 192.168.1.50 -> 192.168.1.200 SIP/SDP Status: 183 Session Progress, with session description 1.869161 192.168.1.200 -> 192.168.1.50 SIP Request: CANCEL sip:664050495179280@192.168.1.50 1.869626 192.168.1.50 -> 192.168.1.200 SIP Status: 200 canceling 1.869738 192.168.1.50 -> 192.168.1.11 SIP Request: CANCEL sip:664050495179280@192.168.1.11:5060;transport=udp 1.869942 192.168.1.11 -> 192.168.1.50 SIP Status: 487 Request Terminated 1.869946 192.168.1.11 -> 192.168.1.50 SIP Status: 200 OK 1.870243 192.168.1.50 -> 192.168.1.11 SIP Request: ACK sip:664050495179280@192.168.1.11:5060;transport=udp 1.870273 192.168.1.50 -> 192.168.1.200 SIP Status: 487 Request Terminated 1.871112 192.168.1.200 -> 192.168.1.50 SIP Request: ACK sip:664050495179280@192.168.1.50 2.317041 192.168.1.50 -> 192.168.1.200 SIP Status: 487 Request Terminated 3.317025 192.168.1.50 -> 192.168.1.200 SIP Status: 487 Request Terminated 5.317028 192.168.1.50 -> 192.168.1.200 SIP Status: 487 Request Terminated 9.317036 192.168.1.50 -> 192.168.1.200 SIP Status: 487 Request Terminated 13.317043 192.168.1.50 -> 192.168.1.200 SIP Status: 487 Request Terminated 17.317049 192.168.1.50 -> 192.168.1.200 SIP Status: 487 Request Terminated 21.317057 192.168.1.50 -> 192.168.1.200 SIP Status: 487 Request Terminated 25.317064 192.168.1.50 -> 192.168.1.200 SIP Status: 487 Request Terminated 29.317071 192.168.1.50 -> 192.168.1.200 SIP Status: 487 Request Terminated 23 packets captured This is my routing logic: ####### Routing Logic ######## # main routing logic route{ xlog("L_INFO", "mylog: starting_main_logic.\n"); if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); xlog("L_INFO","mylog: Too Many Hops.\n"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); xlog("L_INFO","mylog: Message too big.\n"); exit; }; if (!method=="REGISTER") { record_route(); xlog("L_INFO","mylog: Recording Route. Method: [$rm].\n"); } if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); xlog("L_INFO","mylog: Loose Route section. Method: [$rm].\n"); route(1); }; if (method=="INVITE") { if (!load_gws()) { sl_send_reply("503", "Unable to load gateways"); exit; } if(!next_gw()){ sl_send_reply("503", "Unable to find a gateway"); exit; } route(1); } #CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } } route[1] { xlog("L_INFO","mylog: Route 1 section. Method [$rm]."); if (!t_relay()) { xlog("L_INFO","mylog: Route 1 section. T_Relay failed. Method [$rm]."); sl_reply_error(); }; exit; }

2 1

Re: [Kamailio-Users] Need to handle an upstream carrier error - Duplicate INVITE Messages
by Geoffrey Mina 25 Feb '09

25 Feb '09

Hello, I have a carrier who provides PSTN gateway services. They have multiple redundant sip gateway devices in their network. The problem occurs when one of their devices starts to have issues. I will receive an INVITE request from both gateways with the same call-id. The problem is that my Kamailio system doesn't detect that I already set a call up for the INVITE once, and forwards the request to another server in the dispatcher list. What I end up with is a call on two asterisk servers, but only one has the actual RTP stream. The BYE request gets routed to the wrong server, and everything just gets screwy. If anyone could provide any hint on how I might be able to deal with this scenario, I would really appreciate it. I have attached my current config file, and the following is a link to a google spreadsheet which shows the SIP trace. http://spreadsheets.google.com/ccc?key=pU5i2J6Ck3b519-_M6Et3cw I have masked my IP addresses for my own sanity. XX.XX.XX.179 - Kamailio SIP Gateway XX.XX.XX.189 - Asterisk1 XX.XX.XX.186 - Asterisk2 Thanks! Geoff

2 5

[Kamailio-Users] openser not distributing to all DNS SRV entries
by Brian 25 Feb '09

25 Feb '09

Hi, I have a SRV setup with three equal priority entries and equal weights. I expected openser (version 1.3.3) to distribute the INVITE requests equally to all 3 hosts, but it seems that all requests are forwarded to only one host. The SRV entries look like this: _sip._udp.mydomain.com. 8805 IN SRV 100 33 5060 proxy1.mydomain.com. _sip._udp.mydomain.com. 8805 IN SRV 100 34 5060 proxy1.mydomain.com. _sip._udp.mydomain.com. 8805 IN SRV 100 33 5060 proxy1.mydomain.com. And my routing for INVITE request looks like this: rewritehost("mydomain.com"); rewriteport(""); if (!t_relay()) { sl_reply_error(); }; exit; Am I wrong in expecting this to distribute equally to the three proxy hosts? I thought this would just "work". Thanks, Brian.

3 3

Re: [Kamailio-Users] event in kamailio
by Marion.DELEY＠fr.thalesgroup.com 25 Feb '09

25 Feb '09

Hi, Finally, I've used the send() function in kamailio.cfg with the @ of my UDP server to know when invite and register requests arrive. Thanks for your help. marion -----Message d'origine----- De : Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Envoyé : mardi 24 février 2009 21:32 À : Marion.DELEY(a)fr.thalesgroup.com Cc : users(a)lists.kamailio.org Objet : Re: [Kamailio-Users] event in kamailio Marion.DELEY(a)fr.thalesgroup.com wrote: > Hi everybody, > > I'm novice in using Kamailio and I need to know if it's possible to create > event from kamailio.cfg to another program. > Actually, I use MI command but I would find a way to force kamailio to send > me messages and not just responses. If the other programm supports SIP or HTTP you can send SIP requests from kamailio.cfg and generate HTTP requests. regards klaus

2 1

[Kamailio-Users] event in kamailio
by Marion.DELEY＠fr.thalesgroup.com 24 Feb '09

24 Feb '09

Hi everybody, I'm novice in using Kamailio and I need to know if it's possible to create event from kamailio.cfg to another program. Actually, I use MI command but I would find a way to force kamailio to send me messages and not just responses. Some help? thanks marion

3 2

[Kamailio-Users] Error compiling CARRIERROUTE module
by Asim Riaz 24 Feb '09

24 Feb '09

Hello, I am trying to compile CARRIERROUTE module, I am getting errors below; Kamailio 1.4 make prefix=/usr modules=modules/carrierroute modules make[1]: Entering directory `/usr/local/src/kamailio/sip-server/modules/carrierroute' Compiling carrierroute.c gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott -Wold-style-definition -Wmissing-field-initializers -Wredundant-decls -DMOD_NAME='"carrierroute"' -DNAME='"kamailio"' -DVERSION='"1.4.3-notls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.3.2"' -D__CPU_i386 -D__OS_linux -D__SMP_yes -DCFG_DIR='"/usr/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DF_MALLOC -DSVNREVISION='"2:5452M"' -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -I/usr/local/include -c carrierroute.c -o carrierroute.o In file included from carrierroute.c:30: ../../sr_module.h:48:19: error: dlfcn.h: No such file or directory In file included from /usr/lib/gcc/i486-linux-gnu/4.3.2/include-fixed/syslimits.h:7, from /usr/lib/gcc/i486-linux-gnu/4.3.2/include-fixed/limits.h:11, from ../../parser/../flags.h:32, from ../../parser/msg_parser.h:46, from ../../sr_module.h:50, from carrierroute.c:30: /usr/lib/gcc/i486-linux-gnu/4.3.2/include-fixed/limits.h:122:61: error: limits.h: No such file or directory In file included from ../../parser/msg_parser.h:47, from ../../sr_module.h:50, from carrierroute.c:30: ../../parser/../ip_addr.h:38:20: error: string.h: No such file or directory ../../parser/../ip_addr.h:39:23: error: sys/types.h: No such file or directory ../../parser/../ip_addr.h:40:24: error: sys/socket.h: No such file or directory ../../parser/../ip_addr.h:41:24: error: netinet/in.h: No such file or directory ../../parser/../ip_addr.h:42:19: error: netdb.h: No such file or directory In file included from ../../parser/../ip_addr.h:46, from ../../parser/msg_parser.h:47, from ../../sr_module.h:50, from carrierroute.c:30: ../../parser/../dprint.h:54:20: error: syslog.h: No such file or directory ../../parser/../dprint.h:55:18: error: time.h: No such file or directory In file included from ../../parser/../ip_addr.h:46, from ../../parser/msg_parser.h:47, from ../../sr_module.h:50, from carrierroute.c:30: any idea ? Thanks in advance. Asim

1 1

Re: [Kamailio-Users] help me in ldap kamailio
by Henning Westerholt 24 Feb '09

24 Feb '09

On Tuesday 24 February 2009, Husna Bte Mad Baguri wrote: > There is no error msg.just there is no file name ldap.cfg in > usr/local/etc/kamailio.lets me explain how i install. > > 1.i using ubuntu 8.10 > 2.i use kamailio 1.4.3 with tls > 3.i install open ldap --> libldap2-dev,ldap-utils,slapd > 4.then i exclude module =ldap Hi Husna, did i understand you correctly, you excluded the ldap module from compilation? This way it will not be compiled. > 5.then i install.no error but no file ldap.cfg > 6.can you explain what next. It seems that the ldap.cfg file is not installed per default. You find this file in modules/ldap/etc/ in the kamailio 1.4.3 directory. Just copy it to the location you want, and probably adapt it to your setup. Cheers, Henning

1 0

[Kamailio-Users] How can I use domain module?
by Ramu 24 Feb '09

24 Feb '09

Hi, I would like to use domain module to support multiple domain users. How can move forward on this? Thanks, Ramu

5 8

[Kamailio-Users] help me in ldap kamailio
by Husna Bte Mad Baguri 24 Feb '09

24 Feb '09

Can someone help me,i counter this error when try to install ldap in kamailio Compiling ldap_api_fn.c gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott -Wold-style-definition -Wmissing-field-initializers -Wredundant-decls -DMOD_NAME='"ldap"' -DNAME='"kamailio"' -DVERSION='"1.4.3-notls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.3.2"' -D__CPU_i386 -D__OS_linux -D__SMP_yes -DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -I/usr/local/include -c ldap_api_fn.c -o ldap_api_fn.o ldap_api_fn.c:38:18: error: ldap.h: No such file or directory In file included from ldap_api_fn.c:40: Plzzzzz help me regards Husna This e-mail and the attachments are intended solely for the person to whom it has been addressed. It contains privileged and/or confidential information and is privileged or otherwise protected from disclosure. If you are not the person for whom this e-mail was intended, or the e-mail has reached you by mistake, please delete it immediately and inform us of the error. Our e-mail address is administrator(a)uniten.edu.my. All opinions, conclusions and other information in this message that do not relate to the official business of Universiti Tenaga Nasional (UNITEN) shall be understood as neither given nor endorsed by UNITEN. UNITEN shall not be responsible for any activity that may be considered as illegal and/or improper use of e-mail and UNITEN further is claims and shall not accept liability for any content of this e-mail, or for the consequences of any actions taken on the basis of the information provided, unless that information is subsequently confirmed in writing. WARNING Internet communications cannot be guaranteed to be secured or error-free as information could be intercepted, corrupted, lost, arrive late or contain viruses. As such, we do not accept liability for any errors or omissions in the content of this message which may arise as a result of internet transmission. UNITEN does not authorize any of its employees to make any defamatory or seditious statements or commit any offence which is contrary to the laws of Malaysia. Any such communications and/or actions by such employees are outside the scope of employment of the said individuals and UNITEN shall not be liable for such communications and/or actions.

2 1

[Kamailio-Users] help me in ldap kamailio
by Husna Bte Mad Baguri 24 Feb '09

24 Feb '09

Can someone help me,i counter this error when try to install ldap in kamailio Compiling ldap_api_fn.c gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott -Wold-style-definition -Wmissing-field-initializers -Wredundant-decls -DMOD_NAME='"ldap"' -DNAME='"kamailio"' -DVERSION='"1.4.3-notls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.3.2"' -D__CPU_i386 -D__OS_linux -D__SMP_yes -DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -I/usr/local/include -c ldap_api_fn.c -o ldap_api_fn.o ldap_api_fn.c:38:18: error: ldap.h: No such file or directory In file included from ldap_api_fn.c:40: Plzzzzz help me regards Husna This e-mail and the attachments are intended solely for the person to whom it has been addressed. It contains privileged and/or confidential information and is privileged or otherwise protected from disclosure. If you are not the person for whom this e-mail was intended, or the e-mail has reached you by mistake, please delete it immediately and inform us of the error. Our e-mail address is administrator(a)uniten.edu.my. All opinions, conclusions and other information in this message that do not relate to the official business of Universiti Tenaga Nasional (UNITEN) shall be understood as neither given nor endorsed by UNITEN. UNITEN shall not be responsible for any activity that may be considered as illegal and/or improper use of e-mail and UNITEN further is claims and shall not accept liability for any content of this e-mail, or for the consequences of any actions taken on the basis of the information provided, unless that information is subsequently confirmed in writing. WARNING Internet communications cannot be guaranteed to be secured or error-free as information could be intercepted, corrupted, lost, arrive late or contain viruses. As such, we do not accept liability for any errors or omissions in the content of this message which may arise as a result of internet transmission. UNITEN does not authorize any of its employees to make any defamatory or seditious statements or commit any offence which is contrary to the laws of Malaysia. Any such communications and/or actions by such employees are outside the scope of employment of the said individuals and UNITEN shall not be liable for such communications and/or actions.

1 1

[Serusers] Cleaning bad npdi=yes and rn=VALUE tags from INVITEs
by Frank Durda IV 24 Feb '09

24 Feb '09

I have a client who occasionally sends calls where there is a npdi=yes tag (and there may or may not be a rn=VALUE tag present with it), BUT the answer these two tags are providing is wrong! That is, the number in question has or hasn't been ported, but the information provided in the INVITE message doesn't match reality. Either a rn=VALUE should have had value X and it instead has value Y, OR there is no rn=VALUE at all, and there should have been one, OR the number hasn't actually been ported, and no rn=VALUE should have been sent at all but we got a junk one, that probably isn't the right LRN for the true destination of the call. Because of these wrong npdi=yes tags, our PSTN gateway switch ends up failing these incorrectly-marked call, because it takes the provided rn=VALUE or lack of one to be correct and so it doesn't do its own LNP query, and the one provided in the SIP INVITE is sometimes just flat out wrong. If the PSTN gateway switch had ignored these suspect npdi=yes and rn=VALUE tages and simply done the LNP query itself, in virtually all cases these otherwise doomed calls would have completed successfully. For this particular customer, the problem of bogus values appears to be a permanent artifact. They either don't know how to fix it, or don't think it is important, of they are spoofing it to avoid being billed for LNP queries. Whatever the reason, it causes trouble tickets. So, we would like to have SER strip the npdi=yes and any rn=VALUE tags from the INVITE when it is this customer (known by src ip), so that the PSTN switch will do the LNP query itself on the SS7 side and get the call routing right. (We have already determined that getting the outfit that is sending the junk to do the right thing isn't going to happen any time soon.) Does anybody have any suggestions as how to write the ser.cfg code to edit-out a npdi=yes and rn=VALUE if they are present in INVITEs? We just want to always throw npdi=yes and any rn=VALUE that shows up if the call is coming from this given customer, basically treating all npdi=yes and rn= tags coming from this customer as being untrustworthy. There has to be an easier way than stripping the thing down and reassembling a new one with repeated strip(1) type operations, which looks to be extremely inefficient for this task and probably really meant only for stripping country codes and such off the phone numbers, not for cleaning up bad tags. Thanks, in advance!

1 0

Re: [Kamailio-Users] LCR in Kamailio 1.4
by Ovidiu Sas 23 Feb '09

23 Feb '09

Make sure that from_uri is NULL and not a white space. Please copy the mailing list in your reply. Regards, Ovidiu Sas On Thu, Feb 19, 2009 at 11:25 AM, Asim Riaz <ariaz78(a)gmail.com> wrote: > > > On Thu, Feb 19, 2009 at 4:16 PM, Ovidiu Sas <osas(a)voipembedded.com> wrote: >> >> Is your R-URI starting with a '0'? >> You have defined a single route for all '0' prefix destinations. > > yes R-URI is starting with 0. >> >> >> Regards, >> Ovidiu Sas >> >> On Thu, Feb 19, 2009 at 11:06 AM, Asim Riaz <ariaz78(a)gmail.com> wrote: >> > >> > >> > On Thu, Feb 19, 2009 at 3:53 PM, Daniel-Constantin Mierla >> > <miconda(a)gmail.com> wrote: >> >> >> >> Hello, >> >> >> >> On 02/19/2009 05:34 PM, Asim Riaz wrote: >> >>> >> >>> Hello, >> >>> I am using LCR, for some reason load_gws() function is not finding the >> >>> gateway in DB, it was working in the start then added some more routes >> >>> and >> >>> after that everything stopped working, I am not sure where to start >> >>> troubleshooting, any idea ? >> >> >> >> run kamailio with higher debug mode (e.g., debug=5) in configuration >> >> file. >> >> >> >> You should get more debug messages in the syslog, that may give you >> >> some >> >> hints. look for lines containing 'lcr'. >> >> >> >> Cheers, >> >> Daniel >> > >> > Hi Daniel, >> > I tried higher debug level, not getting anything related to LCR all i >> > am >> > getting XLOG messages "unable to find Gateway". >> > >> > Thanks, >> > >> > Asim >> > >> >> >> >>> >> >>> kamctl lcr show >> >>> >> >>> lcr routes >> >>> +----+--------+----------+--------+----------+ >> >>> | id | prefix | from_uri | grp_id | priority | >> >>> +----+--------+----------+--------+----------+ >> >>> | 9 | 0 | | 1 | 1 | >> >>> +----+--------+----------+--------+----------+ >> >>> lcr gateways >> >>> >> >>> >> >>> +------------+---------------+------+------------+-----------+--------+-------+------+-------+ >> >>> | gw_name | ip_addr | port | uri_scheme | transport | grp_id >> >>> | >> >>> strip | tag | flags | >> >>> >> >>> >> >>> +------------+---------------+------+------------+-----------+--------+-------+------+-------+ >> >>> | XXXX | XXXX | 5060 | 1 | 1 | 1 | >> >>> 0 >> >>> | | 0 | >> >>> >> >>> >> >>> +------------+---------------+------+------------+-----------+--------+-------+------+-------+ >> >>> >> >>> >> >>> >> >>> my routing script >> >>> >> >>> if(!load_gws()) { >> >>> xlog("L_INFO", "[ROUTE-X] Unable To Find Gateway "); >> >>> sl_send_reply("503", "unable to find gateway"); >> >>> exit; >> >>> } >> >>> Thanks in advance. >> >>> >> >>> Asim >> >>> >> >>> ------------------------------------------------------------------------ >> >>> >> >>> _______________________________________________ >> >>> Kamailio (OpenSER) - Users mailing list >> >>> Users(a)lists.kamailio.org >> >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >> >> >> >> -- >> >> Daniel-Constantin Mierla >> >> http://www.asipto.com >> >> >> > >> > >> > _______________________________________________ >> > Kamailio (OpenSER) - Users mailing list >> > Users(a)lists.kamailio.org >> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >> > > >

4 15

[Kamailio-Users] problem with 200ok
by BERGANZ François 23 Feb '09

23 Feb '09

Hello, I have Asterisk1---SER---Asterisk2. When I do INVITE from the left, --the asterisk2 send 200ok to the SER --the SER forward to the Asterisk1 --but the asterisk1 directly send the ACK to Asterisk2 Asterisk2 retransmit the 200ok… and error. I think that it need that the ACK come from the SER and not directly from the Asterisk1. So, how can I detect a 200ok and reply a ACK with my SER? Or, anyone have ever seen that problem? Thank you Next: my capture problem SIP/2.0 183 Session Progress Via: SIP/2.0/UDP IP_SER;branch=z9hG4bK815c.d8703d97.0;received=IP_SER Via: SIP/2.0/UDP IP_PROVIDER_ASTERISK:5060;branch=z9hG4bK369cf0be;rport=5060 From: "francois berganz" <sip:170725014@IP_PROVIDER_ASTERISK>;tag=as4a6f8fc6 To: <sip:URI_TEST@IP_SER>;tag=as272dc8d3 Call-ID: 5079515a2060a26155be2cbf6b73dc78@IP_PROVIDER_ASTERISK CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:URI_TEST@IP_ASTERISK> Content-Type: application/sdp Content-Length: 287 v=0 o=root 1097653753 1097653753 IN IP4 IP_ASTERISK s=Asterisk PBX 1.6.0.1 c=IN IP4 IP_ASTERISK t=0 0 m=audio 16386 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 6§¢Iœÿ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP IP_PROVIDER_ASTERISK:5060;branch=z9hG4bK369cf0be;rport=5060 From: "francois berganz" <sip:170725014@IP_PROVIDER_ASTERISK>;tag=as4a6f8fc6 To: <sip:URI_TEST@IP_SER>;tag=as272dc8d3 Call-ID: 5079515a2060a26155be2cbf6b73dc78@IP_PROVIDER_ASTERISK CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:URI_TEST@IP_ASTERISK> Content-Type: application/sdp Content-Length: 287 v=0 o=root 1097653753 1097653753 IN IP4 IP_ASTERISK s=Asterisk PBX 1.6.0.1 c=IN IP4 IP_ASTERISK t=0 0 m=audio 16386 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv SIP/2.0 200 OK v: SIP/2.0/UDP IP_SER_TO_CLIENTS;branch=z9hG4bK6c33.6ae23971.0 v: SIP/2.0/UDP IP_ASTERISK:5060;branch=z9hG4bK598106c4;rport=5060 f: "francois berganz"<sip:170725014@IP_ASTERISK>;tag=as1551f6d3 t: <sip:URI_TEST@IP_SER_TO_CLIENTS>;tag=c0a80101-41e5ef i: 1593ec3c0a617037319349245be7c5ab@IP_ASTERISK CSeq: 102 INVITE Require: timer x: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,I NFO m: <sip:URI_TEST@192.168.1.82:5060;user=phone> Record-Route: <sip:IP_SER_TO_CLIENTS;lr=on;ftag=as1551f6d3> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold c: application/sdp l: 206 v=0 o=URI_TEST 4319348 4319348 IN IP4 192.168.1.82 s=- c=IN IP4 192.168.1.82 t=0 0 m=audio 41000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv SIP/2.0 200 OK v: SIP/2.0/UDP IP_ASTERISK:5060;branch=z9hG4bK598106c4;rport=5060 f: "francois berganz"<sip:170725014@IP_ASTERISK>;tag=as1551f6d3 t: <sip:URI_TEST@IP_SER_TO_CLIENTS>;tag=c0a80101-41e5ef i: 1593ec3c0a617037319349245be7c5ab@IP_ASTERISK CSeq: 102 INVITE Require: timer x: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,I NFO m: <sip:URI_TEST@IP_PHONE:5060;user=phone> Record-Route: <sip:IP_SER_TO_CLIENTS;lr=on;ftag=as1551f6d3> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold c: application/sdp l: 206 v=0 o=URI_TEST 4319348 4319348 IN IP4 192.168.1.82 s=- c=IN IP4 192.168.1.82 t=0 0 m=audio 41000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv ACK sip:URI_TEST@IP_PHONE:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP IP_ASTERISK:5060;branch=z9hG4bK0e97e7e8;rport Route: <sip:IP_SER_TO_CLIENTS;lr=on;ftag=as1551f6d3> Max-Forwards: 70 From: "francois berganz" <sip:170725014@IP_ASTERISK>;tag=as1551f6d3 To: <sip:URI_TEST@IP_SER_TO_CLIENTS>;tag=c0a80101-41e5ef Contact: <sip:170725014@IP_ASTERISK> Call-ID: 1593ec3c0a617037319349245be7c5ab@IP_ASTERISK CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0.1 Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP IP_SER;branch=z9hG4bK815c.d8703d97.0;received=IP_SER Via: SIP/2.0/UDP IP_PROVIDER_ASTERISK:5060;branch=z9hG4bK369cf0be;rport=5060 From: "francois berganz" <sip:170725014@IP_PROVIDER_ASTERISK>;tag=as4a6f8fc6 To: <sip:URI_TEST@IP_SER>;tag=as272dc8d3 Call-ID: 5079515a2060a26155be2cbf6b73dc78@IP_PROVIDER_ASTERISK CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:URI_TEST@IP_ASTERISK> Content-Type: application/sdp Content-Length: 287 v=0 o=root 1097653753 1097653754 IN IP4 IP_ASTERISK s=Asterisk PBX 1.6.0.1 c=IN IP4 IP_ASTERISK t=0 0 m=audio 16386 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv SIP/2.0 200 OK Via: SIP/2.0/UDP IP_PROVIDER_ASTERISK:5060;branch=z9hG4bK369cf0be;rport=5060 From: "francois berganz" <sip:170725014@IP_PROVIDER_ASTERISK>;tag=as4a6f8fc6 To: <sip:URI_TEST@IP_SER>;tag=as272dc8d3 Call-ID: 5079515a2060a26155be2cbf6b73dc78@IP_PROVIDER_ASTERISK CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:URI_TEST@IP_ASTERISK> Content-Type: application/sdp Content-Length: 287 v=0 o=root 1097653753 1097653754 IN IP4 IP_ASTERISK s=Asterisk PBX 1.6.0.1 c=IN IP4 IP_ASTERISK t=0 0 m=audio 16386 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ACK sip:URI_TEST@IP_PHONE:5060;user=phone SIP/2.0 Record-Route: <sip:IP_SER_TO_CLIENTS;lr=on;ftag=as1551f6d3> Via: SIP/2.0/UDP IP_SER_TO_CLIENTS;branch=z9hG4bK6c33.6ae23971.2 Via: SIP/2.0/UDP IP_ASTERISK:5060;branch=z9hG4bK0e97e7e8;rport=5060 Max-Forwards: 70 From: "francois berganz" <sip:170725014@IP_ASTERISK>;tag=as1551f6d3 To: <sip:URI_TEST@IP_SER_TO_CLIENTS>;tag=c0a80101-41e5ef Contact: <sip:170725014@IP_ASTERISK> Call-ID: 1593ec3c0a617037319349245be7c5ab@IP_ASTERISK CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0.1 Content-Length: 0 ACK sip:URI_TEST@IP_ASTERISK SIP/2.0 Via: SIP/2.0/UDP IP_PROVIDER_ASTERISK:5060;branch=z9hG4bK7a6d7cf4;rport From: "francois berganz" <sip:170725014@IP_PROVIDER_ASTERISK>;tag=as4a6f8fc6 To: <sip:URI_TEST@IP_SER>;tag=as272dc8d3 Contact: <sip:170725014@IP_PROVIDER_ASTERISK> Call-ID: 5079515a2060a26155be2cbf6b73dc78@IP_PROVIDER_ASTERISK CSeq: 102 ACK User-Agent: MARS Max-Forwards: 70 Content-Length: 0 Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

2 2

[Kamailio-Users] NAT traversal
by BERGANZ François 23 Feb '09

23 Feb '09

Hello, Is it possible to do NAT traversal without RTP-PROXY ? If yes, have you an example? Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

7 18

[Kamailio-Users] how to prepend area code into a local call phone number
by Jinsong Hu 23 Feb '09

23 Feb '09

Hi, Sir: In the US, assume a person from 1-408-1234567 is dialing phone number 1234568. how do I figure out the caller's country code and area code and prepend to the local number ? I tried prefix($(from.user{s.substr, 0, 4})); and openser reported as parser error in config file, bad argument, string expected. Jimmy

2 1

[Kamailio-Users] Using set_advertised_address with variable
by catalina oancea 23 Feb '09

23 Feb '09

Hi Does anybody know if there is a way to use the set_advertised_address function with an avp or variable parameter instead of plain string? I need to change the IP in the Via header with the value of a variable. Thanks Catalina

2 1

[Kamailio-Users] missing file ldap.h
by Husna Bte Mad Baguri 23 Feb '09

23 Feb '09

Dear Whom May Concern, When I try to compile ldap module in kamailio I get error,ldap.h :No such file or directory. Where can I get that file.please someone help me. Regards Husna ________________________________ This e-mail and the attachments are intended solely for the person to whom it has been addressed. It contains privileged and/or confidential information and is privileged or otherwise protected from disclosure. If you are not the person for whom this e-mail was intended, or the e-mail has reached you by mistake, please delete it immediately and inform us of the error. Our e-mail address is administrator(a)uniten.edu.my. All opinions, conclusions and other information in this message that do not relate to the official business of Universiti Tenaga Nasional (UNITEN) shall be understood as neither given nor endorsed by UNITEN. UNITEN shall not be responsible for any activity that may be considered as illegal and/or improper use of e-mail and UNITEN further is claims and shall not accept liability for any content of this e-mail, or for the consequences of any actions taken on the basis of the information provided, unless that information is subsequently confirmed in writing. WARNING Internet communications cannot be guaranteed to be secured or error-free as information could be intercepted, corrupted, lost, arrive late or contain viruses. As such, we do not accept liability for any errors or omissions in the content of this message which may arise as a result of internet transmission. UNITEN does not authorize any of its employees to make any defamatory or seditious statements or commit any offence which is contrary to the laws of Malaysia. Any such communications and/or actions by such employees are outside the scope of employment of the said individuals and UNITEN shall not be liable for such communications and/or actions.

2 1

Re: [Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc
by Daniel-Constantin Mierla 20 Feb '09

20 Feb '09

Hello, On 02/13/2009 12:08 PM, Abdul Hakeem wrote: > Hi, > Are you able to put me in touch with the guys using the SDP parser ? > direct your emails to mailing list all the time. Those people are there, the developer of SDP parser and others using it. You will get much faster response and maybe better than writing private emails. Thanks, Daniel > Cheers, > Abdul Hakeem > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] > Sent: 11 February 2009 15:20 > To: alhakeem(a)ipextelecom.net; kamailio > Subject: Re: [Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc > > Hello, > > On 02/11/2009 03:20 PM, Abdul Hakeem wrote: > >> Hi, >> >> Just wanted to touch base on the parser to find out if you are still >> pursuing this. >> >> > it is still on my todo list, however didn't make it in the upcoming 1.5.0. > Ovidiu Sas did some commits to the SDP parser and I know couple of others > using it with success. > > Please open a feature request at: > http://sourceforge.net/tracker/?atid=743023&group_id=139143&func=browse > > It is easier to track and not forgot about it for the next release. > > Thanks, > Daniel > > > >> I am in the mind of purchasing some Cisco equipment to do the Sip-MAP >> gateway from a GSM network. >> I have someone else from the OpenBTS with me on this. >> The object is to provide a full fledged sip based routing of voice, >> sms and multimedia calls from gsm handsets to BTS with a Sip server >> > backend. > >> Cheers, >> Abdul Hakeem >> >> -----Original Message----- >> From: users-bounces(a)lists.kamailio.org >> [mailto:users-bounces@lists.kamailio.org] On Behalf Of >> Daniel-Constantin Mierla >> Sent: 03 September 2008 17:45 >> To: Kristian Kielhofner >> Cc: users(a)lists.kamailio.org >> Subject: Re: [Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc >> >> Hello, >> >> On 09/03/08 19:10, Kristian Kielhofner wrote: >> >> >>> Hello everyone, >>> >>> Feel free to set $OSSPROXY to whatever you like = >>> Kamailio/OpenSIPS/OpenSER (for those of you who can't let go). >>> >>> $OSSPROXY is my favorite SIP toolkit. In addition to being an >>> excellent proxy and registrar it serves as a platform to do SIP >>> mangling and wrangling. With that in mind let me propose a crazy >>> idea... >>> >>> How hard would it be to get $OSSPROXY to be able to parse what is >>> commonly referred to as SIP-T (RFC 3372)? I understand $OSSPROXY has >>> no business getting too involved in the SDP but I feel it makes sense >>> in some instances (like this one). What if I want to make a routing >>> decision based on an encapsulated ISUP parameter of some sort? Who >>> knows what it could be!?!? Maybe there are some other uses but that >>> makes the most sense and seems to be the most practical. It should >>> be possible to develop an $OSSPROXY module to: >>> >>> - Use the SDP parser to find the ISUP SDP part: >>> >>> http://www.kamailio.net/dokuwiki/doku.php/development:sdp-parsing >>> >>> - Pass that part into a "real" ISUP parser (if necessary). I >>> couldn't seem to find the perfect library but perhaps something like >>> this could >>> work: >>> >>> http://www.openss7.org/isup.html >>> >>> - Make the ISUP parameters available via AVPs, pseudo variables, etc >>> for use in the $OSSPROXY script. >>> >>> I don't really have a need for this but it seems like it could be >>> cool. Am I nuts? >>> >>> >>> >>> >> I would expose the body as a dynamic structure, so components will be >> accessible in the script, e.g. (naming schema just for example), >> >> if($body(0=>content-type)=="application/sdp") >> { >> if($body(0=>m=>type) == "audio") >> { >> } >> } >> >> Based on the available content type parsers (sdp, isup), the structure >> has different attributes. >> >> Cheers, >> Daniel >> >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com >> >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >> >> > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > -- Daniel-Constantin Mierla http://www.asipto.com

5 13

[Kamailio-Users] lookup("location") problem
by BERGANZ François 19 Feb '09

19 Feb '09

Hello, I do as the example in the module registrar doc: lookup("location"); switch ($?) { case -1: case -3: sl_send_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Not Found"); exit; }; And SER report me ERROR:core:pv_parse_spec: wrong char [$/36] in [$?] at [0 (0)] Know you why? Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

3 3

[Kamailio-Users] call keepalived using rtimer?
by Noel R. Morais 19 Feb '09

19 Feb '09

Hi Guys, I would like to know if is possible to make a keepalive for active calls (dialog module)? Well, I think it is but is there a better way than making a selec into the dialog table and building the INVITEs? Thanks, Noel

6 8

[Kamailio-Users] SIP URI Extracting
by Ramu 19 Feb '09

19 Feb '09

Hi, I would like to extract uri part from SIP Request uri. Say I received a INVITE sip:1234567890@ip Now I would like to take 4567890 the uri and send it as a new INVITE. Ramu

3 2

[Kamailio-Users] LCR in Kamailio 1.4
by Asim Riaz 19 Feb '09

19 Feb '09

Hello, I am using LCR, for some reason load_gws() function is not finding the gateway in DB, it was working in the start then added some more routes and after that everything stopped working, I am not sure where to start troubleshooting, any idea ? kamctl lcr show lcr routes +----+--------+----------+--------+----------+ | id | prefix | from_uri | grp_id | priority | +----+--------+----------+--------+----------+ | 9 | 0 | | 1 | 1 | +----+--------+----------+--------+----------+ lcr gateways +------------+---------------+------+------------+-----------+--------+-------+------+-------+ | gw_name | ip_addr | port | uri_scheme | transport | grp_id | strip | tag | flags | +------------+---------------+------+------------+-----------+--------+-------+------+-------+ | XXXX | XXXX | 5060 | 1 | 1 | 1 | 0 | | 0 | +------------+---------------+------+------------+-----------+--------+-------+------+-------+ my routing script if(!load_gws()) { xlog("L_INFO", "[ROUTE-X] Unable To Find Gateway "); sl_send_reply("503", "unable to find gateway"); exit; } Thanks in advance. Asim

3 6

[Kamailio-Users] related free voip providers
by Daniel-Constantin Mierla 19 Feb '09

19 Feb '09

Hello, I posted a summary of three free voip providers running sip-router.org related sip servers. It is based on my experience as user of all these (an other) services. The providers listed on the page are not ranked, just presented in alphabetic order, pretty summary about each one. I plan a longer version separately, one by one. It is a good starting point for those willing to see what can be done with Kamailio (OpenSER) and/or SIP Express Router (SER). Another benefit is for users looking to get an independent SIP address, not related to current job or location. I created the post after some private inquires, but I think it worth for public audience: http://miconda.wordpress.com/2009/02/17/top-three-free-voip-services/ In the same spirit, you can check the second edition of (my personal) Kamailio (OpenSER) awards: http://miconda.wordpress.com/2009/01/19/kamailio-openser-2008-awards/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] SIREMIS 0.9.2 released - Support for Carrierroute
by Elena-Ramona Modroiu 19 Feb '09

19 Feb '09

Hi all, a new version of SIREMIS, web management interface for Kamailio (OpenSER) and SIP-Router.org, is out as 0.9.2. Main improvements refer to addition of support to manage the carrierroute module: * management of carrierroute table * management of carrierfailureroute table * management of carrier_name table * management of carrier_domain table View screenshots: http://www.asipto.com/gallery/v/siremis/ Demo site (it works on a database with random data, username: admin, password: admin): http://siremis.asipto.com/demo/ Download: http://siremis.asipto.com/pub/downloads/siremis/ Install/Upgrade documentation: http://siremis.asipto.com/install/ Regards, Ramona

1 0

[Kamailio-Users] Config
by NYam 19 Feb '09

19 Feb '09

Dear Sir, I have problem . wenn i root:~ # kamctl start INFO: Starting Kamailio : ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed root:~ # kamailio -c Feb 17 11:01:49 [10767] ERROR:core:main: loading config file(/usr//etc/kamailio/kamailio.cfg): No such file or directory Feb 17 11:01:49 [10767] WARNING:core:fm_free: free(0) called Feb 17 11:01:49 [10767] WARNING:core:fm_free: free(0) called root:~ # kamailio -f /etc/kamailio/kamailio.cfg then i see udp: 127.0.0.1 [127.0.0.1]:5060 ********************* ********************** tcp: 127.0.0.1 [127.0.0.1]:5060 ok Where is Problem ? can you help me? with kind regards Nyam

3 2

[Kamailio-Users] Kamailio 1.4.3 Carrierroute BYE does not
by unknown＠1.org 19 Feb '09

19 Feb '09

Hello: I have this scenario: I already rad about the bye2bye, the dialog module, i was not able to see any light there. Here is the scenario A--GW--------Kamailio(carrierroute)---------GW1---B |--------------GW2 |--------------GW3 (etc) the capture |<-------------BYE---- -<----NOTHING HERE<<<-| the By does go back if the B user hangs it is not even attempted in the kamailio, per the traces. the BY generates records in the CRD from B side if i hang manually the bye generates CDR records from the A side. Again the Bye is understood by the kamailio and generates the CDR records, that is Not the problem, the Problem is i have the user A in silence, i am talking about the Signaling Only, not Media, no NAT. I even tried the mediaproxy (just trying) the config is like the: http://voipembedded.com/resources/openser_cr.cfg i added if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails exit; } route(10); } else { /* uncomment the following lines if you want to enable presence */ ##if (is_method("SUBSCRIBE") && $rd == "your.server.ip.address") { ## # in-dialog subscribe requests ## route(200); ## exit; ##} if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); xlog("L_WARN", "[$mi] discarding ACK\n"); exit; } this is the main difference for now, i changed to use as a basis example the Example 1.45. Configuration example - Routing to user tree. same results. i tried adding the record_route in the invite also, same result. At this point i have no clue what i am missing. Please let me know if this should work or not, and if yes, based in that basic example, what i should add.

2 8

[Kamailio-Users] A2billing Project <-> Openser
by BERGANZ François 18 Feb '09

18 Feb '09

Hello, I use Asterisk2billing to do prepaidit works fine! But, now, I need to auth my users in the openser and forward the call by loadbal to asterisk(s) which use Asterisk2billing. Anybody know how to do? Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

2 14

[Kamailio-Users] SIP-T
by Kaeman Chris 18 Feb '09

18 Feb '09

Hello, Does opensips support SIP-T? Thank you! KC Add more friends to your messenger and enjoy! Go to http://messenger.yahoo.com/invite/

3 3

[Kamailio-Users] SEMS (bugfix) release 1.0.1
by Stefan Sayer 17 Feb '09

17 Feb '09

Hello, The 1.0.1 release of SEMS, the SIP Express Media Server [0], is now available for download [1]. The 1.0.1 release contains all fixes applied to the 1.0 branch, and should be a drop-in replacement for anyone still using 1.0.0. Update is recommended to everyone who is still using 1.0.0-rcX or any later version from the 1.0 branch. Debian packages for etch/64 are also available [2]. Best Stefan Sayer [0] http://iptel.org/sems/ [1] http://ftp.iptel.org/pub/sems/sems-1.0.1.tar.gz [2] http://ftp.iptel.org/pub/sems/1.0/1.0.1/packages/debian Apologies if you receive multiple copies of this message. -- Stefan Sayer VoIP Services stefan.sayer(a)iptego.com www.iptego.com IPTEGO GmbH Wittenbergplatz 1 10789 Berlin Germany Amtsgericht Charlottenburg, HRB 101010 Geschaeftsfuehrer: Alexander Hoffmann

1 0

[Kamailio-Users] minutes for IRC meeting 17.02.2009, release dates
by Henning Westerholt 17 Feb '09

17 Feb '09

Hi all, the minutes for todays IRC meeting are now available in our wiki [1]. Among the discussed topics we decided about a date for the upcoming 1.5.0 release, this is scheduled for the 2th march 2009. The first release that will use the new common sip-router core [2] is scheduled for May. After 1.5.0 we will also do a bug fix release (1.4.4) for our latest stable branch. All dates can be found in our wiki as well at [3]. Cheers, Henning [1]http://www.kamailio.org/dokuwiki/doku.php/development:irc-meeting-minutes… [2]http://sip-router.org/benefits/ [3]http://www.kamailio.org/dokuwiki/doku.php/development:release-policy#date… -- Henning Westerholt - Development Consumer Products / DSL Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

1 0

[Kamailio-Users] IRC meeting 17th februar, 15:00 UTC
by Henning Westerholt 17 Feb '09

17 Feb '09

Hi all, as the release cycle for 1.5.0 slowly come to an end, i think its time for a new IRC developer meeting, to focus on the upcoming development. I propose next Tuesday, 17th Feb 15:00 UTC. I've created a wiki page to collect an agenda for this meeting, and already added some content i would like to discuss. Please add your yours also and comment on already existing topics. Agenda: http://www.kamailio.org/dokuwiki/doku.php/development:irc-meeting-agenda-17… IRC meetings page: http://www.kamailio.org/dokuwiki/doku.php/development:irc-meetings Cheers, Henning

2 3

[Serusers] Problem with REGISTER
by buslik 17 Feb '09

17 Feb '09

Hi all, I have just default ser.cfg And it seems that if (uri==myself) never happens. SER trying to relay to <my domain name>:5060 and there my Gigaset C470IP just replies 405 Not Allowed. How to explain to the SER that it has <my domain name>? I have put alias=<my domain name> into my ser.cfg and SIP_DOMAIN=<my domain name> into env. It does no help. How to get any log out of it? -- Slava.

1 0

[Kamailio-Users] Lost
by daniel medina 17 Feb '09

17 Feb '09

Hi everybody, I am trying to set up OpenSer as a Load balancing for Asterisk. I really need to know where to start. Please can anyone give me a hint where to star or if there is any tutorial??? Thanks in advance Daniel

3 2

Re: [Kamailio-Users] ERROR:uac:replace_from
by Daniel-Constantin Mierla 16 Feb '09

16 Feb '09

Are you sure uac_replace_from() was called for initial invite? There has to be a new param (vsf) in the Recor-Route header and it is not there. Cheers, Daniel On 02/16/2009 04:50 PM, BERGANZ François wrote: > No. Time Source Destination Protocol Info > > [...] > > There is no 200OK because, the bye isn't good! > > > Thank you > > > Cordialement, > BERGANZ François > > > http://www.acropolistelecom.net >  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. > > > -----Message d'origine----- > De : Daniel-Constantin Mierla [mailto:miconda@gmail.com] > Envoyé : lundi 16 février 2009 15:37 > À : BERGANZ François > Cc : users(a)lists.kamailio.org > Objet : Re: [Kamailio-Users] ERROR:uac:replace_from > > > > On 02/16/2009 04:07 PM, BERGANZ François wrote: > >> Now, I have modparam("uac","from_restore_mode","none") >> >> I control everything! Nothing in automatic! >> >> > :-) -- the power in your hands -- as said in the previous email, if you > send me the invite and appropriate 200ok, I can tell whose fault is. > > Cheers, > Daniel > > >> Cordialement, >> BERGANZ François >> >> >> http://www.acropolistelecom.net >>  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. >> >> -----Message d'origine----- >> De : Daniel-Constantin Mierla [mailto:miconda@gmail.com] >> Envoyé : lundi 16 février 2009 14:07 >> À : BERGANZ François >> Cc : users(a)lists.kamailio.org >> Objet : Re: [Kamailio-Users] ERROR:uac:replace_from >> >> >> >> On 02/16/2009 03:03 PM, BERGANZ François wrote: >> >> >>> Then, if it is automatic: >>> >>> If I change with uac_replace_from() the INVITE, the bye have to be the same ? >>> In my case, it doesn't >>> >>> >>> >> Paste the BYE here, maybe the Route parameter used for auto-update is >> missing... >> >> Cheers, >> Daniel >> >> >> >>> Cordialement, >>> BERGANZ François >>> >>> >>> http://www.acropolistelecom.net >>>  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. >>> >>> >>> -----Message d'origine----- >>> De : Daniel-Constantin Mierla [mailto:miconda@gmail.com] >>> Envoyé : lundi 16 février 2009 12:39 >>> À : BERGANZ François >>> Cc : users(a)lists.kamailio.org >>> Objet : Re: [Kamailio-Users] ERROR:uac:replace_from >>> >>> >>> >>> On 02/16/2009 01:33 PM, BERGANZ François wrote: >>> >>> >>> >>>> It is for a bye request >>>> >>>> >>>> >>>> >>> BYE is withing dialog (practically ending the dialog). You do not need >>> to replace the From for it. It is done automatically -- see my other post. >>> >>> Cheers, >>> Daniel >>> >>> >>> >>> >>>> Cordialement, >>>> BERGANZ François >>>> >>>> >>>> http://www.acropolistelecom.net >>>>  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. >>>> >>>> >>>> -----Message d'origine----- >>>> De : Daniel-Constantin Mierla [mailto:miconda@gmail.com] >>>> Envoyé : lundi 16 février 2009 12:25 >>>> À : BERGANZ François >>>> Cc : users(a)lists.kamailio.org >>>> Objet : Re: [Kamailio-Users] ERROR:uac:replace_from >>>> >>>> Hello, >>>> >>>> On 02/16/2009 01:10 PM, BERGANZ François wrote: >>>> >>>> >>>> >>>> >>>>> Hello, >>>>> >>>>> I have an error with uac_replace_from() function. >>>>> >>>>> I want to do uac_replace_from(“$fn”,” sip:181812012@217.64.x.x”); >>>>> >>>>> I have that error: ERROR:uac:replace_from: decline FROM replacing in >>>>> sequential request in auto mode (has TO tag) >>>>> >>>>> Have you an idea? >>>>> >>>>> >>>>> >>>>> >>>>> >>>> you call uac_replace_from() for SIP messages within dialog and the >>>> replace mode is auto (which means is done automatically by kamailio) - >>>> therefore will be a duplicate. >>>> >>>> Either check for to-tag when you do the replacement or set replace mode >>>> to manual. >>>> >>>> Cheers, >>>> Daniel >>>> >>>> >>>> >>>> >>>> >>> >>> >>> >> >> > > -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] ERROR:uac:replace_from
by BERGANZ François 16 Feb '09

16 Feb '09

Hello, I have an error with uac_replace_from() function. I want to do uac_replace_from($fn, sip:181812012@217.64.x.x); I have that error: ERROR:uac:replace_from: decline FROM replacing in sequential request in auto mode (has TO tag) Have you an idea? Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

2 9

Re: [Kamailio-Users] ERROR:uac:replace_from
by Daniel-Constantin Mierla 16 Feb '09

16 Feb '09

On 02/16/2009 03:36 PM, BERGANZ François wrote: > the bye receive by the SER > The route parameter is missing, please send me the INVITE going out from kamailio and the 200ok for invite received by Kamailio. Should reveal whose fault is. Cheers, Daniel > thank you > > > > [...] > > > > Cordialement, > BERGANZ François > > > http://www.acropolistelecom.net >  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. > > > -----Message d'origine----- > De : Daniel-Constantin Mierla [mailto:miconda@gmail.com] > Envoyé : lundi 16 février 2009 14:07 > À : BERGANZ François > Cc : users(a)lists.kamailio.org > Objet : Re: [Kamailio-Users] ERROR:uac:replace_from > > > > On 02/16/2009 03:03 PM, BERGANZ François wrote: > >> Then, if it is automatic: >> >> If I change with uac_replace_from() the INVITE, the bye have to be the same ? >> In my case, it doesn't >> >> > > Paste the BYE here, maybe the Route parameter used for auto-update is > missing... > > Cheers, > Daniel > > >> Cordialement, >> BERGANZ François >> >> >> http://www.acropolistelecom.net >>  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. >> >> >> -----Message d'origine----- >> De : Daniel-Constantin Mierla [mailto:miconda@gmail.com] >> Envoyé : lundi 16 février 2009 12:39 >> À : BERGANZ François >> Cc : users(a)lists.kamailio.org >> Objet : Re: [Kamailio-Users] ERROR:uac:replace_from >> >> >> >> On 02/16/2009 01:33 PM, BERGANZ François wrote: >> >> >>> It is for a bye request >>> >>> >>> >> BYE is withing dialog (practically ending the dialog). You do not need >> to replace the From for it. It is done automatically -- see my other post. >> >> Cheers, >> Daniel >> >> >> >>> Cordialement, >>> BERGANZ François >>> >>> >>> http://www.acropolistelecom.net >>>  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. >>> >>> >>> -----Message d'origine----- >>> De : Daniel-Constantin Mierla [mailto:miconda@gmail.com] >>> Envoyé : lundi 16 février 2009 12:25 >>> À : BERGANZ François >>> Cc : users(a)lists.kamailio.org >>> Objet : Re: [Kamailio-Users] ERROR:uac:replace_from >>> >>> Hello, >>> >>> On 02/16/2009 01:10 PM, BERGANZ François wrote: >>> >>> >>> >>>> Hello, >>>> >>>> I have an error with uac_replace_from() function. >>>> >>>> I want to do uac_replace_from(“$fn”,” sip:181812012@217.64.x.x”); >>>> >>>> I have that error: ERROR:uac:replace_from: decline FROM replacing in >>>> sequential request in auto mode (has TO tag) >>>> >>>> Have you an idea? >>>> >>>> >>>> >>>> >>> you call uac_replace_from() for SIP messages within dialog and the >>> replace mode is auto (which means is done automatically by kamailio) - >>> therefore will be a duplicate. >>> >>> Either check for to-tag when you do the replacement or set replace mode >>> to manual. >>> >>> Cheers, >>> Daniel >>> >>> >>> >>> >> >> > > -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] cfg scripts bank - inside group dial
by Daniel-Constantin Mierla 16 Feb '09

16 Feb '09

Hello, there were many useful hints and examples of how to solve specific needs or add new features just by config language. However, it is hard to search and find them back in the archive after a while. Therefore I created a new wiki page to collect them. If you see such example flowing on the mailing lists, take it and add it there or at least set the link to the mailing list post. Here is the page: http://www.kamailio.org/dokuwiki/doku.php/cfg-scripts-bank:main-index I added first example that adds inside group dial feature. You can group users by interest and they can call each other using short numbers instead of entire extension. Check it here: http://www.kamailio.org/dokuwiki/doku.php/cfg-scripts-bank:group-dial Practically is less than 10 lines, with loading additional modules and setting the parameters. Looking forward to your contributions there! Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Re: [Serusers] Serusers Digest, Vol 70, Issue 4
by 郭灵犀 16 Feb '09

16 Feb '09

help 在2009-02-06，serusers-request(a)lists.iptel.org 写道： >Send Serusers mailing list submissions to > serusers(a)lists.iptel.org > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.iptel.org/mailman/listinfo/serusers >or, via email, send a message with subject or body 'help' to > serusers-request(a)lists.iptel.org > >You can reach the person managing the list at > serusers-owner(a)lists.iptel.org > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of Serusers digest..." > > >Today's Topics: > > 1. Re: configure SER and SEMS to work together (Stefan Sayer) > 2. preload_udomain error (caio) > 3. Re: preload_udomain error (solved) (caio) > 4. Re: Error loading "trusted" table (permissions.so) (caio) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Thu, 05 Feb 2009 13:25:00 +0100 >From: Stefan Sayer <stefan.sayer(a)iptego.com> >Subject: Re: [Serusers] configure SER and SEMS to work together >To: Martin Lane <martin.l.lane(a)gmail.com> >Cc: sems(a)iptel.org, Serusers(a)lists.iptel.org >Message-ID: <498ADA9C.4050302(a)iptego.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >Hello, > > >o Martin Lane [02/04/09 20:17]: >> I am following the 'mini-Howto on how to configure SER and SEMS to work >> together' guide on iptel.org. So I have a separate directory for >as it says at the top of that howto, with a more recent version of SEMS >this is not required any more; especially if you are using 1.1.0 >version. Also note that there is the 1.1.0 release available >(http://ftp.iptel.org/pub/sems/sems-1.1.0.tar.gz). > >For SEMS 1.1.0 (recommended!), you can use the default configuration, >just configure 'application' and the server's IP address and port used >in sems.conf. Then just send the call to the SIP IP:port, e.g. using ser >with t_relay_to_udp("<sip_ip>","<sip_port>"); > >for completeness, the following comments: > >> integrating the SER and SEMS functionality. For development purposes is >> there a switch to change the path were SER looks for the modules? As >no, you need to change every loadmodule line. > >> well control which version of SEMS is being used? I've installed Sip >> Express Media Server (1.1.0-rc1 (i386/linux)) as well as the version >> from the howto for testing. >> >> My basic problem is I currently get a version mismatch error: >> >> 0(9590) ERROR: module version mismatch for >> /usr/local/lib/ser/modules/sl.so; core: ser 0.9.6-sems (i386/linux); >> module: ser 0.9.6 (i386/linux) >> > >Your specific error comes from that the module you are trying to load >was not compiled with the same version, e.g. you have one ser installed >into /usr/local and you have another one in a local directory. > >Regards >Stefan > >> The appropriate modules are located in under my ser-sems test dir and I >> have not been successful in redirecting SER to look there. >> >> Also, I have modified ser.cfg and sems.cfg for my system but I did not >> find a default ser_sems.cfg. Is this file necessary and where is the >> default location? >> >> Thanks for any help or advice. >> >> mll >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers > >-- >Stefan Sayer >VoIP Services > >stefan.sayer(a)iptego.com >www.iptego.com > >IPTEGO GmbH >Am Borsigturm 40 >13507 Berlin >Germany > >Amtsgericht Charlottenburg, HRB 101010 >Geschaeftsfuehrer: Alexander Hoffmann > > >------------------------------ > >Message: 2 >Date: Thu, 05 Feb 2009 12:18:40 -0200 >From: caio <elcaio(a)gmail.com> >Subject: [Serusers] preload_udomain error >To: serusers(a)lists.iptel.org >Message-ID: <498AF540.4030102(a)gmail.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >Hi all, > >Running ser0.9.7 on Solaris 10 (mysql 5.0.51 + serweb 0.9.6) > >Have this problem when I stop ser service, and then start it again (ser >-E -D).. > > 0(0) preload_udomain: ERROR: bad callid record in table aliases > 0(0) preload_udomain: ERROR: for username 579981, contact >sip:579981@mydomain.com > 0(0) preload_udomain: ERROR: skipping... > >And finally it dies with: > > 0(0) get_connection(): Connection not found in the pool > Fallo de segmentaci??n (core dumped) > >It happens with many records on aliases table and happens too with only >one record: > >mysql> select * from aliases\G >*************************** 1. row *************************** > username: 579981 > domain: 10.10.1.1 > contact: sip:579981@mydomain.com > received: NULL > expires: 2009-02-05 11:45:21 > q: 1.00 > callid: >The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything > cseq: 42 >last_modified: 2009-02-05 11:45:46 > replicate: 0 > state: 0 > flags: 128 > user_agent: SIP Express Router FIFO >1 row in set (0.00 sec) > > >SER start normally if I delete all the records on this table. But I need >create the aliases again, and this is not the normal behaviour..I think. > >Here the description of aliases table: > >mysql> desc aliases; >+---------------+---------------------+------+-----+---------------------+-------+ >| Field | Type | Null | Key | Default > | Extra | >+---------------+---------------------+------+-----+---------------------+-------+ >| username | varchar(64) | NO | PRI | > | | >| domain | varchar(128) | NO | PRI | > | | >| contact | varchar(255) | NO | PRI | > | | >| received | varchar(255) | YES | | NULL > | | >| expires | datetime | NO | | 2020-05-28 21:32:15 >| | >| q | float(10,2) | NO | | 1.00 > | | >| callid | varchar(255) | NO | | Default-Call-ID > | | >| cseq | int(11) | NO | | 42 > | | >| last_modified | timestamp | NO | | CURRENT_TIMESTAMP > | | >| replicate | int(10) unsigned | NO | | 0 > | | >| state | tinyint(1) unsigned | NO | | 0 > | | >| flags | int(11) | NO | | 0 > | | >| user_agent | varchar(50) | NO | | > | | >+---------------+---------------------+------+-----+---------------------+-------+ >13 rows in set (0.01 sec) > > >Important: when "location" table keeps some records, ser doesn't start too: > >mysql> select * from location\G >*************************** 1. row *************************** > username: 579981 > domain: mydomain.com > contact: sip:579981@10.10.60.115:5060 > received: NULL > expires: 2009-02-05 11:46:16 > q: -1.00 > callid: 1f46c06d5e034425286f095b11cf4a1a(a)10.10.60.115 > cseq: 6 >last_modified: 2009-02-05 11:45:46 > replicate: 0 > state: 0 > flags: 0 > user_agent: CM5K-TA2S (807100.02) >1 row in set (0.00 sec) > >Here too I must to delete all the records to let SER start successfully. > >Does anyone knows about this issue? > >Regards, >Claudio > > >------------------------------ > >Message: 3 >Date: Thu, 05 Feb 2009 17:13:09 -0200 >From: caio <elcaio(a)gmail.com> >Subject: Re: [Serusers] preload_udomain error (solved) >To: serusers(a)lists.iptel.org >Message-ID: <498B3A45.5070501(a)gmail.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >caio escribi?: >> Hi all, >> >> Running ser0.9.7 on Solaris 10 (mysql 5.0.51 + serweb 0.9.6) >> >> Have this problem when I stop ser service, and then start it again (ser >> -E -D).. >> >> 0(0) preload_udomain: ERROR: bad callid record in table aliases >> 0(0) preload_udomain: ERROR: for username 579981, contact >> sip:579981@mydomain.com >> 0(0) preload_udomain: ERROR: skipping... >> >[cut] > >solved. >there was an old installation of mysql libs (in /usr/sfw/), then I >removed it, and finally recompile mysql.so module including correct >mysql.h (/opt/csw/). > >cheers, >caio > > >------------------------------ > >Message: 4 >Date: Thu, 05 Feb 2009 17:18:03 -0200 >From: caio <elcaio(a)gmail.com> >Subject: Re: [Serusers] Error loading "trusted" table (permissions.so) >To: serusers(a)lists.iptel.org >Message-ID: <498B3B6B.20606(a)gmail.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >Miklos Tirpak escribi?: > >>> Seem that VAL_TYPE of "val + 2" isn't STRING, read it as DOUBLE >>> (DB_DOUBLE = 1). >>> Do not know why. >>> And because of this the 'if' conditions never are satisfy on trusted.c >>> (line #396): >>> >>> if ((ROW_N(row + i) == 3) && >>> (VAL_TYPE(val) == DB_STRING) && !VAL_NULL(val) && >>> (VAL_TYPE(val + 1) == DB_STRING) && !VAL_NULL(val >>> + 1) && >>> (VAL_TYPE(val + 2) == DB_STRING) && !VAL_NULL(val >>> + 2)) { >> >> >> Thank you very much for the debug lines! I think SER simply uses the >> field types returned by the mysql server, so there might be a problem >> with the server or library. You can check what the field types are along >> with their names returned by the server in modules/mysql/res.c, search >> for the mysql_fetch_fields function. >> >> Miklos >> > >Miklos, > >I've resolved this issue and a similar one ("preload_udomain error"). >There was an old installation of mysql libs (in /usr/sfw/), then I >removed it, and finally recompile mysql.so module including correct >mysql.h (/opt/csw/). > >Now fields returned are with their correct type. > >Thank you.. > >Caio > > >------------------------------ > >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers > > >End of Serusers Digest, Vol 70, Issue 4 >***************************************

1 0

[Kamailio-Users] mediaproxy
by NYam 14 Feb '09

14 Feb '09

Dear Sir, As it was stated in your website, KAMAILIO (OpenSER) is available in binary format for several embedded platforms, including Linksys NSLU. Does it mean that mediaproxy is already installed into the platform with KAMAILIO? If not, how mediaproxy will be installed? Will it be hard to do so? with kind regards Nyam Sharibu

3 3

[Kamailio-Users] SIREMIS feedback
by Elena-Ramona Modroiu 14 Feb '09

14 Feb '09

Hi, it seems that meeting at FOSDEM brought lot of ideas around. Stefan already had a good suggestion - integration of SIREMIS with SEMS giving the possibility to monitor SEMS servers in the same nice manner from a web page. I would like to collect suggestions about what would be good to have in SIREMIS, what should be improved. My goal up to Kamailio 1.5 release is to add support for carrierroute, a cdr rating engine and cleanup the source tree in order to upload the sourceforge svn - this would make easier for others to contribute. However, if it is something we find very useful then I will prioritize. There is time to add new features after the release, as well. Please shoot here any crazy idea you have for a web management interface of Kamailio and later SIP Router. Regards, Ramona

4 7

[Kamailio-Users] UDP packet loss
by Alex R.S.M 13 Feb '09

13 Feb '09

Hi, I am performing SIP to SIP performance testing using Kamailio proxy (UAC-Kamailio-UAC). After increasing the load to 300 calls/sec Kamailio begins dropping UDP packets. (250 cps works fine). CPU utilization and network bandwidth looks fine. I think the bottleneck is the Linux UDP buffer size of the proxy server. Here is the proxy server information: $ uname -a Linux lab-linux01 2.6.24-19-server #1 SMP Wed Jun 18 14:44:47 UTC 2008 x86_64 GNU/Linux $ sudo sysctl -a | grep net.core.rmem_max net.core.rmem_max = 131071 The question is if I increase the linux UDP buffer size do I need to change any configuartion parameter (like MAX_RECV_BUFFER_SIZE) in Kamailio? Thanks, Alex R _________________________________________________________________ Windows Live™: Keep your life in sync. http://windowslive.com/howitworks?ocid=TXT_TAGLM_WL_t1_allup_howitworks_022…

3 6

[Kamailio-Users] Trying to compile openser 1.3.3 without PKG_MALLOC support
by toly 13 Feb '09

13 Feb '09

Greetings, Trying to compile openser 1.3.3 without PKG_MALLOC, just using OS malloc Here's me Makefile.defs DEFS+= $(extra_defs) \ -DNAME='"$(MAIN_NAME)"' -DVERSION='"$(RELEASE)"' -DARCH='"$(ARCH)"' \ -DOS='"$(OS)"' -DCOMPILER='"$(CC_VER)"' -D__CPU_$(ARCH) -D__OS_$(OS) \ -D__SMP_$(ISSMP) -DCFG_DIR='"$(cfg-target)"'\ -DSHM_MEM -DSHM_MMAP \ #-DPKG_MALLOC \ -DUSE_IPV6 \ -DUSE_MCAST \ -DUSE_TCP \ -DDISABLE_NAGLE \ -DHAVE_RESOLV_RES \ -DSTATISTICS \ -DCHANGEABLE_DEBUG_LEVEL \ #-DF_MALLOC \ -DQM_JOIN_FREE \ #-DDBG_QM_MALLOC \ #-DDBG_F_MALLOC \ #-DNO_DEBUG \ #-DNO_LOG \ #-DVQ_MALLOC \ #-DDBG_LOCK \ #-DNOSMP \ #-DEXTRA_DEBUG \ #-DUSE_SHM_MEM It's bailing out compiling blacklists.c blacklists.c: In function `rm_dups': blacklists.c:435: error: syntax error before "else" blacklists.c:440: error: syntax error before "else" blacklists.c: At top level: blacklists.c:443: error: syntax error before "else" blacklists.c:445: warning: type defaults to `int' in declaration of `q' blacklists.c:445: error: invalid type argument of `->' blacklists.c:445: warning: data definition has no type or storage class blacklists.c:446: error: syntax error before '}' token make: *** [blacklists.o] Error 1 Did anybody had a problem compiling openser without pkg_malloc support? I would appreciate any help. TIA Toly. -- View this message in context: http://www.nabble.com/Trying-to-compile-openser-1.3.3-without-PKG_MALLOC-su… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

4 12

[Kamailio-Users] Kernel Message
by Brian Chamberlain 12 Feb '09

12 Feb '09

Hi All, Is this something I should worry about? Feb 12 20:27:22 sw1 kernel: [3273100.087148] INFO: task kamailio:1845 blocked for more than 120 seconds. Feb 12 20:27:22 sw1 kernel: [3273100.087206] "echo 0 > /proc/sys/ kernel/hung_task_timeout_secs" disables this message. Feb 12 20:27:22 sw1 kernel: [3273100.087291] kamailio D 8ce9265a 0 1845 1829 Feb 12 20:27:22 sw1 kernel: [3273100.087346] f6a98da0 00000082 23f7e426 8ce9265a 000701ae f6a98f2c c302bfa0 00000002 Feb 12 20:27:22 sw1 kernel: [3273100.087427] c01344f6 001e17ce 00000001 00000000 c01190e6 f6a98da0 f6a98da0 c02c4dc0 Feb 12 20:27:22 sw1 kernel: [3273100.087538] f74eea84 f74eea8c f74eea88 f6a98da0 c02b8846 f68c3d68 f74eea8c f6a98da0 Feb 12 20:27:22 sw1 kernel: [3273100.087648] Call Trace: Feb 12 20:27:22 sw1 kernel: [3273100.087730] [<c01344f6>] ktime_get +0xd/0x21 Feb 12 20:27:22 sw1 kernel: [3273100.087784] [<c01190e6>] hrtick_start_fair+0xeb/0x12c Feb 12 20:27:22 sw1 kernel: [3273100.087840] [<c02b8846>] __mutex_lock_slowpath+0x50/0x7b Feb 12 20:27:22 sw1 kernel: [3273100.087896] [<c02b86dc>] mutex_lock +0xa/0xb Feb 12 20:27:22 sw1 kernel: [3273100.087945] [<c02a7ea9>] unix_dgram_recvmsg+0x3e/0x245 Feb 12 20:27:22 sw1 kernel: [3273100.087998] [<c01318e9>] autoremove_wake_function+0xd/0x2d Feb 12 20:27:22 sw1 kernel: [3273100.088055] [<c011845d>] __wake_up_common+0x2e/0x58 Feb 12 20:27:22 sw1 kernel: [3273100.088109] [<c024ef76>] sock_recvmsg +0xde/0xf9 Feb 12 20:27:22 sw1 kernel: [3273100.088165] [<c01318dc>] autoremove_wake_function+0x0/0x2d Feb 12 20:27:22 sw1 kernel: [3273100.088223] [<c0117c67>] kunmap_atomic+0x55/0x8a Feb 12 20:27:22 sw1 kernel: [3273100.088273] [<c0117c79>] kunmap_atomic+0x67/0x8a Feb 12 20:27:22 sw1 kernel: [3273100.088327] [<c015ae91>] get_page_from_freelist+0x34f/0x3e3 Feb 12 20:27:22 sw1 kernel: [3273100.088387] [<c024fd9a>] sys_recvfrom +0xb4/0x116 Feb 12 20:27:22 sw1 kernel: [3273100.088438] [<c015b0fb>] __alloc_pages_internal+0xb5/0x34e Feb 12 20:27:22 sw1 kernel: [3273100.088497] [<c0163cd3>] handle_mm_fault+0x257/0x690 Feb 12 20:27:22 sw1 kernel: [3273100.088554] [<c025024d>] sys_socketcall+0x135/0x19e Feb 12 20:27:22 sw1 kernel: [3273100.088610] [<c0103853>] sysenter_past_esp+0x78/0xb1 Feb 12 20:27:22 sw1 kernel: [3273100.091119] [<c02b0000>] acpi_pci_root_add+0x165/0x296 Feb 12 20:27:22 sw1 kernel: [3273100.091174] ======================= Feb 12 20:27:22 sw1 kernel: [3273100.091218] INFO: task kamailio:1846 blocked for more than 120 seconds. Feb 12 20:27:22 sw1 kernel: [3273100.091274] "echo 0 > /proc/sys/ kernel/hung_task_timeout_secs" disables this message. Feb 12 20:27:22 sw1 kernel: [3273100.091358] kamailio D 00000002 0 1846 1829 Feb 12 20:27:22 sw1 kernel: [3273100.091413] f6a995e0 00000086 00000000 00000002 00000000 f6a9976c c3017fa0 00000000 Feb 12 20:27:22 sw1 kernel: [3273100.091495] ffffffff 1d621a39 00000000 00000000 00000000 00000000 00000000 000000ff Feb 12 20:27:22 sw1 kernel: [3273100.091605] f74eea84 f74eea8c f74eea88 f6a995e0 c02b8846 f74eea8c f6b21d68 f6a995e0 Feb 12 20:27:22 sw1 kernel: [3273100.091715] Call Trace: Feb 12 20:27:22 sw1 kernel: [3273100.091790] [<c02b8846>] __mutex_lock_slowpath+0x50/0x7b Feb 12 20:27:22 sw1 kernel: [3273100.091847] [<c02b86dc>] mutex_lock +0xa/0xb Feb 12 20:27:22 sw1 kernel: [3273100.091895] [<c02a7ea9>] unix_dgram_recvmsg+0x3e/0x245 Feb 12 20:27:22 sw1 kernel: [3273100.091949] [<c0129ab0>] lock_timer_base+0x19/0x35 Feb 12 20:27:22 sw1 kernel: [3273100.092002] [<c0129b10>] try_to_del_timer_sync+0x44/0x4a Feb 12 20:27:22 sw1 kernel: [3273100.092057] [<c024ef76>] sock_recvmsg +0xde/0xf9 Feb 12 20:27:22 sw1 kernel: [3273100.092108] [<c01297ec>] process_timeout+0x0/0x5 Feb 12 20:27:22 sw1 kernel: [3273100.092163] [<c01318dc>] autoremove_wake_function+0x0/0x2d Feb 12 20:27:22 sw1 kernel: [3273100.092219] [<c015a0d0>] __rmqueue +0x13/0x174 Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c0117c67>] kunmap_atomic+0x55/0x8a Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c0117c79>] kunmap_atomic+0x67/0x8a Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c015ae91>] get_page_from_freelist+0x34f/0x3e3 Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c024fd9a>] sys_recvfrom +0xb4/0x116 Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c015b0fb>] __alloc_pages_internal+0xb5/0x34e Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c0163cd3>] handle_mm_fault+0x257/0x690 Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c025024d>] sys_socketcall+0x135/0x19e Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c0103853>] sysenter_past_esp+0x78/0xb1 Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c02b0000>] acpi_pci_root_add+0x165/0x296 Feb 12 20:27:22 sw1 kernel: [3273100.092221] ======================= Feb 12 20:27:22 sw1 kernel: [3273100.092221] INFO: task kamailio:22820 blocked for more than 120 seconds. Feb 12 20:27:22 sw1 kernel: [3273100.092221] "echo 0 > /proc/sys/ kernel/hung_task_timeout_secs" disables this message. Feb 12 20:27:22 sw1 kernel: [3273100.092221] kamailio D 854aa937 0 22820 22803 Feb 12 20:27:22 sw1 kernel: [3273100.092221] f762fa40 00000086 2603ada4 854aa937 000767a5 f762fbcc c3021fa0 00000001 Feb 12 20:27:22 sw1 kernel: [3273100.092221] c01344f6 001fcdbd 00000001 00000000 c01190e6 f762fa40 f762fa40 c02c4dc0 Feb 12 20:27:22 sw1 kernel: [3273100.092221] f6b38544 f6b3854c f6b38548 f762fa40 c02b8846 f68a3d68 f6b3854c f762fa40 Feb 12 20:27:22 sw1 kernel: [3273100.092221] Call Trace: Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c01344f6>] ktime_get +0xd/0x21 Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c01190e6>] hrtick_start_fair+0xeb/0x12c Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c02b8846>] __mutex_lock_slowpath+0x50/0x7b Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c02b86dc>] mutex_lock +0xa/0xb Feb 12 20:27:22 sw1 kernel: [3273100.092274] [<c02a7ea9>] unix_dgram_recvmsg+0x3e/0x245 Feb 12 20:27:22 sw1 kernel: [3273100.092328] [<c01318e9>] autoremove_wake_function+0xd/0x2d Feb 12 20:27:22 sw1 kernel: [3273100.092383] [<c015a05d>] __rmqueue_smallest+0x83/0xe3 Feb 12 20:27:22 sw1 kernel: [3273100.092438] [<c024ef76>] sock_recvmsg +0xde/0xf9 Feb 12 20:27:22 sw1 kernel: [3273100.092493] [<c01318dc>] autoremove_wake_function+0x0/0x2d Feb 12 20:27:22 sw1 kernel: [3273100.092551] [<c0117c67>] kunmap_atomic+0x55/0x8a Feb 12 20:27:22 sw1 kernel: [3273100.092601] [<c0117c79>] kunmap_atomic+0x67/0x8a Feb 12 20:27:22 sw1 kernel: [3273100.092653] [<c015ae91>] get_page_from_freelist+0x34f/0x3e3 Feb 12 20:27:22 sw1 kernel: [3273100.092712] [<c024fd9a>] sys_recvfrom +0xb4/0x116 Feb 12 20:27:22 sw1 kernel: [3273100.092764] [<c015b0fb>] __alloc_pages_internal+0xb5/0x34e Feb 12 20:27:22 sw1 kernel: [3273100.092822] [<c0163cd3>] handle_mm_fault+0x257/0x690 Feb 12 20:27:22 sw1 kernel: [3273100.092881] [<c025024d>] sys_socketcall+0x135/0x19e Feb 12 20:27:22 sw1 kernel: [3273100.092937] [<c0103853>] sysenter_past_esp+0x78/0xb1 Feb 12 20:27:22 sw1 kernel: [3273100.092991] [<c02b0000>] acpi_pci_root_add+0x165/0x296 Feb 12 20:27:22 sw1 kernel: [3273100.093046] ======================= Feb 12 20:27:22 sw1 kernel: [3273100.093090] INFO: task kamailio:22821 blocked for more than 120 seconds. Feb 12 20:27:22 sw1 kernel: [3273100.093146] "echo 0 > /proc/sys/ kernel/hung_task_timeout_secs" disables this message. Feb 12 20:27:22 sw1 kernel: [3273100.093230] kamailio D 8543066c 0 22821 22803 Feb 12 20:27:22 sw1 kernel: [3273100.110119] f762ede0 00000082 2603ada4 8543066c 000767a5 f762ef6c c3017fa0 00000000 Feb 12 20:27:22 sw1 kernel: [3273100.110201] c01344f6 001fcdbd 00000002 00000000 c01190e6 f762ede0 f762ede0 c02c4dc0 Feb 12 20:27:22 sw1 kernel: [3273100.110312] f6b38544 f6b3854c f6b38548 f762ede0 c02b8846 f6b3854c f6bfdd68 f762ede0 Feb 12 20:27:22 sw1 kernel: [3273100.110422] Call Trace: Feb 12 20:27:22 sw1 kernel: [3273100.110493] [<c01344f6>] ktime_get +0xd/0x21 Feb 12 20:27:22 sw1 kernel: [3273100.110543] [<c01190e6>] hrtick_start_fair+0xeb/0x12c Feb 12 20:27:22 sw1 kernel: [3273100.110598] [<c02b8846>] __mutex_lock_slowpath+0x50/0x7b Feb 12 20:27:22 sw1 kernel: [3273100.110653] [<c02b86dc>] mutex_lock +0xa/0xb Feb 12 20:27:22 sw1 kernel: [3273100.110702] [<c02a7ea9>] unix_dgram_recvmsg+0x3e/0x245 Feb 12 20:27:22 sw1 kernel: [3273100.110755] [<c01318e9>] autoremove_wake_function+0xd/0x2d Feb 12 20:27:22 sw1 kernel: [3273100.110810] [<c011845d>] __wake_up_common+0x2e/0x58 Feb 12 20:27:22 sw1 kernel: [3273100.110864] [<c024ef76>] sock_recvmsg +0xde/0xf9 Feb 12 20:27:22 sw1 kernel: [3273100.110919] [<c01318dc>] autoremove_wake_function+0x0/0x2d Feb 12 20:27:22 sw1 kernel: [3273100.110976] [<c0117c67>] kunmap_atomic+0x55/0x8a Feb 12 20:27:22 sw1 kernel: [3273100.111027] [<c0117c79>] kunmap_atomic+0x67/0x8a Feb 12 20:27:22 sw1 kernel: [3273100.111079] [<c015ae91>] get_page_from_freelist+0x34f/0x3e3 Feb 12 20:27:22 sw1 kernel: [3273100.111138] [<c024fd9a>] sys_recvfrom +0xb4/0x116 Feb 12 20:27:22 sw1 kernel: [3273100.111189] [<c015b0fb>] __alloc_pages_internal+0xb5/0x34e Feb 12 20:27:22 sw1 kernel: [3273100.111248] [<c0163cd3>] handle_mm_fault+0x257/0x690 Feb 12 20:27:22 sw1 kernel: [3273100.111307] [<c025024d>] sys_socketcall+0x135/0x19e Feb 12 20:27:22 sw1 kernel: [3273100.111362] [<c0103853>] sysenter_past_esp+0x78/0xb1 Feb 12 20:27:22 sw1 kernel: [3273100.111417] [<c02b0000>] acpi_pci_root_add+0x165/0x296 Feb 12 20:27:22 sw1 kernel: [3273100.111471] =======================

3 12

[Kamailio-Users] SIREMIS v0.9.1 Released
by Elena-Ramona Modroiu 12 Feb '09

12 Feb '09

Hi everybody, a new version, 0.9.1, of SIREMIS - the web management interface for Kamailio (OpenSER) and SIP-Router.org - is released. New features and improvements: * send MI commands to Kamailio (OpenSER) via UDP * display the output of a MI command * ability to predefine MI commands that can be selected from a drop-down menu * create graphic charts from statistic data * one or more charts per page * one or more graphics per chart * now new coding required, only configuration changes required to add new charts * drop-down selection for user AVP type Goals for next version: * carrierroute management * rating engine for CDRs View screenshots of the 0.9.1: http://www.asipto.com/gallery/v/siremis/ Relevant screenshots for new features: - charts: http://www.asipto.com/gallery/v/siremis/siremis_12.jpg.html - MI: http://www.asipto.com/gallery/v/siremis/siremis_15.jpg.html Demo site (it works on a database with random data, no MI; username: admin, password: admin): http://siremis.asipto.com/demo/ Download: http://siremis.asipto.com/pub/downloads/siremis/ Install/Upgrade documentation: http://siremis.asipto.com/install/ Regards, Ramona

3 3

[Kamailio-Users] how to work with Remote-Party-ID ?
by BERGANZ François 12 Feb '09

12 Feb '09

hello, i need to get the Remote-Party-ID in the header. I have that: Remote-Party-ID: "francois bgz" <sip:170725014@217.64.x.x:5067>;screen=yes;privacy=off;party=calling;-call-i nitiator=5016;-call-initiator-location=int;-redirected-by;-int-ext=5016;-ent -name=Acro;-direction=ext;-call-id=55665. I need to have -ent-name=Acro but, when I use $re, I have sip:170725014@217.64.x.x:5067 how can I do ? thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

3 2

Re: [Kamailio-Users] [Kamailio-Devel] SIREMIS v0.9.1 Released
by Elena-Ramona Modroiu 12 Feb '09

12 Feb '09

Hi Stefan, Stefan Sayer wrote: > Hi, > > o Elena-Ramona Modroiu [02/11/09 11:48]: > >> Hi everybody, >> >> a new version, 0.9.1, of SIREMIS - the web management interface for >> Kamailio (OpenSER) and SIP-Router.org - is released. >> > this looks very nice. > thanks! > Just recently I started with some monitoring support for SEMS > (http://lists.iptel.org/pipermail/semsdev/2009-February/003517.html) > The way this works is to push some call related info to the monitoring > module, and access it from the outside via DI functions, e.g. via XMLRPC > (http://svn.berlios.de/wsvn/sems/trunk/doc/Readme.monitoring). So you > can get e.g. a list of callids by executing the XMLRPC function > server.listActive(), and get the info for a call with server.get(callid). > > At the moment it logs ruri/from/to/application/direction in the core, > for conference application additionally the conference room. To > illustrate, you would get this from the server: > >>> s.list() > ['3c2d4bde0d60-gds81g00uz1n'] > >>> s.get(s.list()[0]) > [{'from': '"2(a)stefan-lap.office.iptego.de" > <sip:2@stefan-lap.office.iptego.de>', 'app': 'conference', 'to': > '<sip:35@stefan-lap.office.iptego.de;user=phone>', 'conf_id': '35', > 'ruri': 'sip:35@stefan-lap.office.iptego.de;user=phone', 'dir': 'in'}] > > This is of course extensible, e.g. codec used, call start/end time etc. > > Maybe, if we would add a way to monitor SEMS servers from within > SIREMIS, it would give a good opportunity to monitor e.g. conference > bridge, voicemail server, prepaid app, or b2bua (like sbc). > > What do you think? > I think it would be very useful. It looks to me more like an extension to communicate via XMLRPC with SEMS - send command and display the result. What is the XMLRPC you are using in SEMS? The C library used in Kamailio proved quite unstable, therefore SIREMIS uses MI UDP - simpler protocol and easy to troubleshoot. There should be XMLRPC PHP lib, therefore easy to extend SIREMIS. Is it possible to get statistic information out of SEMS? Like number of active calls? Such data can be used to build charts, which is nice to monitor the service health and evolution. Regards, Ramona

2 2

[Serusers] Forking Proxy
by Amanda Mattiuz 12 Feb '09

12 Feb '09

Hello all, Is is possible to configure SER to act as a forking proxy, where the INVITE messages that arrive at the server are forwarded to multiple clients ins parallel? I saw something about call hunt, but it seems that the INVITES are sent sequentially, could I make it become concurrently? Thanks, Amanda

2 1

[Kamailio-Users] IP SEC tunel
by ALAEDDINE abbech 11 Feb '09

11 Feb '09

hi, How can i do a tunel (use IP SEC)whith my serveur openser. is there a code to write to the configuration file to do this. thanks __________________________________________________________________________________________________ Ne pleurez pas si votre Webmail ferme ! Récupérez votre historique sur Yahoo! Mail ! http://fr.docs.yahoo.com/mail/transfert_mails.html

2 1

[Kamailio-Users] Kamilio Failover Options
by Stagg Shelton 11 Feb '09

11 Feb '09

I am looking for some ideas on how to create a more redundant setup in my operation. About 60 days ago our openser 1.3.2 system started sending out 500 errors. Looking through the log file I found alot of ERROR:tm:new_t: out of mem that corresponded to the INVITES. A simple restart of openser freed the memory and calls processed normally as expected. Since then we have upgraded to openser 1.3.3 and haven't experienced the problem since. Currently I am testing kamilio 1.4.3 and planning my migration into kamilio from openser. What I would like to create is a system that has two kamailio systems where one can fail to the other in the event of an application error that prevents calls from processing. Does anyone have any suggestions on implementing a redundant kamilio system, or can point me in the direction of where to start reading and researching. Thanks Stagg Shelton www.3pconnect.com

3 2

Re: [Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc
by Daniel-Constantin Mierla 11 Feb '09

11 Feb '09

Hello, On 02/11/2009 03:20 PM, Abdul Hakeem wrote: > Hi, > > Just wanted to touch base on the parser to find out if you are still > pursuing this. > it is still on my todo list, however didn't make it in the upcoming 1.5.0. Ovidiu Sas did some commits to the SDP parser and I know couple of others using it with success. Please open a feature request at: http://sourceforge.net/tracker/?atid=743023&group_id=139143&func=browse It is easier to track and not forgot about it for the next release. Thanks, Daniel > I am in the mind of purchasing some Cisco equipment to do the Sip-MAP > gateway from a GSM network. > I have someone else from the OpenBTS with me on this. > The object is to provide a full fledged sip based routing of voice, sms and > multimedia calls from gsm handsets to BTS with a Sip server backend. > > Cheers, > Abdul Hakeem > > -----Original Message----- > From: users-bounces(a)lists.kamailio.org > [mailto:users-bounces@lists.kamailio.org] On Behalf Of Daniel-Constantin > Mierla > Sent: 03 September 2008 17:45 > To: Kristian Kielhofner > Cc: users(a)lists.kamailio.org > Subject: Re: [Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc > > Hello, > > On 09/03/08 19:10, Kristian Kielhofner wrote: > >> Hello everyone, >> >> Feel free to set $OSSPROXY to whatever you like = >> Kamailio/OpenSIPS/OpenSER (for those of you who can't let go). >> >> $OSSPROXY is my favorite SIP toolkit. In addition to being an >> excellent proxy and registrar it serves as a platform to do SIP >> mangling and wrangling. With that in mind let me propose a crazy >> idea... >> >> How hard would it be to get $OSSPROXY to be able to parse what is >> commonly referred to as SIP-T (RFC 3372)? I understand $OSSPROXY has >> no business getting too involved in the SDP but I feel it makes sense >> in some instances (like this one). What if I want to make a routing >> decision based on an encapsulated ISUP parameter of some sort? Who >> knows what it could be!?!? Maybe there are some other uses but that >> makes the most sense and seems to be the most practical. It should be >> possible to develop an $OSSPROXY module to: >> >> - Use the SDP parser to find the ISUP SDP part: >> >> http://www.kamailio.net/dokuwiki/doku.php/development:sdp-parsing >> >> - Pass that part into a "real" ISUP parser (if necessary). I couldn't >> seem to find the perfect library but perhaps something like this could >> work: >> >> http://www.openss7.org/isup.html >> >> - Make the ISUP parameters available via AVPs, pseudo variables, etc >> for use in the $OSSPROXY script. >> >> I don't really have a need for this but it seems like it could be >> cool. Am I nuts? >> >> >> > I would expose the body as a dynamic structure, so components will be > accessible in the script, e.g. (naming schema just for example), > > if($body(0=>content-type)=="application/sdp") > { > if($body(0=>m=>type) == "audio") > { > } > } > > Based on the available content type parsers (sdp, isup), the structure has > different attributes. > > Cheers, > Daniel > > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] Looking for a highly available / load balanced setup
by John D 11 Feb '09

11 Feb '09

I currently have three Asterisk servers in my environment. None of them speak to each other just yet. The idea is to install Kamailio and be able to load balance/distribute SIP calls to any of the three asterisk boxes. If an entire box plug gets pulled, the operation continues. If a Kamailio box gets the plug pulled (and due to limited resources, I plan at first to run kamailio on all three of the asterisk boxes), the operation will continue. Does anyone have good documentation or insight in order to facilitate this sort of setup? Thanks.

4 6

[Kamailio-Users] New developer: Don Jackson
by Henning Westerholt 10 Feb '09

10 Feb '09

Hi all, its my pleasure to announce a new developer: Don Jackson. He'll maintain in the future the OpenBSD port of our favorite SIP server, and make sure that Kamailio will work there as good as on other operating systems. His sf.org id is 'dcjackson'. Thank you Don for the packaging you've done, i'm looking forward to work further with you. Best regards, Henning

1 0

[Kamailio-Users] OpenBSD port for Kamailio 1.4.3
by Don Jackson 10 Feb '09

10 Feb '09

Hello, I have updated my OpenBSD port to support Kamailio 1.4.3 I sent the updated port to the ports@openbsd mailing list: http://archives.neohapsis.com/archives/openbsd/2009-01/1107.html In addition, the port is available here: http://www.kamailio.org/pub/kamailio/1.4.3/packages/openbsd Thanks to Henning Westerholt for putting the files up on kamailio.org

1 0

Re: [Kamailio-Users] [Kamailio-Devel] FOSDEM2009 - first remarks/photos
by Daniel-Constantin Mierla 10 Feb '09

10 Feb '09

On 02/10/2009 04:44 PM, Iñaki Baz Castillo wrote: > 2009/2/10 Johansson Olle E <oej(a)edvina.net>: > >>> I have been to XMPP interop, although they focused on Voice/Jingle, I >>> tried the new purple module -- still some things to sort out and I will >>> post a resume about this particular case and entire event. >>> >> I spent monday on the XMPP dev meeting a got a few ideas for improvements. >> > > Nooo, forget XMPP!! > SIP SIMPLE already has enough cool specifications, the problem is that > nobody implements them! > :-) -- maybe SIP extension writers can learn from xmpp where they go first for prototype implementation and then to standardization. You are right, there are cool specification, but many do not map properly to reality. > Looking for interoperability with XMPP will not help, and I don't want > a future in which my softphone implements SIP for voice and XMPP for > presence/IM, that's a dirty workaround! > I think a competing open protocol is good all the time, makes things improve in both sides. Apart of that, sip and xmpp does not really match, SIP is more a routing-like protocol, while xmpp is a content-like protocol. XMPP network exist and cannot be ignored. A good usability of xmpp protocol will be communication between various points of your environment, sip servers, web servers, etc... Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 2

[Kamailio-Users] FOSDEM2009 - first remarks/photos
by Daniel-Constantin Mierla 10 Feb '09

10 Feb '09

Hello, thanks to Alexandr Dubovikov, a selection of photos from FOSDEM2009 and Kamailio/SIP Router Meeting are now available at: http://www.kamailio.org/events/2009-FOSDEM/FOSDEM2009-Album/ Some movies should become available as well. As for the FOSDEM itself, there were a lot of sessions focusing on scalability for backends, mainly from MySQL track. There are some goodies that will help improving performance with 0-lines of code in the sip servers. I have been to XMPP interop, although they focused on Voice/Jingle, I tried the new purple module -- still some things to sort out and I will post a resume about this particular case and entire event. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 1

[Kamailio-Users] dialog enhancements #3
by Daniel-Constantin Mierla 09 Feb '09

09 Feb '09

Hello, the dialog module for upcoming release kamailio/openser 1.5.0 (http://www.kamailio.org/dokuwiki/doku.php/features:new-in-1.5.x) has the ability to execute a script route when a dialog timeouts. Just set the $dlg_ctx(timeout_route) before creating the dialog. route { ... $dlg_ctx(timeout_route) = 11; dlg_manage(); ... } route[11] { ... } Before the execution of timeout route the $dlg(...) psudo-variable is set to the expired dialog. One particular use case is to write full cdr, via sqlops and using $dlg(...) pseudo-variable. All functions that can be executed in a request route are safe here - there is a faked SIP message given as parameter. There are couple of other new properties that can be set before dialog creation: - send BYE at timeout - to make that happen you have to set: $dlg_ctx(timeout_bye) = 1; - set an integer that can be used as a bitmask flag checking later: $dlg_ctx(flags) = ...; Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] errors installing on centos 5 64 bits
by Fabian Borot 06 Feb '09

06 Feb '09

Hello I am trying to install it on Centos 5 64 bits using this command: NICER=1 make prefix=/usr/local/kamailio-install-dir all and it fails with this message: make[1]: Entering directory `/usr/src/kamailio-1.4.3-notls/modules/db_mysql' Linking db_mysql.so /usr/bin/ld: skipping incompatible /usr/lib/libssl.so when searching for -lssl /usr/bin/ld: skipping incompatible /usr/lib/libssl.a when searching for -lssl /usr/bin/ld: cannot find -lssl collect2: ld returned 1 exit status make[1]: *** [db_mysql.so] Error 1 make[1]: Leaving directory `/usr/src/kamailio-1.4.3-notls/modules/db_mysql' make: *** [modules] Error 2 I think that somehow it is trying to use the lib for 32 bits or that my specific lib file is not what it should be or it is not on my system. I have the folders /usr/lib and usr/lib64 with its respective library files. Is this "libssl" related to openssl? I have these installed on my system already: [root@proxy1 lib64]# yum install openssl Setting up Install Process Parsing package install arguments Package openssl - 0.9.8b-10.el5_2.1.x86_64 is already installed. Package openssl - 0.9.8b-10.el5_2.1.i686 is already installed. Nothing to do [root@proxy1 lib64]# I also looked for "libssl" and this is what I see, it was only found in the folder for 32 bits libs if I am not mistaken. if that is the case how can i get the 64 bits equivalent? [root@proxy1 lib64]# find / -name "libssl.so" /usr/lib/libssl.so [root@proxy1 lib64]# thank you very much Fabian _________________________________________________________________ Windows Live™: E-mail. Chat. Share. Get more ways to connect. http://windowslive.com/online/hotmail?ocid=TXT_TAGLM_WL_HM_AE_Faster_022009

3 6

Re: [Kamailio-Users] nathelper module: function rtpproxy_stream2uac/uas
by Rash 06 Feb '09

06 Feb '09

Hi Daniel, thanks for the quick repsonse, you delivered the magic hint! Putting the count paramaeter into double quotes works. Thanks for your effort! Rash

2 1

[Kamailio-Users] nathelper module: function rtpproxy_stream2uac/uas
by Rash 06 Feb '09

06 Feb '09

Hello, I'm using latest version of Kamailio (OpenSER) from daily snapshot, as i wanted to test the rtpproxy_stream2uac function of the nathelper module in order to make announcements with rtpproxy. Integration of Kamailio and RTPProxy was successfull and I added the following line to the kamailio.cfg: rtpproxy_stream2uas("/var/rtpproxy/prompts/music_on_hold", -1); as described in the nathelper documentation ( http://www.kamailio.org/docs/modules/devel/nathelper.html#rtpproxy_stream2x… ) After adding this line to the appropriate section (onreply_route in my case) kamctl refuses to start.: kamailio: CRITICAL:core:yyerror: parse error in config file, line 414, column 29-30: syntax error kamailio: CRITICAL:core:yyerror: parse error in config file, line 414, column 30-31: bad arguments where line 414 corresponds to the line shown above. I can't find the error as the line is copied from nathelper module. Playing around with the parameters, didn't lead to anything. It would be nice if anyone could help or tryout the example from nathelper documentation and feedback. Best regards Rash

2 1

[Kamailio-Users] dialog enhancements #2
by Daniel-Constantin Mierla 06 Feb '09

06 Feb '09

Hello, another addition to the dialog module is the new pseudo-variable class $dlg(...) http://www.kamailio.org/dokuwiki/doku.php/pseudovariables:devel#dlg_attr With it you can get access to all attributes of current processed dialog, e.g., caller and callee URI, start time, flags... A typical use case is to write full cdr when the BYE comes over - the dialog stores the start time and you can get the end time via $Ts. You can either write the acc record at dialog ending or update the start acc event written when INVITE was processed - you have access to the call-id, from tag and to tag in the right order and one shot of sql_query() will do it. Cheers, Daniel

3 2

[Kamailio-Users] location table
by joy yue 05 Feb '09

05 Feb '09

Hi, Is there a limitation on the number of rows in location table? In my rig, whatever number of users I use (>2million users), I notice the number of users in location table is 343707, which is far less than the number of users in my test. I thought previously that location table saves all the users in my test. Also when I use large number of users (>2million), openser pops up an error: ERROR: registrar:update_contacts: invalid cseq for aor <xxxx>. Has anyone saw this before? I am using openser1.3.2. Thanks, -Joy

3 12

[Serusers] Error loading "trusted" table (permissions.so)
by caio 05 Feb '09

05 Feb '09

Hello all, Does anyone knows about this "permission.so" module error when try to load "trusted" table (mysql)? If "trusted" table is empty, there is no error, but whan at least there is 1 record, the error appears. for example: mysql> select * from trusted; +---------------+-------+--------------+ | src_ip | proto | from_pattern | +---------------+-------+--------------+ | 10.124.35.253 | any | ^sip:.*$ | +---------------+-------+--------------+ and when I try lo restart ser: Jan 15 15:34:45 delcolimba /usr/local/sbin/ser[703]: [ID 966737 daemon.info] permissions - initializing Jan 15 15:34:45 delcolimba /usr/local/sbin/ser[703]: [ID 414293 daemon.warning] Default allow file (/usr/local/etc/ser/permissions.allow) not found => empty rule set Jan 15 15:34:45 delcolimba /usr/local/sbin/ser[703]: [ID 844440 daemon.warning] Default deny file (/usr/local/etc/ser/permissions.deny) not found => empty rule set Jan 15 15:34:45 delcolimba /usr/local/sbin/ser[703]: [ID 895822 daemon.error] ERROR: permissions: trusted_reload(): Database problem Jan 15 15:34:45 delcolimba /usr/local/sbin/ser[703]: [ID 650101 daemon.crit] init_trusted(): Reload of trusted table failed Jan 15 15:34:45 delcolimba /usr/local/sbin/ser[703]: [ID 345573 daemon.error] Error while initializing allow_trusted function Jan 15 15:34:45 delcolimba /usr/local/sbin/ser[703]: [ID 660978 daemon.error] init_mod(): Error while initializing module permissions or when executing serctl fifo cmd: root@delcolimba /usr/local # sbin/serctl fifo trusted_dump 200 OK root@delcolimba /usr/local # sbin/serctl fifo trusted_reload 400 Trusted table reload failed root@delcolimba /usr/local # tail /var/adm/messages Jan 15 15:45:37 delcolimba /usr/local/sbin/ser[717]: [ID 895822 daemon.error] ERROR: permissions: trusted_reload(): Database problem Jan 15 15:45:37 delcolimba /usr/local/sbin/ser[717]: [ID 450600 daemon.error] ERROR: fifo_server: command (trusted_reload) processing failed Thank you, Caio

2 7

[Serusers] preload_udomain error
by caio 05 Feb '09

05 Feb '09

Hi all, Running ser0.9.7 on Solaris 10 (mysql 5.0.51 + serweb 0.9.6) Have this problem when I stop ser service, and then start it again (ser -E -D).. 0(0) preload_udomain: ERROR: bad callid record in table aliases 0(0) preload_udomain: ERROR: for username 579981, contact sip:579981@mydomain.com 0(0) preload_udomain: ERROR: skipping... And finally it dies with: 0(0) get_connection(): Connection not found in the pool Fallo de segmentaciÃ³n (core dumped) It happens with many records on aliases table and happens too with only one record: mysql> select * from aliases\G *************************** 1. row *************************** username: 579981 domain: 10.10.1.1 contact: sip:579981@mydomain.com received: NULL expires: 2009-02-05 11:45:21 q: 1.00 callid: The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything cseq: 42 last_modified: 2009-02-05 11:45:46 replicate: 0 state: 0 flags: 128 user_agent: SIP Express Router FIFO 1 row in set (0.00 sec) SER start normally if I delete all the records on this table. But I need create the aliases again, and this is not the normal behaviour..I think. Here the description of aliases table: mysql> desc aliases; +---------------+---------------------+------+-----+---------------------+-------+ | Field | Type | Null | Key | Default | Extra | +---------------+---------------------+------+-----+---------------------+-------+ | username | varchar(64) | NO | PRI | | | | domain | varchar(128) | NO | PRI | | | | contact | varchar(255) | NO | PRI | | | | received | varchar(255) | YES | | NULL | | | expires | datetime | NO | | 2020-05-28 21:32:15 | | | q | float(10,2) | NO | | 1.00 | | | callid | varchar(255) | NO | | Default-Call-ID | | | cseq | int(11) | NO | | 42 | | | last_modified | timestamp | NO | | CURRENT_TIMESTAMP | | | replicate | int(10) unsigned | NO | | 0 | | | state | tinyint(1) unsigned | NO | | 0 | | | flags | int(11) | NO | | 0 | | | user_agent | varchar(50) | NO | | | | +---------------+---------------------+------+-----+---------------------+-------+ 13 rows in set (0.01 sec) Important: when "location" table keeps some records, ser doesn't start too: mysql> select * from location\G *************************** 1. row *************************** username: 579981 domain: mydomain.com contact: sip:579981@10.10.60.115:5060 received: NULL expires: 2009-02-05 11:46:16 q: -1.00 callid: 1f46c06d5e034425286f095b11cf4a1a(a)10.10.60.115 cseq: 6 last_modified: 2009-02-05 11:45:46 replicate: 0 state: 0 flags: 0 user_agent: CM5K-TA2S (807100.02) 1 row in set (0.00 sec) Here too I must to delete all the records to let SER start successfully. Does anyone knows about this issue? Regards, Claudio

1 1

[Serusers] configure SER and SEMS to work together
by Martin Lane 05 Feb '09

05 Feb '09

I am following the 'mini-Howto on how to configure SER and SEMS to work together' guide on iptel.org. So I have a separate directory for integrating the SER and SEMS functionality. For development purposes is there a switch to change the path were SER looks for the modules? As well control which version of SEMS is being used? I've installed Sip Express Media Server (1.1.0-rc1 (i386/linux)) as well as the version from the howto for testing. My basic problem is I currently get a version mismatch error: 0(9590) ERROR: module version mismatch for /usr/local/lib/ser/modules/sl.so; core: ser 0.9.6-sems (i386/linux); module: ser 0.9.6 (i386/linux) The appropriate modules are located in under my ser-sems test dir and I have not been successful in redirecting SER to look there. Also, I have modified ser.cfg and sems.cfg for my system but I did not find a default ser_sems.cfg. Is this file necessary and where is the default location? Thanks for any help or advice. mll

2 1

[Kamailio-Users] dialog enhancements #1
by Daniel-Constantin Mierla 04 Feb '09

04 Feb '09

Hello, during the week before freezing the code for release kamailio (openser) 1.5.0, I have added several enhancements to the dialog module. I am going to present them in several emails, as I get the documentation properly updated. There is now a new function called dlg_bridge(From, To, outbound_proxy). It can be used to connect From and To users. The mechanism is a simple call transfer scenario. kamailio => INVITE (hold) =>From kamailio <= 200ok <= From kamailio => REFER (To) =>From kamailio <= 200ok <= From ... From => INVITE ... =>To ... kamailio => BYE =>From kamailio <= 200ok <= From The function can be called from configuration file or via MI interface. It opens the door for several cute features you can do with the sip server. Here are couple of usage examples: - easy integration with web/external applications for Click-To-Dial. Doing it with kamctl will look loke: kamctl fifo dlg_bridge sip:from@kamailio.org sip:to@kamailio.org - wake up/reminder/scheduled call - use rtimer module to run scheduled calls - you can connect your user to a media server playing a message, to a conference room or where the user sets its preference - call again - when a user is calling a busy destination, when that destination becomes available, the server can initiate the call to connect the caller with the destination Documentation is available at: http://kamailio.org/docs/modules/1.5.x/dialog.html Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 4

[Kamailio-Users] Reg. SIP conferencing support for RFC 4579
by Padmaja R 04 Feb '09

04 Feb '09

Hi all, Is it possible to use Openser as the SIP conference server? I understand that Openser cannot do the media mixing required of a centralized media server but if it can do the centralized signaling by acting as a focus and let a media proxy do the media mixing, that would be good. I need ad hoc conferencing capability. Is there any media proxy that can achieve this together with openser? My scenario is as follows:- Say A and B are in a SIP call, and A and B suddenly decide to have a conference with C. A puts B on hold and dials C' s number. After call between A and C is up, if A presses hook flash, this sends an invite to the conference application namely: conf@openseratlocalhost:5060 (which is the ip and port number at which my openser proxy would be listening). The server should then send back 200 Ok with the contact header containing the isfocus" parameter of the form -Contact: <sip:conference1@mediaproxyip;isfocus> The isfocus is an RFC 4579 based feature. The 200 Ok responses to other invites (normal calls that do not create conference) should not include this isfocus in the contact header of 200 Ok. Finally, the openser proxy should be able to contact the media proxy on receiving an invite to create the conference and send that ip and focus information in the 200 Ok. The UA -A would then send a REFER to B and C to refer to the media server and they disconnect with A. Then all three UAs would individually send invite to the uri returned in the contact header and join the conference. Thanks, Padmaja

4 4

[Serusers] SER and media server
by Chatrapathi. H. Rahul 04 Feb '09

04 Feb '09

Hi I am using solaris 9v. I tried to use SEMS as media server with SER. SEMS is not compatible with Solaris 9v. Gives compilation errors. My query is whether any other opensource Media Server is available that uses SER and compatible with solaris 9v.This would help me a lot. Thanks in advance. -- Regards Rahul

1 0

[Kamailio-Users] FOSDEM Meeting - Feb 7, 2009, 19:30, Le Mess
by Daniel-Constantin Mierla 04 Feb '09

04 Feb '09

Hello, the place for the social networking event at FOSDEM 2009 has been selected: Le Mess http://www2.resto.be/mess/home-en.htm Louis Schmidtlaan - Av. Louis Schmidt 1 1040 BRUXELLES (ETTERBEEK) It is in walking distance from FOSDEM site. More details at: http://www.kamailio.org/mos/view/News/NewsItem/FOSDEM-2009---Social-Network… If you haven't confirmed your participation, you can still join, subject of available free places in the restaurant. See you Saturday! Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] New module : PURPLE - IM/Presence multi-protocol gateway
by Eric PTAK 04 Feb '09

04 Feb '09

Hello everyone, Let me introduce to you a new module, available in trunk since yesterday, so it will be included in 1.5.0 release : >From http://kamailio.org/docs/modules/devel/purple.html: *Purple module is a multi-protocol instant-messaging gateway module which use libpurple, the library on bottom of Pidgin (ex-gaim).* *This plugin was wrote in order to proof the concept of gateway between SIP and MSN/Live networks.* *[...]* *It's no really adapted to a production environment with a huge user count but can be used in order to demonstrate the concept or to use it in SOHO environment.* I hope you'll enjoy purple module, and I'll welcome all feedbacks. Eric.

4 12

Re: [Kamailio-Users] sip trace & duplicate_uri
by Daniel-Constantin Mierla 03 Feb '09

03 Feb '09

On 02/02/2009 09:35 PM, kokoska rokoska wrote: > > [...] >>> Hello, >>> >>> i have increased debug to 6 and I can not see errors in syslog. >>> Messages are traced, stored into DB, but not duplicated. >>> I'm at 1.4.3 on CentOS 5 - if it makes sense... >>> >>> Thank you very much, Daniel, for your interest! >>> >>> >>> >> ok, checked the source code. For some reason I do not recall now, when I >> wrote the module the duplication is done only when traced_user_avp is set. >> http://kamailio.org/docs/modules/devel/siptrace.html#id2452997 >> >> Set that avp and try again. If works I will try to fix and make it work >> without avp. >> >> Thanks, >> Daniel >> >> >> > > > Thank you very much, Daniel, for your help! > > It works :-) > > BTW: My idea is to offload proxy from (blocking ?) DB operatinos and > still have all possible logs => I think I can duplicate all messages to > another kamailio server and store them there to another DB. Is it > doable? With sip trace you still get them in a DB, because it first stores to DB and then sends to another address. However, what you want to achieve is something good, so for the future I will have in mind the option just to replicate to another address and skip writing to db. > Or, better to say, can I trace and duplicate messages on one > server and not log them? > If you want to duplicate the messages that come to kamailio, you can use the send() function in route and onreply_route block. http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:devel#send_destinat… If you want the siptrace module to do it, you have to hack it a bit. Practically is to introduce a new parameter, e.g., skip_db_storage, and put the code that writes in db inside a if not skip_db_storage. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 4

Re: [Kamailio-Users] [Kamailio-Devel] New module : PURPLE - IM/Presence multi-protocol gateway
by Eric PTAK 03 Feb '09

03 Feb '09

I agree with you, the module needs improvements on that point. It's a really old point I missed when we decided to reuse and to reverse that module. I think about two manners in order to solve it : * implement a return code for purple_send_message * sending a reply directly in the module Regarding the design of the module, I think both possibilities needs a lot of mod. I'm not sure how to consider that mod as it's not a bug but a limitation. Is it a new feature I'll code when the trunk will be unfreezed ? If yes, we may add that limitation in the doc for now. Eric. 2009/1/29 Iñaki Baz Castillo ibc(a)aliax.net > Why is needed to reply "200"? doesn't this module act as a proxy > waiting for a response from downstread (i.e. MSN network) and mapping > that response to the appropiate SIP code? > > This is: if I send a MESSAGE to a MSN account not online (or not > existing) I will receive "200" anyway, shouldn't I receive a 480 or > 404? > > Regards. > > > -- > Iñaki Baz Castillo > <ibc(a)aliax.net> > > _______________________________________________ > Kamailio (OpenSER) - Devel mailing list > Devel(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/devel > http://lists.openser-project.org/cgi-bin/mailman/listinfo/devel

4 7

[Kamailio-Users] wiki documentation index for kamailio
by Henning Westerholt 03 Feb '09

03 Feb '09

Hello all, thanks to Frederick Bullik we've now a nice documentation index for the Kamailio 1.4.x and 1.5.x releases available. You can easily lookup all module exports like functions [1] or module parameters [2]. The index is sorted alphabetically and provides convinient links to jump directly to the respective place in the module documentation. You'll find the new index in our wiki at: http://kamailio.org/dokuwiki/, as well as the old ones for past OpenSER releases. Best regards, Henning Westerholt [1] http://www.kamailio.org/dokuwiki/doku.php/modules:1.5.x:index-functions [2] http://www.kamailio.org/dokuwiki/doku.php/modules:1.4.x:index-parameters -- Henning Westerholt - Development Consumer Products / DSL Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany I'm going to FOSDEM

2 1

[Kamailio-Users] [Fwd: Re: sip trace & duplicate_uri]
by kokoska rokoska 02 Feb '09

02 Feb '09

Daniel-Constantin Mierla napsal(a): > Hello, > > On 02/02/2009 08:08 PM, kokoska rokoska wrote: >> >> Daniel-Constantin Mierla napsal(a): >> >>> Hello, >>> >>> do you get any error in the syslog? >>> >>> Cheers, >>> Daniel >>> >>> >> >> >> Hello, >> >> i have increased debug to 6 and I can not see errors in syslog. >> Messages are traced, stored into DB, but not duplicated. >> I'm at 1.4.3 on CentOS 5 - if it makes sense... >> >> Thank you very much, Daniel, for your interest! >> >> > > ok, checked the source code. For some reason I do not recall now, when I > wrote the module the duplication is done only when traced_user_avp is set. > http://kamailio.org/docs/modules/devel/siptrace.html#id2452997 > > Set that avp and try again. If works I will try to fix and make it work > without avp. > > Thanks, > Daniel > > Thank you very much, Daniel, for your help! It works :-) BTW: My idea is to offload proxy from (blocking ?) DB operatinos and still have all possible logs => I think I can duplicate all messages to another kamailio server and store them there to another DB. Is it doable? Or, better to say, can I trace and duplicate messages on one server and not log them? Best regards, kokoska.rokoska

1 0

[Kamailio-Users] sip trace & duplicate_uri
by kokoska rokoska 02 Feb '09

02 Feb '09

Hi all, could someone be so kind and explain me how to use (or how exactly works) duplicate_uri in siptrace module? I think, if I setup modparam("siptrace","duplicate_uri","sip:1.2.3.4:5060") that all (and what is it "all"? :-) traced messages will be copied to defined sip uri (1.2.3.4:5060). I see nothing "on the wire" with ngrep, but all messages are succesufly inserted into local DB so I thing the siptrace module works. But doesn't copy messages to 1.2.3.4:5060 :-) Any hitns are really welcome! Best regards, kokoska.rokoska

2 3

[Kamailio-Users] store a double in avp
by ALAEDDINE abbech 02 Feb '09

02 Feb '09

hi all, can i store a double in avp? i try to store a double from my DB in the avp, so i have alwyse null. who have a solution for this pb. thanks

2 2

[Kamailio-Users] FOSDEM 2009 - Kamailio (OpenSER)/SIP-Router Social Networking Event
by Daniel-Constantin Mierla 02 Feb '09

02 Feb '09

Hello, a bunch of us will be at FOSDEM 2009: http://fosdem.org/2009/ Therefore is the opportunity for first Social Networking Event in 2009. Everyone interested in Kamailio (OpenSER), SIP Router, SER, SIP and not only are invited to join -- participation is free, everybody pays for its expenses. The event is very likely to be a dinner on Saturday evening, Feb 7, 2009. So far, from Kamailio (OpenSER) community you can meet there: - Henning Westerholt - http://www.kamailio.org/mos/view/Management/ - Elena-Ramona Modroiu - http://www.kamailio.org/mos/view/Management/ - Daniel-Constantin Mierla - http://www.kamailio.org/mos/view/Management/ - Dan Bogos - http://itsyscom.com/ In order to know properly the size of the place we need, please drop me an email if you intend to attend. People knowing Brussels are welcome to suggest a nice place to meet (good beer and food are a must). Looking forward to meeting you in Brussels. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 1

[Kamailio-Users] No UDP Address already in use?
by mayamatakeshi 31 Jan '09

31 Jan '09

Hello, If I start a second instance of kamailio by mistake, if the cfg file instructs it to listen to tcp ports, the second instance will exit with something like this: tcp_init: bind(7, 0x81b28ec, 16) on 127.0.0.1:5060 : Address already in use But if the cfg file only instructs kamailio to bind to udp ports, the second instance will not exit. Is this a bug? I'm curious because if we start kamailio and a udp port is in use by another application (not kamailio), kamailio will exit with udp_init: bind(4, 0x819f274, 16) on 192.168.88.29: Address already in use Why is that so? regards, mayama

3 4

[Serusers] Modified "To" header on REGISTER
by caio 30 Jan '09

30 Jan '09

Hello, Does save() method of registrar module use original sip message or can it use a modified one like for example by the subst() method of textops module? For example: A coming "To" header of the a REGISTER is modified and then the save() is called: Code: # REGISTER subst_uri('/^(.*)sip:(ip_address)(.*)$/\1sip:mydomain\3/i'); subst('/^To:(.*)sip:(.*)@(ip_address)(.*)$/To:\1sip:\2\@mydomain\4/i'); save("location"); Log: Jan 12 10:09:34 [/usr/sbin/ser] subst_run: matched (0, 17): [sip:200.2.127.187]_ Jan 12 10:09:34 [/usr/sbin/ser] textops: subst_uri_f: match - old uri= [sip:ip_address], new uri= [sip:mydomain]_ Jan 12 10:09:34 [/usr/sbin/ser] subst_run: running. r=0_ Jan 12 10:09:34 [/usr/sbin/ser] subst_run: matched (172, 40): [To: "579997" <sip:579997@ip_address>_]_ Jan 12 10:09:34 [/usr/sbin/ser] textops: subst_f: replacing at offset 208 [To: "579997" <sip:579997@ip_address>_] with [To: "579997" <sip:579997@mydomain>_]_ Cheers.. Caio

1 1

[Kamailio-Users] STUN and TM
by Jason Penton 30 Jan '09

30 Jan '09

Hi *, What is the logic behind the transation matching in Kamailio. I am finding a problem whereby a call setup transaction (INVITE, 180 Ringing, PRACK, 200OK, ACK). between 2 Snom phones works in normal mode. However, when one phone is behind a NAT with STUN support, everything works perfectly (i.e. call setup, 2-way audio, etc), BUT the tm module doesnt match the transaction between the original invite and the subsequent messages (180, 200ok, etc). the impact here is that the fr_inv_timer eventually fires even though the call has been answered. Note: this only happens when the call is destined for the device behind the NAT. Any ideas. Perhaps an explanation on how the tm module does transaction manager. RFC3261 states that transaction matching is done based on the via branch parameter. Thanks in advance Jason

2 7

[Kamailio-Users] code frozen for 1.5.0
by Daniel-Constantin Mierla 29 Jan '09

29 Jan '09

Hello, starting with today, the release cycle for Kamailio (OpenSER) v1.5.0 enters in testing phase. Like in the past, this mean no new features will be added for a while, the focus is on testing, bug fixes and documentation improvements. I can say, being here from day one, this upcoming release had a shorter development period, but it is the one with the largest new features set. See a summary here: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-1.5.x A special care was given to code cleanup and core. The core got slimmer - lot of config-related features were moved to modules, therefore now it is less exposed to bugs and more performant. From the project's environment: - we got three new developers - a brand new web management interface - http://siremis.asipto.com - next week will be a meeting in Brussels, at Fosdem with many developers showing up there: http://www.kamailio.org/mos/view/News/NewsItem/FOSDEM-2009---Social-Network… Worth to mention that this is the last release based on current core, next ones will use the SIP-Router.org core framework: http://sip-router.org, that will make possible to combine all the features from Kamailio (OpenSER), SIP Express Router (SER) and OpenIMSCore projects in a single application, in a collaboration environment sustained by the developers from all three projects. First I want to thank to all developers for the work and time dedicated to develop version 1.5.0 and to our community members for valuable feedback and testing. That being said, I invite everyone to test and report any issue found. To get the source code from SVN use: svn co https://openser.svn.sourceforge.net/svnroot/openser/trunk kamailio Please report to devel(a)lists.kamailio.org or project's tracker: http://sourceforge.net/tracker/?group_id=139143 Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] textops: new function - set_rpl_body(...)
by Daniel-Constantin Mierla 29 Jan '09

29 Jan '09

Hello, few days ago I added a new function to textops module: - set_rpl_body(body, content-type) It can be used to set the body of a reply generated by kamailio/openser. route { if(is_method("HELLO") && $ru == "sip:server@kamailio.org") { set_rpl_body("Hi $fn!", "text/plain"); send_reply("200", "ok"); exit; } The insert_body() function (also new in 1.5.0) has been renamed to set_body() for better coherence. This one can be used to set the body of current processed sip message. An an example, set_rpl_body() can be used for pull-like data transfer between servers. Using uac module (and eventually rtimer module), servers in a farm can interrogate each other and exchange data. Some useful links: http://kamailio.org/docs/modules/1.5.x/uac.html#id2467913 http://kamailio.org/docs/modules/1.5.x/rtimer.html http://kamailio.org/docs/modules/1.5.x/textops.html#id2468015 Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Serusers] SER Consultants in US
by Tichner, Jeromy E (US SSA) 29 Jan '09

29 Jan '09

Hello, We're looking for some phone support from a consulting firm for a SER configuration. Looking for an hourly estimate and must be available ASAP. Must be a US citizen. Regards, Jeromy

1 0

[Kamailio-Users] ldap in kamailio
by Husna Bte Mad Baguri 29 Jan '09

29 Jan '09

Help, How to install ldap module in kamailio Regards Husna ________________________________ This e-mail and the attachments are intended solely for the person to whom it has been addressed. It contains privileged and/or confidential information and is privileged or otherwise protected from disclosure. If you are not the person for whom this e-mail was intended, or the e-mail has reached you by mistake, please delete it immediately and inform us of the error. Our e-mail address is administrator(a)uniten.edu.my. All opinions, conclusions and other information in this message that do not relate to the official business of Universiti Tenaga Nasional (UNITEN) shall be understood as neither given nor endorsed by UNITEN. UNITEN shall not be responsible for any activity that may be considered as illegal and/or improper use of e-mail and UNITEN further is claims and shall not accept liability for any content of this e-mail, or for the consequences of any actions taken on the basis of the information provided, unless that information is subsequently confirmed in writing. WARNING Internet communications cannot be guaranteed to be secured or error-free as information could be intercepted, corrupted, lost, arrive late or contain viruses. As such, we do not accept liability for any errors or omissions in the content of this message which may arise as a result of internet transmission. UNITEN does not authorize any of its employees to make any defamatory or seditious statements or commit any offence which is contrary to the laws of Malaysia. Any such communications and/or actions by such employees are outside the scope of employment of the said individuals and UNITEN shall not be liable for such communications and/or actions.

3 4

[Kamailio-Users] detecting the presence of a header
by Daniel Corbe 28 Jan '09

28 Jan '09

I need to quickly detect the presence of the Alert-Info header and do special processing if it is present. Is there facilities to do this currently? -Daniel

2 1

[Kamailio-Users] function to send ACK message
by Alex R.S.M 28 Jan '09

28 Jan '09

Hi, I need kamailio proxy to reply with an ACK to 200 OK, i.e. the proxy stays in the call: UAC sends INVITE to kamailio, kamailio forward it to UAS UAS sends 180 to kamailio, kamailio forward it to UAC UAS sends 200 to kamailio, kamailio forward it to UAC AND sends ACK to UAS I was wondering what function I can use so kamailio can generate an ACK message to response to 200 OK. Apparently both t_reply () and sl_send_reply() functions create responses and not an ACK message. Thanks, Alex _________________________________________________________________ Windows Live™ Hotmail®…more than just e-mail. http://windowslive.com/howitworks?ocid=TXT_TAGLM_WL_t2_hm_justgotbetter_how…

2 2

[Kamailio-Users] new user
by Andy Nedelcu 28 Jan '09

28 Jan '09

Hey, I am new at working with SIP Servers and I was wondering if Kamailio has the same features as Asterisk. I have bought a PCI card for connecting to the phone line and I want to know how can I make Kamailio aware of this card or a simple phone modem. I use FreeBSD and the only version that I found was OpenSer 1.2. Because I love being me and I love loving her, SC BMS SRL - Departamentul IT - Nedelcu Andrei

3 2

[Kamailio-Users] Database error: failed to request mysql lock ()
by Brian Del Shasta 27 Jan '09

27 Jan '09

Database error: failed to request mysql lock () Has anyone else experienced this issue?

4 3

[Kamailio-Users] Silently replying 500 ?
by Aurelien Grimaud 27 Jan '09

27 Jan '09

Hi, I have a route block which issues a t_relay(). route[1] { t_relay() ; } If this t_relay fails, is there a default reply from openser to caller ? Let me explain. In my logs, I have an incoming INVITE, followed 2 ms after t_relay by an ACK from uac. The callee never gets the INVITE. UAC -- INVITE --> openser : t_relay() UAC -- ACK --> openser I guess UAC gets a default reply from openser, I want to make sure of it. I have no ERROR in my log file. UAC logs tells that openser hang up with a server internal error (no sip trace, no sip code, just an indication of what cause could be). This is openser 1.2.0 Is anyone aware f this issue ? Could someone point me where the 5xx reply is in source code ? If it is a default reply, is there now an ERROR message before reply ? Thanks, Aurelien

2 3

[Kamailio-Users] openSER concept question
by Alex R.S.M 27 Jan '09

27 Jan '09

Hi, In a SIP environment with SIP end points (UAC, UAS) and several SIP proxy hubs including openSER, is each hub like openSER supposed to reply with an ACK to 200 OK and BYE messages coming from end points? (i.e. openSER should stay on the call) Or each hub supposed to only propagate all messages including ACKs ? Thanks, Alex _________________________________________________________________ Windows Live™ Hotmail®…more than just e-mail. http://windowslive.com/howitworks?ocid=TXT_TAGLM_WL_t2_hm_justgotbetter_how…

2 1

[Kamailio-Users] error, please help me
by Daniel Kemayou 26 Jan '09

26 Jan '09

Hello all I have a error when i finish the setup of SIREMIS, i have window with > userid and password when i enter the userid and password i have this > error > > [2009-01-24 19:11:48 (GMT)] An exception occurred while executing this > script: > Error message: #0, The mysql driver is not currently installed Script > name and line number of error: > /var/www/siremis-0.9.0/openbiz/others/zendfrwk/library/Zend/Db/Adapter > /Pdo/Abstract.php:104 > > can you help me please DISCLAIMER: This email and any attachments are confidential and are intended solely for the addressee. If you are not the addressee tell the sender immediately and destroy it. Do not open, read, copy, disclose, use or store it in any way, or permit others to do so. Emails are not secure and may suffer errors, viruses, delay, interception, and amendment. Creolink Communications and its subsidiaries do not accept liability for damage caused by this email and may monitor email traffic. DENI : Ce mail, y compris toute piece jointe, est confidentiel. Il est exclusivement adresse aux destinataires designes. Si vous le recevez par erreur, veuillez immediatement aviser l'expediteur et detruisez-le sans l'ouvrir ni le lire. Toute copie, diffusion, conservation, utilisation sous quelque forme que ce soit ou l'autorisation de toute personne a le faire est interdite. Le mail n'est pas un moyen sur et peut contenir des erreurs ou des virus, comme il peut etre retarde, intercepte ou modifie. Creolink Communications et ses filiales declinent toute responsabilite pour des dommages causes par ce mail et se reservent le droit de controler la circulation des mails.

2 1

[Kamailio-Users] carrierroute
by michel freiha 26 Jan '09

26 Jan '09

Hi all, I have a problem with carrierroute module...I have configured it as follow in my config file: loadmodule "carrierroute.so" ######Carrier Route modparam("carrierroute", "db_url", "mysql://opensips:123456@MYSQL_IP /opensips") modparam("carrierroute", "config_source", "db") In the route function i have created the following function: if (!cr_route("default", "0", "$rU", "$rU", "call_id")) { sl_send_reply("403", "Not allowed"); } else { # In cas of failure, re-route the request t_on_failure("1"); # Relay the request to the gateway t_relay(); } When trying to restart kamailio server I got a the below error: Jan 24 11:10:22 [12841] DBG:core:fix_actions: fixing sl_reply_error, line 381 Jan 24 11:10:22 [12841] DBG:core:fix_actions: fixing cr_route, line 390 Jan 24 11:10:22 [12841] ERROR:core:pv_parse_spec: bad parameters Jan 24 11:10:22 [12841] INFO:carrierroute:carrier_fixup: carrier tree default has id 3 Regards

2 3

[Kamailio-Users] DNS unreachable
by Zappasodi Daniele 26 Jan '09

26 Jan '09

Hello, I'm testing the proxy behaviour (Openser 1.3.2) when DNS servers are unreachable and I observe that with few requests that require the DNS, I have the proxy completely blocked, while it could serve all the other methods that don't need the DNS (method with IP adddress or incoming REGISTERs). I observe that each child process remains suspended until all DNS requests are finished (NAPTR, SRV and A). If I configure two DNS server and X retransmissions after Y seconds, the child remain suspended for: 2 * X * Y * 3 seconds (3=NAPTR+SRV+A queries) Even if I set low values for X and Y (X=2 retr and Y=2 sec.) the child is suspended for a long interval: 2*2*2*3=24 seconds. Is it possible to avoid that chlidren remain suspended? Is there any configuration param that can reduce this interval? Is it possibile to introduce a little change in sip_resolvehost (check errno) in order to avoid the SRV and A queries if the NAPTR fails because the DNS servers are unreachable? ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message inerror. **********************************************************************

2 1

[Kamailio-Users] Doubt about nat=yes
by mayamatakeshi 25 Jan '09

25 Jan '09

Hello, I'm trying to understand this sample cfg: http://voip-info.org/wiki/view/OpenSER+And+RTPProxy This cfg is used as foundation for some of our cfg files here but nobody is sure about exactly how some things work. So I'm trying to clear things up. In particular, I am clueless about why there is code removing "nat=yes" from the URI and adding it to the Contact header (actually I don't know what is its meaning. I suppose this is a way of a client advertising that it is behind NAT, but I couldn't find which RFC defines this): subst_uri('/(sip:.*);nat=yes/\1/') search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); Is this something that always must be done when dealing with nat or it was a particular situation that the writer of the cfg had to workaround? I'm rewriting our cfg file (to be used with m4) and I'm trying to remove unnecessary things. Another thing in the cfg: in case a reinvite is issued, should not unforce_rtp_proxy be called before force_rtp_proxy is called again? Could this lead to having rtpproxy leaking resources? <http://www.kamailio.org/docs/modules/1.3.x/nathelper.html#AEN367> regards, mayama

2 2

[Kamailio-Users] Openser-Asterisk Codec conversion..
by Rawshan Iajdani 24 Jan '09

24 Jan '09

My scenario is like that.. UA----->OpenSer(Outbound Proxy)---------Register Server | | | | Asterisk(codec converion)---------------------- The UA will register to Register server through outbound proxy OpenSer. When UA makes call it first comes to Openser, OpenSer should route the media to Register server through Asterisk for codec conversion. OpenSer will not hold any User account rather it will act as a proxy. The way I am thinking.. Openser could be used as outbound proxy that's fine. If it were used as outbound proxy, can it also route the media? In the asterisk side I may need username/password pair of UA to dial to Register Server in the second leg. Can I read the User/Pasword from Openser after UA is registered to OpenSer. I can have both OpenSer and Asterisk to reside on the same machine with different IPs so they can access the database if required..

4 6

[Kamailio-Users] CarrierRoute With + Prefix
by Ali Jawad 23 Jan '09

23 Jan '09

Hi All I was able to setup carrierroute in database mode and when I make a call with prefix 00 it works fine. For that I did setup an entry in carrierroute, I did duplicate that same entry for prefix "+". If I call 004353453 it works fine however if I dial +12341234 I get a 403 forbidden error. The log shows: Jan 23 14:22:34 [25866] DBG:carrierroute:get_route_tree_by_id: tree default, domain 0 : 0 Jan 23 14:22:34 [25866] INFO:carrierroute:rewrite_uri_recursor: URI or route tree nodes empty, empty flag list Jan 23 14:22:34 [25866] INFO:carrierroute:cr_do_route: rewrite_uri_recursor doesn't complete, uri +9613041705, carrier 5, domain 0 Any guidance please ? With Regards

3 8

[Kamailio-Users] [presence] use case of a user that doesn't un-REGISTER and un-PUBLISH
by Pascal Maugeri 23 Jan '09

23 Jan '09

Hi Using kamailio v1.5 (trunk), I'm testing the following simple presence scenario where one user shutdowns his application without prior un-PUBLISH and un-REGISTER. 1) user A subscribes to user B presence. Both use X-Lite. 2) user B shutdowns his X-Lite (without doing un-REGISTER and un-PUBLISH of his presence information), 3) user B starts again X-Lite, registers and publishes his presence, from that point when user B updates his presence information, user A does not see the updates ! I've configured kamailio to use save("location", "0x04") instead of save("location") for incoming REGISTER but this does not help. Is there a way to clean presence DB tables properly when user B register again ? Regards, Pascal

2 9

[Kamailio-Users] Carrieroute help
by Ali Jawad 23 Jan '09

23 Jan '09

Hi All I was finally able to setup carrierroute on Solaris :). I am using carrierroute in db mode and I did setup two entries with rewritehost in carrierroute and two entries in route_tree with one route being default. I am still very new to carrierroute as a sample start I do have two astersisk servers that will act as pstn_gw and I want to split the traffic in half between both servers. Can anyone please provide sample instructions on what to put into the route{} statement in this scenario and any main ideas about route_tree and carrierroute ? Thanks.

2 7

[Kamailio-Users] Siremis configuration error
by Juan Asencio 22 Jan '09

22 Jan '09

Hi Ramona, I have installed Siremis, but I get the following error: Unable to write in the session save path [/home/www/siremis/siremis/session] Fatal error: Call to a member function GetSessionContext() on a non-object in /home/www/siremis/openbiz/bin/BizSystem.php on line 9 Do you have any suggestions about how to fix this? Thanks in advance, Juan.-

2 1

[Kamailio-Users] Carrierroute and AVP
by rod 22 Jan '09

22 Jan '09

Hi, I'd like to know if there is a way to retrieve the information from the carrierroute module in AVPs. Example: - a prefix has matched, and I'd like to get the field "description" associated with this route and add it in the SIP header - a prefix has matched, and I'd like to get the field "rewrite_host" associated with this route and use it with xlog Thanks, rod.

3 6

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users