- sr-users - kamailio.org

New developer: Federico Cabiddu
by Daniel-Constantin Mierla 10 Sep '14

10 Sep '14

Hello, I would like to announce that Federico Cabiddu has now write access to Kamailio git repository. He is going to push a new module named tsilo, which facilitates adding new branches to active transactions -- the typical case implemented so far is to track invites per callee and add a new branch if the callee registers while invite transaction is active, quite useful for services with mobile clients that are getting push notifications to wake up. Federico is one of the guys using Kamailio for many years, currently working at Orange, France. His commit user id is: fcabiddu My warm welcome and thanks for joining the effort to develop Kamailio! Looking forward to future contribution to the project, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany

1 0

new module db_kazoo
by Luis Azedo 10 Sep '14

10 Sep '14

Hello Awesome Kamailio Community, We are writing on behalf of 2600hz, where we have been using Kamailio for some time now and are very pleased with our results! So, seems time to commit something back, dontchya think? We would like to present to you a new module, to hopefully be included in master. We call it db_kazoo (although a new name is fine too). db_kazoo is a general purpose AMQP connector (connects to our rabbitmq-server). It exposes publish/consume capabilities into Kamailio. Why is this amazing, you ask? Well even if you didn’t ask, we will explain… >From a high-level, the purpose of the module might be for things like: - Integrate to an AMQP application to make real-time routing decisions (instead of using, say, a SQL database) - Provide a real-time integration into your program, instead of your database, so you can overlay additional logic in your preferred language while also utilizing a message bus - Utilize messaging to have a distributed messaging layer, such that machines processing requests/responses/events can go up/down or share the workload and your Kamailio node will still be happy With this module, someone can: 1 - publish json payloads to rabbitmq 2 - publish json payloads to rabbitmq and wait for correlated response message 3 - subscribe to an exchange with a routing key The module works with a main forked process that does the communication with rabbitmq for issuing publishes, waiting for replies and consuming messages. When it consumes a message it defers the process to a worker process so that it doesn't block this main process. The worker process issues an event-route where we can act on the received payload. The name of the event-route is composed by values extracted from the payload. Consumed messages have the option of being acknowledge in two ways: 1 - immediately when received 2 - after processing by the worker One unique feature of our implementation revolves around failover of the message bus itself. In our design, the module supports multiple RabbitMQ servers and will fallback from one to the next in a list of RabbitMQ servers if the connection fails to the current connected server. When using acknowledge in db_kazoo with clustering in RabbitMQ, we have simulated and experienced full reconnects while in the middle of processing pending messages. In this way, when we get disconnected from one server we proved that, even at high speeds, we are able to connect to the next in the list and continue interacting with our application. We have run a variety of sipp load tests on this module and believe it is ready for prime time. We monitored memory, response accuracy and overall stability and it seemed OK. But we would, of course, love for others to help us find what we have missed, or contribute more features, or overall just use the work we slaved over for so many hours. Or just make comments and suggestions! If nothing else, we hope we’ve provided something useful to the Kamailio community, as you all have provided useful items to us. Thanks to everyone for all the work on Kamailio and related products. Look forward to hearing from you all. This is our first formal / large contribution to Kamailio so if we’ve done something wrong process, code or otherwise please let us know!

3 5

Location of Encryption Algorithm in TLS Module
by aawaise 10 Sep '14

10 Sep '14

Hello, I have downloaded and unzipped kamailio package available online. I want to know about TLS module, My question is that where is the encryption module located in TLS module ?? Is it in some .c file of TLS module or does TLS module use AES encryption available in LINUX or UNIX systems?? This question arises because I want to customize AES algorithm ?? Can it be done in TLS module or not ?? Regards, Aawaise -- View this message in context: http://sip-router.1086192.n5.nabble.com/Location-of-Encryption-Algorithm-in… Sent from the Users mailing list archive at Nabble.com.

2 1

Freezing development for v4.2 at the end of today
by Daniel-Constantin Mierla 10 Sep '14

10 Sep '14

Hello, the development of new features for next major release, v4.2.0, is going to be frozen by end of today. As we have developers all over the world, we are going to allow everyone to benefit of the full day, so technically is like end of Wednesday, September 10, in Hawaii ( :-) a good fit, because it is the home region for the name of the project. If there is a patch with new features that you want to push, but needs review, we are going to take care of them in the next days, but upload them to the tracker or send them to sr-dev mailing list by today. Then follows about one month of testing and bug fixing for the existing code, with the target to have the 4.2.0 release out around mid of October. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany

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New developer: Luis Azedo
by Daniel-Constantin Mierla 10 Sep '14

10 Sep '14

Hello, I would like to announce that a new developer has now write access to Kamailo git repository: Luis Azedo. His commit username is: lazedo He is going to push soon the 'kazoo' module that was discussed these days on the mailing list - a module connector for the Kazoo - the open source, distributed VoIP platform using Kamailio for core SIP routing. He also provided many patches to other components so far. Luis works for 2600hz, an USA company, and will be accompanied by Darren Schreiber and Karl Andreson in maintenance of the new module. My warm welcome and thanks for joining the effort of developing Kamailio further! Looking forward to future contributions to the project! Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany

1 0

No BYE from Called party
by Yuriy Gorlichenko 09 Sep '14

09 Sep '14

Hello All. I have kamailio with provider connection (trunk) When I call to external number through my provider call extablished Ok. But when i try hangup call from external number no BYE sended to me. When I hangup call from my kamailio (internal num) I send by to exteral number and it respond me Ok so session if fully complete. I guess that BYE from external number not recieves to me because I have wrong routing header fields at my INVITe or ACK messages, but can not find any information what what header must recieve info to external number where send BYE at hangup or thomething like this. This is my little dump for situation wherer I hangup from internal number and BYE finished successfully: IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 1599 E....< .@.'. ...6........G.RINVITE sip:12345678900@my.provider.com:5060 SIP/2.0 Record-Route: <sip:my.kamailio.com:5068;nat=yes;ftag=as5872f19e;lr=on> Via: SIP/2.0/UDP my.kamailio.com:5068 ;branch=z9hG4bK5c63.57e9ba848fadd4e228caa4445baf9d76.2 Via: SIP/2.0/UDP my.client.internal.ip:50600;branch=z9hG4bK4d90cebf;rport=50600 Max-Forwards: 70 From: <sip:TrunkNum@my.provider.com>;tag=as5872f19e To: <sip:12345678900@my.provider.com:5068> Contact:<TrunkNum@my.kamailio.com:5068> Call-ID: 42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600 CSeq: 102 INVITE User-Agent: Asterisk PBX 12.5.0 Date: Thu, 04 Sep 2014 21:53:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 544 Proxy-Authorization: Digest username="TrunkNum", realm="my.provider.com", nonce="VAjgfVQI31EsekTXtYjBzKa59XFSJB5P", uri=" sip:12345678900@my.provider.com:5060", qop=auth, nc=00000001, cnonce="3619116795", response="f5bc1d8125dd9e448d2e73764823adee", algorithm=MD5 v=0 o=root 1022912010 1022912010 IN IP4 my.kamailio.com s=Asterisk PBX 12.5.0 c=IN IP4 my.kamailio.com t=0 0 a=ice-lite m=audio 30032 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=rtcp:30033 a=ice-ufrag:3o8JrqkF a=ice-pwd:m8q4khQT8NKSABqeUkKs7ed8ClR2 a=candidate:TgT1dfTnI3kBgWQ IP my.proider.com.5060 > my.kamailio.com.5068: UDP, length 1882 E..... ./...6... ........b..SIP/2.0 200 OK Via: SIP/2.0/UDP my.kamailio.com:5068;rport=5068;received=my.kamailio.com ;branch=z9hG4bK5c63.57e9ba848fadd4e228caa4445baf9d76.2 Via: SIP/2.0/UDP my.client.internal.ip:50600;branch=z9hG4bK4d90cebf;rport=50600 Record-Route: <sip:my.proider.com;lr=on;ftag=as5872f19e> Record-Route: <sip:my.kamailio.com:5068;nat=yes;ftag=as5872f19e;lr=on> From: <sip:TrunkNum@my.provider.com>;tag=as5872f19e To: <sip:12345678900@my.provider.com:5068>;tag=5rF0FNamQ99gH Call-ID: 42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600 CSeq: 102 INVITE Contact: <sip:12345678900@externail.number.end.ip:5060;transport=udp> User-Agent: provider agent Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 746 X-provider agentOutboundGateway: sip:5574012345678900@62.93.147.149 X-provider agentOutboundCarrierID: 23705946361020 X-provider agentCarrierRate: 0.20180 X-provider agentCloudRate: 0.00300 Remote-Party-ID: "Outbound Call" <sip:5060@my.provider.com >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1409844386 1409844387 IN IP4 externail.number.end.ip s=FreeSWITCH c=IN IP4 externail.number.end.ip t=0 0 a=msid-semantic: WMS hQBNQPw0VjJInvOA0H6HPZyePNO0IIqP m=audio 23216 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=ssrc:2990874569 cname:pRs5xP IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 596 E..p.@..@.K\ ...6........\`.ACK sip:12345678900@externail.number.end.ip:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP my.kamailio.com:5068 ;branch=z9hG4bK5c63.8c0bfc7de0669f1b8252b97b8431b2e1.0 Via: SIP/2.0/UDP my.client.internal.ip:50600;branch=z9hG4bK5420b559;rport=50600 Route: <sip:my.proider.com;lr=on;ftag=as5872f19e> Max-Forwards: 70 From: <sip:TrunkNum@sip.callsion.com>;tag=as5872f19e To: <sip:12345678900@my.provider.com:5068>;tag=5rF0FNamQ99gH Call-ID: 42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600 CSeq: 102 ACK User-Agent: Asterisk PBX 12.5.0 Content-Length: 0 IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 617 E....D..@.KC ...6........q.ZBYE sip:12345678900@externail.number.end.ip:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP my.kamailio.com:5068 ;branch=z9hG4bK6c63.3deda3c1fee0ef86d7eae9549dd1b194.0 Via: SIP/2.0/UDP my.client.internal.ip:50600;branch=z9hG4bK65e48b85;rport=50600 Route: <sip:my.proider.com;lr=on;ftag=as5872f19e> Max-Forwards: 70 From: <sip:TrunkNum@sip.callsion.com>;tag=as5872f19e To: <sip:12345678900@my.provider.com:5068>;tag=5rF0FNamQ99gH Call-ID: 42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600 CSeq: 103 BYE User-Agent: Asterisk PBX 12.5.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 IP my.proider.com.5060 > my.kamailio.com.5068: UDP, length 568 E..T....-.676... ........@..SIP/2.0 200 OK Via: SIP/2.0/UDP my.kamailio.com:5068;received=my.kamailio.com ;branch=z9hG4bK6c63.3deda3c1fee0ef86d7eae9549dd1b194.0 Via: SIP/2.0/UDP my.client.internal.ip:50600;branch=z9hG4bK65e48b85;rport=50600 From: <sip:TrunkNum@sip.callsion.com>;tag=as5872f19e To: <sip:12345678900@my.provider.com:5068>;tag=5rF0FNamQ99gH Call-ID: 42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600 CSeq: 103 BYE User-Agent: provider agent Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Length: 0 Thanks for help.

2 4

Using app_java
by Tom Johnson 09 Sep '14

09 Sep '14

Any tips on compiling the app_java module? I am on CentOS 6. And I have libgcj-4.4.7 installed, but the build process complains about missing jni.h and other files.

2 1

Creating array out of hashtable
by aft 09 Sep '14

09 Sep '14

[1]Is it possible to iterate a htable without using "keys" like an array? [2] Is it possible to search for "keys" using regxp? -- -Cheers -Arif

2 2

Call connection from one server to another
by aawaise 09 Sep '14

09 Sep '14

Hello, I want to deploy two kamailio servers, both using same MYSQL database placed on Server 2008. Both are using different static IP's. Hopefully, I can add two A records in DNS server, both against same domain name, pointing to two different IP's to which two servers are attached. My question is that in load balancing scenario if user 1 is attached to server 1, and user 2 is attached to server 2. How can user1 call user2. Because request will go to server 1 to which user 1 is attached. But it wont be able to locate user 2 because it is not connected to server1 (but connected to server 2). I am using same domain name for two servers. What can be done in kamailio to solve this issue ?? Regards, aawaise -- View this message in context: http://sip-router.1086192.n5.nabble.com/Call-connection-from-one-server-to-… Sent from the Users mailing list archive at Nabble.com.

3 3

TLS Handshake failing with WSS
by Manuel Camarg 08 Sep '14

08 Sep '14

I'm trying to implement WSS with Kamailio Thing is that WS works fine, I've followed: http://nil.uniza.sk/sip/kamailio/configuring-kamailio-4x-websocket modparam("tls", "config", "webrtc/tls.cfg") In a tls.cfg file I have : [server:default] method = SSLv23 verify_certificate = no require_certificate = no private_key = webrtc/private.key certificate = webrtc/ssl.pem ca_list = webrtc/ca_list.pem In the log file: /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_server.c:178]: tls_complete_init(): Using TLS domain TLSs<default> /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_domain.c:700]: sr_ssl_ctx_info_callback(): SSL handshake started /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_main.c:2556]: tcpconn_do_send(): tcp_send: sending... /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_main.c:2590]: tcpconn_do_send(): tcp_send: after real write: c= 0x7f7513516958 n=5524 fd=11 /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_main.c:2591]: tcpconn_do_send(): tcp_send: buf=#012#026#003#003 /usr/local/sbin/kamailio[4025]: DEBUG: <core> [io_wait.h:390]: io_watch_add(): DBG: io_watch_add(0x8e0040, 11, 2, 0x7f7513516958), fd_no=1 /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_domain.c:712]: sr_ssl_ctx_info_callback(): SSL handshake done /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_domain.c:715]: sr_ssl_ctx_info_callback(): SSL disable renegotiation /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_server.c:348]: tls_accept(): TLS accept successful /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_server.c:355]: tls_accept(): tls_accept: new connection from 123.123.123.123:63300 using TLSv1/SSLv3 AES256-SHA 256 /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_server.c:358]: tls_accept(): tls_accept: local socket: 124.124.124.124:10443 /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_server.c:369]: tls_accept(): tls_accept: client did not present a certificate /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_main.c:2556]: tcpconn_do_send(): tcp_send: sending... /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_main.c:2590]: tcpconn_do_send(): tcp_send: after real write: c= 0x7f7513516958 n=282 fd=11 /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_main.c:2591]: tcpconn_do_send(): tcp_send: buf=#012#026#003#003 /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_read.c:296]: tcp_read_data(): EOF on 0x7f7513516958, FD 11 /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_read.c:1293]: tcp_read_req(): tcp_read_req: EOF /usr/local/sbin/kamailio[4025]: DEBUG: <core> [io_wait.h:617]: io_watch_del(): DBG: io_watch_del (0x8e0040, 11, -1, 0x10) fd_no=2 called /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_read.c:1437]: release_tcpconn(): releasing con 0x7f7513516958, state -1, fd=11, id=2 /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_read.c:1438]: release_tcpconn(): extra_data 0x7f7513510a88 /usr/local/sbin/kamailio[4029]: DEBUG: <core> [tcp_main.c:3385]: handle_tcp_child(): handle_tcp_child: reader response= 7f7513516958, -1 from 1 /usr/local/sbin/kamailio[4029]: DEBUG: tls [tls_server.c:597]: tls_h_close(): Closing SSL connection 0x7f7513510a88 In sipml5 the error: *Disconnected: Failed to connect to the server* In the Chrome console: *__tsip_transport_ws_onerror * *__tsip_transport_ws_onclose * SSL certificates seem to be ok: # openssl verify -CAfile ca_list.pem ssl.pem ssl.pem: OK Can't figure out a solution :( Any ideas? *Manuel Camargo* Teléfono: 638000836 eMail: sir.louen(a)gmail.com

2 3

Change caller name in kamailio as the extension remains same.
by Talha Bukhari 08 Sep '14

08 Sep '14

Hi, I am new to kamailio. i configure kmailio server and make call as local extensions like 1001 and etc. Now i want to know how to change the header name appear on softphone using script in kamailo as the extension remain same as 1001. what are the files i used to alter and where are to locate them. Hope you don't mind. Regards, Talha

2 4

out of memory ; needs 455
by Arun Kumar 08 Sep '14

08 Sep '14

Hi Server : kamailio 4.1.1 Configuration :Outbound with engage_media_proxy (See the attachment ) Shared Mem : 1024 MB Problem : after 1 month kamailio hang up with out of memory error but we have enough free shared memory , it might be memory pool size problem, i dont know how and where to set memory pool size

2 1

Kamailio - voicemail platform - asterisk
by Henny 08 Sep '14

08 Sep '14

Hello, I am new to kamailio...... I have an idea for a community voice-mail platform (with Asterisk)and was wondering if Kamailio could be of any use in this plan. Does somebody have any advice ? Thanx!

2 1

dialog match on loop back
by Kelvin Chua 06 Sep '14

06 Sep '14

hi guys, i'm running out of ideas i have a situation where I need to generate 2 sets of entries INVITE,BYE in acc (acc_db_request won't cut it) it's when a DID should be redirected to a user outside my domain, I need to generate a cdr for DID and another CDR for the outbound call. what i am trying to do is loop it back to itself before routing it out. the idea is kamailio assigns the DID call to a dialog. then the outbound call will be assigned to a separate dialog after it receives the loopback. this is not happening since the from_tag, to_tag and the callid are all the same. kamailio treats this call as just 1, not 2. is there any ideas out there to work around this scenario? Kelvin Chua

2 1

Adding IPs from which 503 messages originates to a database
by aft 06 Sep '14

06 Sep '14

I'm trying to configure HOMER sip capture server to do some accounting of the cases when "503 Service Unavailable" message arrives from capture agents. I've implemented the following to record the source IPs from which 503 message originates. onreply_route { ...................... ...................... ...................... else if(status == "503") { if($sht(a=>alarm::503) == $null) { $sht(a=>alarm::503) = 0; $sht(a=>alarm::$si::503) = 0; } xlog("Got 503"); xlog("($sht(a=>alarm::$si::503))"); $sht(a=>alarm::$si::503) = $sht(a=>alarm::$si::503) + 1; } ................... ................... ................... The following route supposed run periodically to update the databases. } route[TIMER_STATS] { ..................... ..................... xlog("($sht(a=>alarm::$si::503))"); if($sht(a=>alarm::$si::503) > 0) { sql_query("cb", "INSERT INTO alarm_data (create_date, type, total, description, source_ip) VALUES(NOW(), 'Too Many 503', $sht(a=>alarm::$si::503), 'service unavailable', $si)"); xlog("adding stuff to db"); } $sht(a=>alarm::503) = 0; $sht(a=>alarm::$si::503) = 0; } But the xlog("adding stuff to db") never executed. But in the first block of code, the log message to print the value $sht(a=>alarm::$si::503), which is shown to be non zero. Don't understand why its not working. -- -Cheers -Arif

2 4

Add custom parameter to To: header, and then recover it
by Alex Villacís Lasso 05 Sep '14

05 Sep '14

What functions exist (if any), to add custom parameters to the From: and To: headers?

2 1

Kamailio as a front end?
by Tom Johnson 05 Sep '14

05 Sep '14

I am looking for something that would act as front end to various SIP PBX systems. I am using Asterisk, but compatibility with others might be necessary. What I want to do is pass subscribe requests with certain content types to an external program (written in Java) and the rest onto the SIP PBX. Can Kamailio do this and if so, is there some example config files somewhere that I can look at to get started? Tom Johnson<mailto:tjohnson@microautomation.com> | Senior Software Engineer O: 703-543-2118 | F: 703-543-2099 5870 Trinity Parkway, Suite 600, Centreville, VA 20120 [cid:image001.png@01CE18AF.8F231EB0]<http://www.microautomation.com/>

2 1

mt_match avp behavior
by Juha Heinanen 05 Sep '14

05 Sep '14

looking at the code, i appears that mt_match does not initialize pv_values pv when it stores matched values thus keeping possible existing values from previous calls. is that intended behavior? readme uses word "store". -- juha

2 5

mtree README missing params
by Juha Heinanen 05 Sep '14

05 Sep '14

looks like 4.1 and master mtree README is missing these parameters: {"pv_dstid", STR_PARAM, &dstid_param.s}, {"pv_weight", STR_PARAM, &weight_param.s}, {"pv_count", STR_PARAM, &count_param.s}, i didn'd find in the code any use for the last. -- juha

2 1

has "defined" been documented somewhere?
by Juha Heinanen 05 Sep '14

05 Sep '14

defined($avp(foo)) can be used to test if avp $avp(foo) has been defined. i looked through kamailio wiki, but didn't find defined function documented anywhere. -- juha

2 1

ldap returning attributes list
by Virmantas Variakojis 05 Sep '14

05 Sep '14

Hi, I'm trying to fetch attribute list with kamailio ldap_search: ldap_search("ldap://agents/ou=Xxxxx,dc=xxxx,dc=yyyyyy,dc=zz?memberOf?sub?(sAMAccountName=$fU)"); ldap_result("memberOf/$avp(s:ldapmemberof1)"); xlog("L_INFO", "LDAP $fU memberOf = $avp(s:ldapmemberof1)"); if (ldap_result_next()) { xlog("L_INFO", "if loop ldap_result"); ldap_result("memberOf/$avp(s:ldapmemberof2)"); xlog("L_INFO", "LDAP $fU memberOf = $avp(s:ldapmemberof2)"); } There's must be seven returning attributes list of memberOf, but I can only retrieve only one of them (looks like last of seven). First xlog output log into my log file, but if loop didn't start. What I'm doing wrong?

3 2

create loop for REGISTER SEND in Kamailio.cfg
by ANTHONY HESNAUX 05 Sep '14

05 Sep '14

hello everybody, I would like loop in kamailio.cfg because I want register soft phone on all asterisk system so I have this configuration but also if I want add another asterisk system my Kamailio.cfg #!define WITH_ASTERISK #!define WITH_CLUST_AST #!ifdef WITH_ASTERISK asterisk.bindip = "asterisk.home.fr" desc "Asterisk IP Address" asterisk.bindport = "5080" desc "Asterisk Port" kamailio.bindip = « kamailio.home.fr" desc "kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif #!ifdef WITH_CLUST_AST #!define HOST 2 asterisk1.bindip = "192.168.0.10" desc « Asterisk - 1 - IP Address" asterisk1.bindport = "5080" desc "Asterisk Port" asterisk2.bindip = "192.168.0.20" desc « Asterisk - 2 - IP Address" asterisk2.bindport = "5080" desc "Asterisk Port" #!endif # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk1.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk1.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); $var(rip) = $sel(cfg_get.asterisk2.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk2.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } #!endif thank you for your help anthony

2 1

Dispatcher algorithm 9 ignores weight=xx attribute
by Alejandro Recarey 05 Sep '14

05 Sep '14

Hi, I have a kamailio with the dispatcher module setup. I use: ds_select_dst("1", "9"); to load balance between 3 gateways. My database looks like this: id setid destination flags priority attrs 5 1 sip:1.1.1.1 0 0 weight=10 8 1 sip:1.1.1.2 0 0 weight=45 9 1 sip:1.1.1.3 0 0 weight=45 As I understand it, most of the calls should be split evenly between gateways 8 and 9, and some calls (around 10%) going to gateway 5. Well, gateway 8 gets 99% of the calls. I am using kamailio 4.1 and I really cannot understand why this is happening. Running ds_list shows me that all gateays and weights are correctly parsed into kamailio. This is the result of my ds_list: SET:: 1 URI:: sip:1.1.1.1 flags=AP priority=0 attrs=weight=10 URI:: sip:1.1.1.2 flags=AP priority=0 attrs=weight=45 URI:: sip:1.1.1.3 flags=AP priority=0 attrs=weight=45 I have tried changing the priority to 100, or to 1 in every gateway, and the result is still the same. I have tried adding ; after the weight parameter, the colon is gone in the ds_list once kamailio parses it. If, keeping the same database, I use ds_select_dst("1", "0") the calls are split evenly across all 3 gateways. This is great, but not what I want, but it at least proves that something is working. Is there anything else I can try? Thanks a lot for the help! Alex

2 1

OT: RTCP logging
by Daniel Tryba 04 Sep '14

04 Sep '14

Slightly offtopic, but for QA reasons I'm interested in logging RTCP reports from peers that send them. It looks like rtpproxy-ng might be used to do this (I'm already using rtpproxy anyway). Is someone already doing this and willing to share some pointers?

3 4

Looking for some help
by Sharan Harkisoon 04 Sep '14

04 Sep '14

I am in need of a Kamailio expert that has some availability for consulting services (remote is fine). Feel free to contact me for details. Thanks, Sharan Harkisoon

2 1

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