- sr-users - kamailio.org

Writing information into a table
by John Doe 15 Aug '13

15 Aug '13

Hi A client of ours needs to insert information into a table at some point into the call. I am in betwwen these 2 options: 1- Using http_query function from the UTILS module to call a web service that will insert the information passed in the URL in to the table 2- or using the sql_query function from the SQLOPS module. My priority is to use the fastest one. Since both are TCP they kind of come with some reliability regarding re-tries, etc (vs using a method based on UDP) I am king of leaning versus the option #1, b/c I am worried about the cost of creating a db connection everytime etc but we dont have experience on none of the methods. Also please take into account that they are still using version 1.52, and upgrading is not an option at the moment. Thank you very much, your answer is much appreciated! john

1 0

Kamailio 3.2 -> 3.3.5 Migration
by Brandon Armstead 15 Aug '13

15 Aug '13

Hello, Just wanted to shoot a quick email to the user list and see if anyone had any 'gotchas' to keep in mind when migrating from 3.2 to 3.3.5. I know table structure has changed and will need to be updated, but are there otherwise any syntax and or major configuration differences? Thank you for your time in advance. Sincerely, Brandon Armstead

1 0

Kamailio v4.0.3 Released
by Daniel-Constantin Mierla 15 Aug '13

15 Aug '13

Hello, Kamailio SIP Server v4.0.3 stable release is out. This is a maintenance release of the latest stable branch, 4.0, that includes fixes since release of v4.0.0. There is no change to database schema or configuration language structure that you have to do on installations of v4.0.0. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.0.3. For more details about version 4.0.3 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2013/08/kamailio-v4-0-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

How does rtpproxy handle handover?
by Khoa Pham 15 Aug '13

15 Aug '13

I have SIP proxy (Kamailio) works in conjunction with rtpproxy<http://www.rtpproxy.org/> to support client communication. When SIP proxy sends command to rtpproxy to create new session, rtpproxy will create 2 ports (let's called them port1 and port2). rtpproxy has 1 listen interface Supposed A and B are 2 clients that use rtpproxy to relay RTP stream, and works fine. A <---> port1 [*rtpproxy*] port2 <---> B Now that A loses his current network, and enter network2 (imagine a network handover) to become A2. In this case, I see rtpproxy still works fine by relaying stream between A2 and B A2 <---> port1 [*rtpproxy*] port2 <---> B But when A2 lose his network2 and enters network3 to become A3, rtpproxy stills relay stream between A2 and B. It seems that A can change his network only once. A2 <---> port1 [*rtpproxy*] port2 <---> B A3 Why did the first handover succeed? How can I change rtpproxy behavior to support many handovers ? -- Khoa Pham HCMC University of Science www.fantageek.com

3 7

releasing v4.0.3
by Daniel-Constantin Mierla 15 Aug '13

15 Aug '13

Hello, I am planning releasing v4.0.3 on Wednesday, Aug 14. If anyone has backports for branch 4.0, they have to be done before the noon on that day. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 3

is_subscriber function
by Daniel W. Graham 15 Aug '13

15 Aug '13

I am having an issue with the is_subscriber function provided by auth_db module. What I want to accomplish is: if the call is not to a local subscriber then route via LCR. If the call is to a local subscriber then route via LOCATION. My subscriber table is called sip_accounts and I am using multidomain so domain is populated in subscriber definitions. Here is my routing block- if (!is_subscriber("$ru", "sip_accounts", "1")){ route(LCR); exit; } else { route(LOCATION); exit; } Call always routes via LCR despite calling a local subscriber. INVITE is correct: INVITE sip:user2@domain.com SIP/2.0. From: user1 <sip:user1@domain.com>;tag=520c9f9ade1e7026o0. To: <sip:user2@domain.com>. Any ideas or direction where my issue could be? -dan

2 4

Re: [SR-Users] Loose Routing with Cisco router
by David Wilson 15 Aug '13

15 Aug '13

Alex, Thanks. It's good to know that it should work. Daniel, It's a 2811 running IOS 12.4(25g). >> [a] ===> [kamailio] ===> [cisco] ===> [b] This is exactly what I need. I'm trying to modify the request at [kamailio] so that [cisco] forwards to [b]. Possible complication is that [cisco] is also configured as a voice gateway with a SIP trunk to an external provider and a Call manager on the inside. However, this is independent SIP traffic that [cisco] appears to try to route according to voice rules. I can get the call to proceed by adding a dial-peer to [cisco] config, but if I have understood loose-routing, that shouldn't be necessary. I have tried several combinations of header and r-URI modifications to test [cisco]'s behaviour, including Route:[cisco];lr but it still seems to respond to the r-uri in the way I initially described. All the best, Dave On 14 Aug 2013, at 5:09 pm, sr-users-request(a)lists.sip-router.org wrote: > > > Message: 3 > Date: Wed, 14 Aug 2013 13:06:19 +0400 > From: Alex Balashov <abalashov(a)evaristesys.com> > To: miconda(a)gmail.com, "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Subject: Re: [SR-Users] Loose Routing with Cisco router > Message-ID: <65a98f42-c370-49fe-91c4-d018b3d61249(a)email.android.com> > Content-Type: text/plain; charset=UTF-8 > > I use Kamailio extensively with Cisco AS5xxx series gateways and they have never had a problem dealing with Kamailio's RRs and lr values. > > > Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: >> Hello, >> >> what model the cisco router is? >> >> From what I understand, you want to go from: >> >> [a] ===> [kamailio] ===> [b] >> >> to: >> >> [a] ===> [kamailio] ===> [cisco] ===> [b] >> >> The usual will be that kamailio sets the r-uri to [b] and dsturi >> (outbound proxy) to [cisco] and then relay. The cisco should have some >> config options just to forward the traffic based on r-uri, however I >> never had to deal cisco configs. Maybe you add a Route header with >> [cisco];lr, then it should match itself in the route header and send to >> >> r-uri. >> >> On the other hand, when I had to interoperate with cisco >> gateways/b2bua, >> they had no problem understanding loose routing added by kamailio. >> >> Cheers, >> Daniel >> >> On 8/13/13 10:05 AM, David Wilson wrote: >>> Hello All, >>> >>> I'm running Kamailio 4.0.2 on Ubuntu 12.04 (precise). >>> >>> I have SIP messaging flowing nicely, with UACs registering via >> Kamailio (as a proxy) to a Registrar. >>> >>> Now having problems trying to route messages via a Cisco router >> (12.4(25g)) using loose routing. The plan is to allow the router to >> see the SIP messaging but forward it to the original location. >>> >>> RFC 3261 seems to cover this case in Section 16.6 part 6: "A proxy >> MAY have a local policy that mandates that a request visit a specific >> set of proxies before being delivered to the destination." >>> >>> Cisco documentation (SIP Configuration Guide, Cisco IOS Release 12.4) >> states that it implements RFC 3261, including Loose-routing. However, >> from my observation the router doesn't like a request-URI with anything >> other than its own IP address (it returns 400 Bad Request - 'Invalid IP >> address'), but with any attempt to 'decorate' the URI with lr and/or >> maddr= parameters it returns 400 Bad Request - 'Malformed/Missing' URL. >> One of these messages is returned regardless of anything I've tried >> with Via, Record-Route or Route headers. >>> >>> Desired behaviour is for the router to remain in the route-set. A >> different configuration has shown me that this will achieve the >> required outcome, the tricky part now is just getting that first >> request to route. I know several ways to make Kamailio send to the >> router, but the router is not behaving as I expect when I preload a >> route set (with Route: headers). >>> >>> Recognising that this is probably more of a Cisco problem than >> Kamailio, can anyone confirm whether a Cisco router implements >> loose-routing when receiving messages, or only when sending? >>> >>> Regards, >>> Dave. >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >> list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Sent from my mobile, and thus lacking in the refinement one might expect from a fully fledged keyboard. > > Alex Balashov - Principal > Evariste Systems LLC > 235 E Ponce de Leon Ave > Suite 106 > Decatur, GA 30030 > United States > Tel: +1-678-954-0670 > Web: http://www.evaristesys.com/, http://www.alexbalashov.com > > >

1 0

Kamailio as SIP outbound proxy in front of Asterisk. Transparent proxy ?
by Renaud Dubois 14 Aug '13

14 Aug '13

All, I have just setup kamailio as SPI outbound proxy, in front of Asterisk. I'm novice with Kamailio, it's the first time I use it. The setup is working but I need your advises: 1) When I type the "sip show peers" command in ASterisk, I see the ip address of the sip proxy. The qualify (monitoring/keepalive) seems to be sent to the sip proxy and not to the phone. Is there an alternative to directly monitor the phone ? localhost*CLI> sip show peers phone2a/phone2a * <IP_SIP_PROXY> * D N 53 OK (299 ms) 2) If I use the command "localhost*CLI> sip show peer phone2a" The ip of the phone is visible in the field "Reg. Contact" only. In the field " Addr->IP", it's the IP of the SIP Proxy. * Name : phone2a Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : client2 Subscr.Cont. : <Not set> Language : AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : MOH Suggest : Mailbox : VM Extension : *97 LastMsgsSent : 32767/65535 Call limit : 0 Max forwards : 0 Dynamic : Yes Callerid : "" <> MaxCallBR : 384 kbps Expire : 554 Insecure : no Force rport : Yes ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr->IP : *<SIP PROXY IP>* Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: phone2a SIP Options : (none) Codecs : 0xa (gsm|alaw) Codec Order : (gsm:20,alaw:20) Auto-Framing : No Status : OK (299 ms) Useragent : LinphoneAndroid/2.1.2 (eXosip2/3.6.0) Reg. Contact : sip:phone2a@*<IP PHONE>* line=8b1b24fbaaf794a Qualify Freq : 60000 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No 3) I'm running a fail2ban protection to protect against scanners and my fail2ban is blocking the SIP Proxy when the threshold is reached, which means that all the clients behind the sip outbound proxy are blocked. I think the points 1, 2, 3 are related and if the SIP Proxy could be "transparent" Here is a debug of a register request, taken on the kamailio server REGISTER sip:pbx-qa.mydomain.com SIP/2.0 Via: SIP/2.0/UDP <IP SIP PROXY>;branch=z9hG4bKfea4.34a8fd83.0 Via: SIP/2.0/UDP 100.96.196.103:4294;received=<IP PHONE>;rport=6738;branch=z9hG4bK1329841729 From: <sip:phone2a@mydomain.com>;tag=1870152222 To: <sip:phone2a@pbx-qa.domain.com> Call-ID: 1455540209 CSeq: 2 REGISTER Contact: <sip:phone2a@100.96.196.103:4294;line=c6f956d7cdb0eb5> Authorization: Digest username="phone2a", realm="asterisk", nonce="176e8fa1", uri="sip:pbx-qa.domain.com", response="4c68e98p1ea7cb0ee81674a8384ca6e4", algorithm=MD5 Max-Forwards: 69 User-Agent: LinphoneAndroid/2.1.2 (eXosip2/3.6.0) Expires: 600 Content-Length: 0 P-hint: outbound Any ideas to solve my problem, to get a "more" transparent proxy ? Regards, Renaud Dubois

1 0

Logging with sip client
by Kethzer Docteur 14 Aug '13

14 Aug '13

I'm installing kamailio with a public IP on a server but for some reason in can't logging with my sip client. Any help please DISTRIB_ID=Ubuntu DISTRIB_RELEASE=10.04 DISTRIB_CODENAME=lucid DISTRIB_DESCRIPTION="Ubuntu 10.04.2 LTS" Iptables config *filter :INPUT ACCEPT [4231862:931412269] :FORWARD ACCEPT [0:0] :OUTPUT ACCEPT [5258380:5441121579] -A INPUT -p tcp -m tcp --dport 22 -j ACCEPT -A INPUT -p udp -m udp --dport 10000:65000 -j ACCEPT -A INPUT -p udp -m udp --sport 10000:65000 -j ACCEPT -A INPUT -p tcp -m tcp --dport 10000:65000 -j ACCEPT -A INPUT -p tcp -m tcp --sport 10000:65000 -j ACCEPT -A INPUT -p tcp -m tcp --dport 5060 -j ACCEPT -A INPUT -p tcp -m tcp --dport 5061 -j ACCEPT -A INPUT -p udp -m udp --dport 7722 -j ACCEPT -A INPUT -p udp -m udp --dport 5060 -j ACCEPT -A INPUT -p udp -m udp --dport 5061 -j ACCEPT COMMIT -- Kethzer Docteur

1 0

Logging with sip client
by Kethzer Docteur 14 Aug '13

14 Aug '13

I'm installing kamailio with a public IP on a server but for some reason in can't logging with my sip client. Any help please DISTRIB_ID=Ubuntu DISTRIB_RELEASE=10.04 DISTRIB_CODENAME=lucid DISTRIB_DESCRIPTION="Ubuntu 10.04.2 LTS" -- Kethzer Docteur

1 0

Licensing inquiry
by Ziad Habchi 14 Aug '13

14 Aug '13

Hi, I have an inquiry on the licensing and terms and conditions for using Kamailio to build my own application. I want to use Kamailio to build my own Application Server. I will write my own commercial application that will interact with Kamailio using SEAS module, and I would like to know under which licensing scheme will that fall, and what terms and conditions will apply? Thank you in advance for your reply. Ziad Habchi

1 0

Loose Routing with Cisco router
by David Wilson 14 Aug '13

14 Aug '13

Hello All, I'm running Kamailio 4.0.2 on Ubuntu 12.04 (precise). I have SIP messaging flowing nicely, with UACs registering via Kamailio (as a proxy) to a Registrar. Now having problems trying to route messages via a Cisco router (12.4(25g)) using loose routing. The plan is to allow the router to see the SIP messaging but forward it to the original location. RFC 3261 seems to cover this case in Section 16.6 part 6: "A proxy MAY have a local policy that mandates that a request visit a specific set of proxies before being delivered to the destination." Cisco documentation (SIP Configuration Guide, Cisco IOS Release 12.4) states that it implements RFC 3261, including Loose-routing. However, from my observation the router doesn't like a request-URI with anything other than its own IP address (it returns 400 Bad Request - 'Invalid IP address'), but with any attempt to 'decorate' the URI with lr and/or maddr= parameters it returns 400 Bad Request - 'Malformed/Missing' URL. One of these messages is returned regardless of anything I've tried with Via, Record-Route or Route headers. Desired behaviour is for the router to remain in the route-set. A different configuration has shown me that this will achieve the required outcome, the tricky part now is just getting that first request to route. I know several ways to make Kamailio send to the router, but the router is not behaving as I expect when I preload a route set (with Route: headers). Recognising that this is probably more of a Cisco problem than Kamailio, can anyone confirm whether a Cisco router implements loose-routing when receiving messages, or only when sending? Regards, Dave.

3 2

How to install ims modules of kamailio and test the functionality
by Sai Krishna Kota 14 Aug '13

14 Aug '13

Hello, I am new user of Kamailio sip server. I am able install kamailio sip server 4.0.2. Now I am trying modules also. I need to make use of kamailio ims modules(Ims_cscf) so please help me to install ims modules with kamailio server. It will be more help full If details of step by step compilation from the source code is given. Thank you, Kota.

1 0

empty ruid error messages
by Juha Heinanen 13 Aug '13

13 Aug '13

in kamailio 4.0, what could cause these kind of error messages: Aug 2 09:31:04 voip /usr/sbin/sip-proxy[1779]: ERROR: usrloc [ucontact.c:842]: db_update_ucontact_ruid(): updating record in database failed - empty ruid Aug 2 09:31:04 voip /usr/sbin/sip-proxy[1779]: ERROR: usrloc [urecord.c:363]: wb_timer(): updating contact in db failed Aug 2 09:32:04 voip /usr/sbin/sip-proxy[1779]: ERROR: usrloc [ucontact.c:842]: db_update_ucontact_ruid(): updating record in database failed - empty ruid Aug 2 09:32:04 voip /usr/sbin/sip-proxy[1779]: ERROR: usrloc [urecord.c:363]: wb_timer(): updating contact in db failed relevant usrloc params are: modparam("usrloc", "db_mode", 1) modparam("usrloc", "db_ops_ruid", 1) modparam("usrloc", "db_check_update", 1) modparam("usrloc", "timer_procs", 2) how is it possible that ruid in memory record is empty? when db_mode=1, why kamailio does try to update (rather than delete) a record in database by timer? -- juha

2 4

Content-Type Header missing from third part registration
by Ziad Habchi 13 Aug '13

13 Aug '13

Hi , I managed to run Kamailio to replace OpenIMSCore. I am using boghe client to sign in, when I do so , my registration is forwarded as per the trigger point to my AS server. As I notice , the REGISTER request miss Content-Type header which is mandatory in the 3gpp TS . Is this configurable ? Am I doing something wrong or is it a bug? Thank you for your help Ziad Habchi

3 7

kamailio call probelm
by Jason Sia 13 Aug '13

13 Aug '13

Hi, I installed kamailio v 3.1.6 rpm. I used the default configuration. Kamailio started. I have two clients one is an android phone using native sip client, and the other one x-lite. I can call x-lite to phone but not the other way around. When I restarted kamailio, I can call phone to x-lite but not the other way around. The first one I did when I restart kamailio will be the only leg that will be working, then I have to restart again if I want to do the reverse. Has anyone encountered this problem? How do I fix it? Thanks, Jason

2 1

Kamailio v3.3.5 Released
by Daniel-Constantin Mierla 12 Aug '13

12 Aug '13

Hello, Kamailio SIP Server v3.3.5 stable release is out. This is a maintenance release of the previous stable branch, 3.3, that includes fixes since release of v3.3.4. There is no change to database schema or configuration language structure that you have to do on installations of previous v3.3.x. Deployments running previous v3.x.x versions are strongly recommended to be upgraded to v3.3.5. For more details about version 3.3.5 (including links and hints to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2013/08/kamailio-v3-3-5-released/ Note that latest stable series is now 4.0.x, with upcoming v4.0.3 to be released later this week. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio as a credit control application (RFC 4006)
by Alexey Rybalko 12 Aug '13

12 Aug '13

Hi! Does Kamailio has features of CCA for usage outside a compiled code (cdp_avp)? There's a module *ims_ro_interface *listed at http://www.kamailio.org/w/2013/05/ims-kamailio/#more-1664 What status does it have? regards, Alexey

2 1

Duplicate relayed INVITE message using siptrace not working
by Grant Bagdasarian 12 Aug '13

12 Aug '13

Hello, I'm adding some additional SIP headers to the INVITE received, but when I use the siptrace module to duplicate the messages to a capture server, the INVITE does not contain the additional headers. I'm using a softphone to test my configuration, and it looks like Kamailio only sends the INVITE which is coming from the softphone to the capture server. When Kamailio relays the INVITE to its destination the headers are present. Is there a way to duplicate the invite leaving kamailio to the capture server instead of the INVITE which is coming into kamailio? I already tried to put the sip_trace() function after the message is sent to its destination, but the extra headers didn't show up in the database. Note, that I'm using the "X-" as a prefix. Regards, Grant

2 7

CDR generation in Kamailio.
by Nandini madhu 12 Aug '13

12 Aug '13

Dear All, I am working on Kamailio (4.0.1) server with GUI package Siremis (4.0) of it. I was successful in get through Calls and SMS features in our VoIP phones using this kamailio sever. But i am encountering some issues related to CDR data generation into mysql database. I have activated WITH_ACCDB in kamailio cfg file. Siremis is configured to kamailio DB properly as i am getting accounting list in the siremis GUI well, but only the CDR data is not getting reflecting in GUI. But this CDR data is successfully logging into syslog (see the paragraph below), but my requirement is to get the CDR data to be reflect in my GUI. Here below is the CDR data that logged into syslog: Aug 3 17:36:25 kamailio /usr/local/sbin/kamailio[12391]: INFO: acc [acc.c:275]: acc_log_request(): ACC: transaction answered: timestamp=1375531585;method=INVITE;from_tag=170287292812;to_tag=373699520912;call_id= 6595ee13-bfec-15a7-ae-4b-00210705d550(a)192.168.2.99;code=200;reason=OK;src=;dst= Aug 3 17:36:25 kamailio /usr/local/sbin/kamailio[12393]: INFO: acc [acc.c:275]: acc_log_request(): ACC: request acknowledged: timestamp=1375531585;method=ACK;from_tag=170287292812;to_tag=373699520912;call_id= 6595ee13-bfec-15a7-ae-4b-00210705d550(a)192.168.2.99;code=200;reason=OK;src=;dst= Aug 3 17:36:29 kamailio /usr/local/sbin/kamailio[12391]: INFO: acc [acc_cdr.c:192]: write_cdr(): st=1375531585.654; et=1375531588.863; duration=3.209; src_user=102; src_domain=192.168.2.41; src_ip=192.168.2.99; dst_ouser=101; dst_user=101; dst_domain=192.168.2.77 And my kamailio.cfg script relating to CDR's is like this: /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACCDB) /* FLT_ACCDB is 2 */ modparam("acc", "db_missed_flag", FLT_ACCMISSED) /* FLT_ACCMISSED is 3 */ modparam("acc", "db_url", mysql://kamailio:kamailiorw@kamailio/kamailio) #modparam("acc", "db_extra","src_user=$fU;src_domain=$fd;src_ip=$si;""dst_ouser=$tU;dst_user= $rU;dst_domain=$rd;""duration=$DLG_lifetime") #modparam("acc", "db_extra","${cdr_extra}") modparam("acc", "db_extra","src_user=$fU;src_domain=$fd;src_ip=$si;""dst_ouser=$tU;dst_user= $rU;dst_domain=$rd") modparam("acc", "db_table_acc", "acc") #modparam("acc", "db_table_acc", "acc_$time(year)_$time(mon)") modparam("acc", "db_table_missed_calls", "missed_calls") modparam("acc", "acc_method_column", "method") modparam("acc", "acc_from_tag_column", "from_tag") modparam("acc", "acc_to_tag_column", "to_tag") modparam("acc", "acc_callid_column", "callid") modparam("acc", "acc_sip_code_column", "sip_code") modparam("acc", "acc_sip_reason_column", "sip_reason") modparam("acc", "acc_time_column", "time") modparam("acc", "db_insert_mode", 0) #!endif # -------- for siremis CDRs ----------- modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "cdr_facility", "LOG_DAEMON") #modparam("acc", "cdr_extra", "c1=$dlg_var(caller);c2=$dlg_var(callee)") modparam("acc", "cdr_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;""dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "cdr_start_id", "st") modparam("acc", "cdr_end_id", "et") modparam("acc", "cdr_duration_id", "duration") . modparam("rtimer", "timer", "name=cdr;interval=300;mode=1;") modparam("rtimer", "exec", "timer=cdr;route=2") modparam("sqlops", "sqlcon", "ca=>mysql://kamailio:kamailiorw@localhost/kamailio") modparam("sqlops", "sqlres", "ra") . . # Populate CDRs Table of Siremis route[2] { sql_query("ca","call kamailio_cdrs()","ra"); } Can u guess where i am wrong ? how can i solve this issue ? And for your notice, in acc module document it is said like this: 'Note that CDR generation does not involve any kind of database storage (yet). In order to persist the CDRs into a database you will have to set up an exterior process (i.e., a script living outside of Kamailio) and implement the storage task yourself'. What does it mean ? of course 'cdrs' table in kamailio database is empty. Any help will greatly appreciate. Regards, Nandini.

2 1

error in tm.so after REGISTER
by Nguyen Anh Tuan 12 Aug '13

12 Aug '13

Dear all, Kamailio was crashed and I got syslog file, but I don't understand what was happened with this error? Aug 9 16:27:33 Kamailio /usr/sbin/kamailio[32414]: WARNING: tm [t_lookup.c:1559]: WARNING: script writer didn't release transaction Aug 9 16:27:33 Kamailio /usr/sbin/kamailio[32418]: NOTICE: <script>: Set NAT Transaction Flag REGISTER from sip:1008@10.84.8.55 (IP:203.160.19.30:58493) to sip:1008@10.84.8.55 in route[NATDETECT] Aug 9 16:27:33 Kamailio /usr/sbin/kamailio[32418]: INFO: <script>: Unregister user [sip:1008@10.84.8.55] Aug 9 16:27:33 Kamailio /usr/sbin/kamailio[32404]: INFO: <script>: Unregister user [sip:1008@10.84.8.55] Aug 9 16:27:34 Kamailio /usr/sbin/kamailio[32403]: NOTICE: <script>: Set NAT Transaction Flag REGISTER from sip:1004@10.84.8.55 (IP:203.187.4.6:12344) to sip:1004@10.84.8.55 in route[NATDETECT] Aug 9 16:27:34 Kamailio /usr/sbin/kamailio[32403]: INFO: <script>: Unregister user [sip:1004@10.84.8.55] Aug 9 16:27:34 Kamailio /usr/sbin/kamailio[32392]: NOTICE: <script>: Set NAT Transaction Flag REGISTER from sip:1004@10.84.8.55 (IP:203.187.4.6:12344) to sip:1004@10.84.8.55 in route[NATDETECT] Aug 9 16:27:35 Kamailio /usr/sbin/kamailio[32419]: NOTICE: <script>: Set NAT Transaction Flag REGISTER from sip:1008@10.84.8.55 (IP:203.160.19.30:58493) to sip:1008@10.84.8.55 in route[NATDETECT] Aug 9 16:27:35 Kamailio /usr/sbin/kamailio[32390]: NOTICE: <script>: Set NAT Transaction Flag REGISTER from sip:1004@10.84.8.55 (IP:203.187.4.6:12344) to sip:1004@10.84.8.55 in route[NATDETECT] Aug 9 16:27:35 Kamailio kernel: [75478.500981] kamailio[32395] general protection ip:7f9fbd41089d sp:7fffde4e5620 error:0 in tm.so[7f9fbd3d5000+8e000] Aug 9 16:27:35 Kamailio /usr/sbin/kamailio[32406]: NOTICE: <script>: Set NAT Transaction Flag REGISTER from sip:1004@10.84.8.55 (IP:203.187.4.6:17209) to sip:1004@10.84.8.55 in route[NATDETECT] Aug 9 16:27:35 Kamailio /usr/sbin/kamailio[32446]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 19 Aug 9 16:27:35 Kamailio /usr/sbin/kamailio[32387]: ALERT: <core> [main.c:788]: child process 32395 exited by a signal 11 Aug 9 16:27:35 Kamailio /usr/sbin/kamailio[32387]: ALERT: <core> [main.c:791]: core was generated Aug 9 16:27:35 Kamailio /usr/sbin/kamailio[32446]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 19 Aug 9 16:27:35 Kamailio /usr/sbin/kamailio[32387]: ALERT: <core> [main.c:788]: child process 32395 exited by a signal 11 Aug 9 16:27:35 Kamailio /usr/sbin/kamailio[32387]: ALERT: <core> [main.c:791]: core was generated Aug 9 16:27:35 Kamailio /usr/sbin/kamailio[32387]: INFO: <core> [main.c:803]: INFO: terminating due to SIGCHLD Aug 9 16:27:35 Kamailio kernel: [75478.963781] kamailio[32387] general protection ip:7f9fbd3ede22 sp:7fffde4e5c90 error:0 in tm.so[7f9fbd3d5000+8e000] Best Regards, Nguyen Anh Tuan Network solution engineer

2 1

releasing v3.3.5
by Daniel-Constantin Mierla 12 Aug '13

12 Aug '13

Hello, I am considering to package soon a new version out of branch 3.3. If anyone has spotted fixes that should be backported to that branch, reply here. I'm not sure yet of an exact date, but should be rather soon, being just a patch update. A new release out of branch 4.0 will be done short after this one. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 4

Profile sharing using xcap
by Premchandiran 12 Aug '13

12 Aug '13

Hi All, I am looking for softphone which supports profile-sharing using xcap. Please let me know if there is any softphone available. Regards, Prem Chandiran M

1 0

Generate UUID in Kamailio Script
by Grant Bagdasarian 09 Aug '13

09 Aug '13

Hello, Is it possible to generate a UUID inside the kamailio script? Is there a module available which does this already or do I need to call something external from the script? Regards, Grant

3 8

Trouble when bridging call between UDP and TCP clients behind NAT
by Roberto Fichera 09 Aug '13

09 Aug '13

Hi All, I'm try to setup kamilio v4.0.2 under a Fedora18 64bits having a public address in order to bridge calls between UDP and TCP clients. Everything works pretty fine between UDP clients even if behind NAT. But as soon as one TCP client is being called I'm getting this below on kamailio side. Note that the port is wrong and the TCP client doesn't get aliased nor tcp port remappend or received field is set: Call-ID: 23306738-94b3-49b5-b7e8-dc1b8a8061db CSeq: 50696 REGISTER User-Agent: PJSUA v2.1-svn Linux-3.10.5.44/x86_64/glibc-2.17 Supported: outbound, path Contact: <sip:528@94.94.X.X:1117;transport=TCP;ob>;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0000be3c954f>" Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP 94.94.X.X:56120;rport=56120;branch=z9hG4bKPj75004c2a-7ba3-4807-805c-a3ac8016e5b7;alias From: <sip:528@test.domain;hide>;tag=93cc1d8e-9ccc-4d36-8587-923ff44daf89 To: <sip:528@test.domain;hide>;tag=333a0370df4a40d5d5a0c21bb156e2a6.aab2 Call-ID: 23306738-94b3-49b5-b7e8-dc1b8a8061db CSeq: 50696 REGISTER WWW-Authenticate: Digest realm="test.domain", nonce="UgOcJlIDmvotaQR8dxYacL2fFlyotlC2" Server: kamailio (4.0.2 (x86_64/linux)) Content-Length: 0 REGISTER sip:test.domain;transport=tcp;hide SIP/2.0 Via: SIP/2.0/TCP 94.94.X.X:56120;rport;branch=z9hG4bKPja2e404d0-9812-4b7b-b86b-3f3b1a27d204;alias Max-Forwards: 70 From: <sip:528@test.domain;hide>;tag=93cc1d8e-9ccc-4d36-8587-923ff44daf89 To: <sip:528@test.domain;hide> Call-ID: 23306738-94b3-49b5-b7e8-dc1b8a8061db CSeq: 50697 REGISTER User-Agent: PJSUA v2.1-svn Linux-3.10.5.44/x86_64/glibc-2.17 Supported: outbound, path Contact: <sip:528@94.94.X.X:1117;transport=TCP;ob>;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0000be3c954f>" Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Authorization: Digest username="528(a)test.domain", realm="test.domain", nonce="UgOcJlIDmvotaQR8dxYacL2fFlyotlC2", uri="sip:test.domain;transport=tcp;hide", response="d6e93d582744a3cf12cc48989f66872e" Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/TCP 94.94.X.X:56120;rport=56120;branch=z9hG4bKPja2e404d0-9812-4b7b-b86b-3f3b1a27d204;alias From: <sip:528@test.domain;hide>;tag=93cc1d8e-9ccc-4d36-8587-923ff44daf89 To: <sip:528@test.domain;hide>;tag=333a0370df4a40d5d5a0c21bb156e2a6.d863 Call-ID: 23306738-94b3-49b5-b7e8-dc1b8a8061db CSeq: 50697 REGISTER Contact: <sip:528@94.94.X.X:1117;transport=TCP;ob>;expires=300;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0000be3c954f>";reg-id=1 Server: kamailio (4.0.2 (x86_64/linux)) Content-Length: 0 Thus, kamailio trigger this error: Aug 9 09:20:47 proxy /usr/sbin/kamailio[3553]: ERROR: <core> [tcp_main.c:4247]: handle_tcpconn_ev(): connect 94.94.X.X:1117 failed >From the client side (pjsua client) the log looks: REGISTER sip:test.domain;transport=tcp;hide SIP/2.0 Via: SIP/2.0/TCP 192.168.2.90:56120;rport;branch=z9hG4bKPj75004c2a-7ba3-4807-805c-a3ac8016e5b7;alias Max-Forwards: 70 From: <sip:528@test.domain;hide>;tag=93cc1d8e-9ccc-4d36-8587-923ff44daf89 To: <sip:528@test.domain;hide> Call-ID: 23306738-94b3-49b5-b7e8-dc1b8a8061db CSeq: 50696 REGISTER User-Agent: PJSUA v2.1-svn Linux-3.10.5.44/x86_64/glibc-2.17 Supported: outbound, path Contact: <sip:528@192.168.2.90:5060;transport=TCP;ob>;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0000be3c954f>" Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 --end msg-- SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP 192.168.2.90:56120;rport=56120;branch=z9hG4bKPj75004c2a-7ba3-4807-805c-a3ac8016e5b7;alias From: <sip:528@test.domain;hide>;tag=93cc1d8e-9ccc-4d36-8587-923ff44daf89 To: <sip:528@test.domain;hide>;tag=333a0370df4a40d5d5a0c21bb156e2a6.aab2 Call-ID: 23306738-94b3-49b5-b7e8-dc1b8a8061db CSeq: 50696 REGISTER WWW-Authenticate: Digest realm="test.domain", nonce="UgOcJlIDmvotaQR8dxYacL2fFlyotlC2" Server: kamailio (4.0.2 (x86_64/linux)) Content-Length: 0 --end msg-- REGISTER sip:test.domain;transport=tcp;hide SIP/2.0 Via: SIP/2.0/TCP 192.168.2.90:56120;rport;branch=z9hG4bKPja2e404d0-9812-4b7b-b86b-3f3b1a27d204;alias Max-Forwards: 70 From: <sip:528@test.domain;hide>;tag=93cc1d8e-9ccc-4d36-8587-923ff44daf89 To: <sip:528@test.domain;hide> Call-ID: 23306738-94b3-49b5-b7e8-dc1b8a8061db CSeq: 50697 REGISTER User-Agent: PJSUA v2.1-svn Linux-3.10.5.44/x86_64/glibc-2.17 Supported: outbound, path Contact: <sip:528@192.168.2.90:5060;transport=TCP;ob>;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0000be3c954f>" Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Authorization: Digest username="528(a)test.domain", realm="test.domain", nonce="UgOcJlIDmvotaQR8dxYacL2fFlyotlC2", uri="sip:test.domain;transport=tcp;hide", response="d6e93d582744a3cf12cc48989f66872e" Content-Length: 0 --end msg-- SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.2.90:56120;rport=56120;branch=z9hG4bKPja2e404d0-9812-4b7b-b86b-3f3b1a27d204;alias From: <sip:528@test.domain;hide>;tag=93cc1d8e-9ccc-4d36-8587-923ff44daf89 To: <sip:528@test.domain;hide>;tag=333a0370df4a40d5d5a0c21bb156e2a6.d863 Call-ID: 23306738-94b3-49b5-b7e8-dc1b8a8061db CSeq: 50697 REGISTER Contact: <sip:528@192.168.2.90:5060;transport=TCP;ob>;expires=300;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0000be3c954f>";reg-id=1 Server: kamailio (4.0.2 (x86_64/linux)) Content-Length: 0 --end msg-- REGISTER sip:test.domain;transport=tcp;hide SIP/2.0 Via: SIP/2.0/TCP 192.168.2.90:56120;rport;branch=z9hG4bKPj3f8c138b-df5e-40a1-8482-fa792c362291;alias Max-Forwards: 70 From: <sip:528@test.domain;hide>;tag=2256e1a1-bad1-4f02-a9ff-45ff349f2c52 To: <sip:528@test.domain;hide> Call-ID: 23306738-94b3-49b5-b7e8-dc1b8a8061db CSeq: 50698 REGISTER User-Agent: PJSUA v2.1-svn Linux-3.10.5.44/x86_64/glibc-2.17 Supported: outbound, path Contact: <sip:528@192.168.2.90:5060;transport=TCP;ob>;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0000be3c954f>" Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 --end msg-- SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP 192.168.2.90:56120;rport=56120;branch=z9hG4bKPj3f8c138b-df5e-40a1-8482-fa792c362291;alias From: <sip:528@test.domain;hide>;tag=2256e1a1-bad1-4f02-a9ff-45ff349f2c52 To: <sip:528@test.domain;hide>;tag=333a0370df4a40d5d5a0c21bb156e2a6.15dd Call-ID: 23306738-94b3-49b5-b7e8-dc1b8a8061db CSeq: 50698 REGISTER WWW-Authenticate: Digest realm="test.domain", nonce="UgOdTVIDnCHViWa5LC/JUVsF0HRn4wA5" Server: kamailio (4.0.2 (x86_64/linux)) Content-Length: 0 --end msg-- REGISTER sip:test.domain;transport=tcp;hide SIP/2.0 Via: SIP/2.0/TCP 192.168.2.90:56120;rport;branch=z9hG4bKPj9539d611-09a9-4cac-bd29-bfe68132c058;alias Max-Forwards: 70 From: <sip:528@test.domain;hide>;tag=2256e1a1-bad1-4f02-a9ff-45ff349f2c52 To: <sip:528@test.domain;hide> Call-ID: 23306738-94b3-49b5-b7e8-dc1b8a8061db CSeq: 50699 REGISTER User-Agent: PJSUA v2.1-svn Linux-3.10.5.44/x86_64/glibc-2.17 Supported: outbound, path Contact: <sip:528@192.168.2.90:5060;transport=TCP;ob>;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0000be3c954f>" Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Authorization: Digest username="528(a)test.domain", realm="test.domain", nonce="UgOdTVIDnCHViWa5LC/JUVsF0HRn4wA5", uri="sip:test.domain;transport=tcp;hide", response="3d2ed2fc1da090f592424cfa5804943e" Content-Length: 0 --end msg-- SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.2.90:56120;rport=56120;branch=z9hG4bKPj9539d611-09a9-4cac-bd29-bfe68132c058;alias From: <sip:528@test.domain;hide>;tag=2256e1a1-bad1-4f02-a9ff-45ff349f2c52 To: <sip:528@test.domain;hide>;tag=333a0370df4a40d5d5a0c21bb156e2a6.cf79 Call-ID: 23306738-94b3-49b5-b7e8-dc1b8a8061db CSeq: 50699 REGISTER Contact: <sip:528@192.168.2.90:5060;transport=TCP;ob>;expires=300;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0000be3c954f>";reg-id=1 Server: kamailio (4.0.2 (x86_64/linux)) Content-Length: 0 Does anyone can show me how solve this problem? Thanks in advance, Roberto Fichera.

2 2

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