- sr-users - kamailio.org

functions for modifying pAsserted
by Alex rsm 27 May '10

27 May '10

Hi, I couldn't find the module that has the functions for modifying pAsserted field. Your help would be appreciated. Thanks, AR _________________________________________________________________ Hotmail has tools for the New Busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:…

3 2

Siremis 1.0 login fails (or, does nothing)
by Joe Micciche 27 May '10

27 May '10

RHEL5.5 server Kamailio 3.0 built from source via Git Apache/2.2.3 PHP 5.3.2 MySQL 5.1.47 chown -R apache /var/www/html/siremis-1.0.0, apache is the httpd user. Used the instructions from Asipto's site for Siremis install and configure, added the alias to httpd.conf, Config.xml is: <?xml version="1.0" standalone="no"?> <Config> <DataSource> <Database Name="Default" Driver="Pdo_Mysql" Server="127.0.0.1" DBName="openser" User="openser" Password="*****" /> <Database Name="Siremis" Driver="Pdo_Mysql" Server="127.0.0.1" DBName="openser" User="openser" Password="*****" /> </DataSource> </Config> I navigate to the Siremis page, http://10.5.10.21/siremis/ and get the login screen, enter the login credentials, Submit, and nothing happens. httpd/error_log: [Wed May 26 12:49:45 2010] [error] [client 10.3.226.168] File does not exist: /var/www/html/siremis-1.0.0/siremis-web/js/sound.js, referer: http://10.5.10.21/siremis/bin/ [Wed May 26 12:49:45 2010] [error] [client 10.3.226.168] File does not exist: /var/www/html/siremis-1.0.0/siremis-web/js/sound.js, referer: http://10.5.10.21/siremis/bin/ I have wiped Siremis and done a fresh install, restarted httpd, and still nothing. Does anyone have ideas or suggestions on where else to look? -- ================================================================== Joe Micciche jmiccich(a)redhat.com Red Hat, Inc. http://www.redhat.com Senior Communications Engineer X(81) 44554 +1.919.754.4554 ==================================================================

6 15

t_check_trans return false on CANCEL
by Ernest Mavrel 27 May '10

27 May '10

3 2

MAX_BRANCHES
by Alex Balashov 27 May '10

27 May '10

This may seem like an absurd question, but is there a particular reason why MAX_BRANCHES is #defined as 12 in config.h? Any particular reason for 12? Is it okay to make it higher? -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

4 5

prefix with kamailio
by alexis heron 27 May '10

27 May '10

hello I want to absorb a number with kamailio just to make out only three but I can not. In fact I send 1941 kamailio and i spring 941 with kamailio so I absorb a number. Here is how I : if ($rU=~"^1+941$") { log(1, "kamailio vers CCM 9xx"); route(10); } route[10] { log(1, "In route[10]"); rewritehostport("142.145.98.138:5060"); append_hf("P-hint: redirection CCM\r\n"); xlog("L_INFO", "$rm from $fu to $tu"); t_on_failure("2"); if (!t_relay()) { sl_reply_error(); }; exit; } Can someone help me?? best regards

3 4

Question about inbound Alias Routing
by dotnetdub 26 May '10

26 May '10

Hi List, At the moment I use dbalias to send inbound from the PSTN <--> subscribers. I have a new opportunity that requires me to send inbound to a URI - SIP/353xxxxxxx(a)customer.proxy - not really sure how to accomplish this. I have a feeling there is a module I should be looking at to do this possibly? If someone could point me in the correct direction. TIA. Stephen.

2 1

A Couple of Questions about carrierroute moduel
by JR Richardson 26 May '10

26 May '10

Hi All, I'm testing carrierroute module on version: kamailio 1.5.4-notls (i386/linux). I'm distributing calls to a group of Asterisk servers, all in the same carrier and domain, wanted to load balance the calls and I'm curious how this works. I see the 'probability' parameter in the carrierroute table and set it to equal value across all servers, I have 4 servers I'm sending calls to, so I set each prob value to '0.25'. I'm testing with sipp and hashing over call_id so this is the same for every call. I send 40 calls and the distribution of calls to the asterisk servers are 11, 12, 6, 11, pretty close to even but not 100%. How does the distribution algorithm work for carrieroute? Does it statefully track all active calls and load balance across each server in destination group? Is there another algorithm that can offer that can get closer to a round robin effect? Another question is do I need to include the record_route() and the t_check_trans() functions prior to calling the carrierroute functions? I see in the debug "DBG:core:forward_reply: found module tm, passing reply to it" and dialogs seem to process fine, so does the carrierroute module handle those requirements for stateful processing? Thanks. JR -- JR Richardson Engineering for the Masses

2 2

Routing from Asterisk to Microsoft UM for voicemail.
by Nelson Pereira 26 May '10

26 May '10

If someone has a known good working route configuration for use with Microsoft Unified Messaging, please post your config. Thanks Nelson Pereira

2 1

Calculate ha1 with mysql trigger
by Ole Kaas 26 May '10

26 May '10

Hello, When editing the subscriber table using phpmyadmin, updating the ha1 fields are no fun and error prone. The trigger below will calculate the hashes if the input fields are left empty. Navigate to the subscriber table in phpmyadmin and drop the code in the query field under the "SQL" tab. Remeber to set the delimiter to // /Ole DROP TRIGGER IF EXISTS calcha1 // CREATE TRIGGER calcha1 BEFORE INSERT ON subscriber FOR EACH ROW BEGIN IF NEW.ha1 = "" THEN SET NEW.ha1 = MD5(CONCAT(NEW.username,":",NEW.domain,":",NEW.password)); END IF; IF NEW.ha1b = "" THEN SET NEW.ha1b = MD5(CONCAT(NEW.username,"@",NEW.domain,":",NEW.domain,":",NEW.password)); END IF; END//

4 3

'serctl add <name> <password> <email>'
by Sanjeev Singh 26 May '10

26 May '10

Hi All, Please help me this i am using below command to add subscribers in the ser so that i can make a phone call. 1) what is the meaning of this command 2) how to use this command? *'serctl add <name> <password> <email>'* can you please help me on this also. 1) how to register my sip client with ser server 2) how to add subscribers so that i can make a phone call. Regards, Sanjeev Singh

1 0

permissions module
by Asim Riaz 25 May '10

25 May '10

Hello List, from the documentation, permissions.allow, allows the calls either by matching From-URI or R-URI and permission.deny denies the call from matching uri. Do I need to divide the file into different sections, for R-URI and for From-URI? Not sure exact syntax could someone explain me that if I want to match in permissions.allow if a phone 10.10.10.10 dial(r-uri) 1234 allow it and everything else is blocked? What is the preference does kamailio check for permissions.deny first if there is no matching then checks permissions.allow? Thanks in advance, Asim

2 1

Re: [SR-Users] sipp and stateful transaction problem
by Daniel-Constantin Mierla 25 May '10

25 May '10

Hello, On 5/22/10 2:22 AM, JR Richardson wrote: > On Fri, May 21, 2010 at 4:46 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com> wrote: > >> Hello, >> >> On 5/21/10 10:47 PM, JR Richardson wrote: >> >>> Hi All, >>> >>> I'm doing some testing with kamailio 1.5: >>> >>> kamailio1:/etc/kamailio# kamailio -V >>> version: kamailio 1.5.4-notls (i386/linux) >>> flags: STATISTICS, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, >>> SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, >>> MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4194304 >>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >>> svnrevision: 2:6005M >>> @(#) $Id: main.c 5608 2009-02-13 16:48:17Z henningw $ >>> main.c compiled on 10:14:11 May 18 2010 with gcc 4.3.2 >>> >>> Using dispatcher module trying to load balance SIP calls across some >>> Asterisk servers. I have it working fine when I test in this >>> scenario: >>> >>> sip phone dial out><asterisk><kamailio><round robin to several asterisk >>> servers >>> >>> This works stateful and stateless, handles everything gracefully. >>> >>> This scenario is giving me fits: >>> >>> sipp dial out><kamailio><round robin to several asterisk servers >>> >>> I get retransmits on every call back to sipp with errors like >>> "Discarding message which can't be mapped to a known SIPp call" and >>> "SIP/2.0 481 Call leg/transaction does not exist" >>> >>> This happens with kamailio setup stateful or stateless. I'm wondering >>> if sipp is the problem or just doesn't play well with kamailio? >>> >>> I've kept the config as simple as possible for testing, it is listed >>> here http://pastebin.com/BZ8hJvJv >>> >>> Here is my sipp usage: >>> >>> sipp -sn uac 10.10.12.53 -i 10.10.14.97 -s 55 -d 7000 -l 10 -r 1 >>> -trace_err >>> >>> Any insight would be appriciated. >>> >>> >>> >> the problem is in your sipp scenario. The uac calls do not map to uas. >> kamailio does not reply 481, check the uas scenario, that is the one that >> sends back the 481. >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla >> Kamailio (OpenSER) Advanced Training >> Miami, Fl, USA - June 21-23, 2010 >> http://www.asipto.com/index.php/kamailio-advanced-training/ >> >> >> > Thanks Daniel, I reveiwed the sipp docs, '-sn uas' just sits there as > a responder, it will not initiate calls to kamailio. I don't > understand what you are getting at? How would I use this type of > scenario to test? > keeping the mailing list cc-ed is recommended, since others can respond faster and new people can benefit of the discussion. What I wanted to say is that kamailio does not reply 481. So the problem is in the responder of requests sent by UAC and forwarded by Kamailio. Somehow, the dialog is destroyed before the BYE (or other in-dialog request) is sent by UAS. If you can grab the SIP trace of such call (e.g., using ngrep on kamailio server), I can give more hits (try to select the sip flow for one such call only, sending full sip trace will be too big). Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

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Re: [SR-Users] kamailio 3.0.2 for Thu, May 27, 2010
by Daniel-Constantin Mierla 25 May '10

25 May '10

On 5/25/10 8:51 PM, SIP wrote: > Daniel-Constantin Mierla wrote: > >> Hello, >> >> I am planning to release kamailio 3.0.2 this Thursday. There were some >> fixes since 3.0.1 that worth to be packaged. If you have major reports >> for current stable version, please write to sr-dev(a)lists.sip-router.org >> >> Cheers, >> Daniel >> >> > Is this a drop-in upgrade from 3.0.1 (i.e. recompile modules and core > and keep the rest in place)? > > yes, changes to last number in version string mean that config language and database structure are the same. This is a minor release, bringing only fixes to the code. So upgrading from 3.0.0 or 3.0.1 to 3.0.2 is exactly what you said, just recompile and reinstall modules and core, keep the rest in place. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

1 0

could not bind to the TM-Module
by Javier Menendez 25 May '10

25 May '10

Hi, I have been configuring an Kamailio 3.0.1, with the dispatcher module. Everything seems to work just fine but when a gateway is in Probing mode it remains in that state for ever. checking the logs I found some weird messages, I think that is the rrot cause of the problem but I have no clue how to solve it. These are the messages I mentioned before May 21 14:56:34 localhost /usr/local/sbin/kamailio[21539]: DEBUG: <core> [sr_module.c:516]: find_export_record: <load_tm> not found May 21 14:56:34 localhost /usr/local/sbin/kamailio[21539]: WARNING: dispatcher [dispatcher.c:358]: could not bind to the TM-Module, automatic re-activation disabled. May 21 14:56:34 localhost /usr/local/sbin/kamailio[21539]: DEBUG: <core> [sr_module.c:871]: DEBUG: init_mod: sl May 21 14:56:34 localhost /usr/local/sbin/kamailio[21539]: DEBUG: <core> [statistics.c:105]: statistics manager successfully initialized May 21 14:56:34 localhost /usr/local/sbin/kamailio[21539]: DEBUG: <core> [md5utils.c:87]: DEBUG: MD5 calculated: f8f2ab2c1295e90ed7dbb499b30f44b2 May 21 14:56:34 localhost /usr/local/sbin/kamailio[21539]: DEBUG: <core> [sr_module.c:506]: find_export_record: found <load_tm> in module tm [/usr/local/lib/kamailio/modules/tm.so] As you may see load_tm is not found in the dispatcher but it is found in md5utils?? These are the modules included mpath="/usr/local/lib/kamailio/modules/:/usr/local/lib/kamailio/modules_k/" loadmodule "tm.so" loadmodule "dispatcher.so" loadmodule "sl.so" loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "permissions.so" loadmodule "drouting.so Any clue?

2 3

could not open module kamailio
by Bruno Bresciani 25 May '10

25 May '10

I compiled the route.so module to the kamailio but when I start kamailio the follow error is generated in the log file: "ERROR:core:sr_load_module: could not open module </home2/local/kamailio/lib/kamailio/modules/route.so>: /home2/local/kamailio/lib/kamailio/modules/route.so: undefined symbol: add_hash" Someone can help me? Cheers, Bruno

2 1

kamailio 3.0.1 rpm for centos 5.5
by Daniel-Constantin Mierla 25 May '10

25 May '10

Hello, for those interested, kamailio 3.0.1 rpm packages for centos 5.5 are available for download at: http://www.kamailio.org/pub/kamailio/3.0.1/packages/centos-5.5/ Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

1 0

R: SIP Router devel meeting, Berlin, June 8, 2010
by Daniel-Constantin Mierla 25 May '10

25 May '10

Hello, a short reminder for those that want to participate to SIP Router devel meeting and haven't been in contact with me yet, you need to send email to: *registration(a)lists.sip-router.org* See more details at: http://sip-router.org/2010/05/12/sip-router-devel-meeting-june-8-2010-berli… Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

1 0

no Contact in REFER generated with dlg_bridge
by Anton Roman 25 May '10

25 May '10

Hi all, I'm trying to implement a Click2Dial service by using the dlg_bridge command from the dialog module. Although it works in this case, I found two problems in the scenario and I would like to read your opinion about them. 1) Fisrt problem: the REFER generated by Kamailio doesn't contain a contact header. As per RFC 3515, this request should have a Contact header. This REFER is not being accepted by a proprietary device. This is can be worked around with the append_hf() command from the textops module, but I don know if it is a good solution. 2) The second one arises when the ACK generated by Kamailio, goes through Kamailio (caller and calle involved in click2dial are registered on the Kamailio server): the branch of the second via header is not correctly populated. Its content is '0' when it must be tha same as in the INVITE. Below you can find the traces of the whole scenario. Could you take a look at them and give me your feedback? Thanks in advance, best regards Antón Román T +8.969957 127.0.0.1:44785 -> 127.0.0.1:5060 [AP] POST HTTP/1.0 User-Agent: PEAR XML_RPC Host: 127.0.0.1 Content-Type: text/xml Content-Length: 331 <?xml version="1.0" encoding="UTF-8"?> <methodCall> <methodName>mi_fifo</methodName> <params> <param> <value><string>dlg_bridge</string></value> </param> <param> <value><string>sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103></string></value> </param> <param> <value><string>sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103></string></value> </param> </params> </methodCall> ## U +0.001779 10.1.20.103:5060 -> 10.1.20.103:5060 INVITE sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> SIP/2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.21dbafc4.0 To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd CSeq: 10 INVITE Call-ID: 3d3d4ede-7172(a)127.0.0.1 Content-Length: 147 User-Agent: kamailio (3.0.0 (i386/linux)) Contact: <sip:kamailio.org:5060> Content-Type: application/sdp v=0 o=kamailio-bridge 0 0 IN IP4 0.0.0.0 s=kamailio c=IN IP4 0.0.0.0 t=0 0 m=audio 9 RTP/AVP 8 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 # U +0.001330 10.1.20.103:5060 -> 10.1.20.103:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.21dbafc4.0 To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd CSeq: 10 INVITE Call-ID: 3d3d4ede-7172(a)127.0.0.1 Server: kamailio (3.0.0 (i386/linux)) Content-Length: 0 Warning: 392 10.1.20.103:5060 "Noisy feedback tells: pid=7160 req_src_ip=10.1.20.103 req_src_port=5060 in_uri=sip:111@10.1.20.103<sip%3A111(a)10.1.20.103>out_uri= sip:111@10.1.3.15:5061 via_cnt==1" # U +0.000727 10.1.20.103:5060 -> 10.1.3.15:5061 INVITE sip:111@10.1.3.15:5061 SIP/2.0 Max-Forwards: 10 Record-Route: <sip:10.1.20.103;lr=on;did=768.90707151> Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.31dbafc4.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.21dbafc4.0 To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd CSeq: 10 INVITE Call-ID: 3d3d4ede-7172(a)127.0.0.1 Content-Length: 147 User-Agent: kamailio (3.0.0 (i386/linux)) Contact: <sip:kamailio.org:5060> Content-Type: application/sdp v=0 o=kamailio-bridge 0 0 IN IP4 0.0.0.0 s=kamailio c=IN IP4 0.0.0.0 t=0 0 m=audio 9 RTP/AVP 8 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 # T +0.000442 127.0.0.1:5060 -> 127.0.0.1:44785 [AP] HTTP/1.0 200 OK Via: SIP/2.0/TCP 127.0.0.1:44785 Server: kamailio (3.0.0 (i386/linux)) Content-Length: 211 Warning: 392 127.0.0.1:5060 "Noisy feedback tells: pid=7172 req_src_ip=127.0.0.1 req_src_port=44785 in_uri=sip:127.0.0.1:9 out_uri=sip: 127.0.0.1:9 via_cnt==1" <?xml version="1.0"?> <methodResponse> <params> <param> <value><array><data> <value><string>200 OK </string></value> <value><string> </string></value> </data></array></value> </param> </params> </methodResponse> ##### U +0.021063 10.1.3.15:5061 -> 10.1.20.103:5060 SIP/2.0 100 Trying To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 10 INVITE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.31dbafc4.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.21dbafc4.0 Record-Route: <sip:10.1.20.103;lr=on;did=768.90707151> Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.030585 10.1.3.15:5061 -> 10.1.20.103:5060 SIP/2.0 180 Ringing To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 10 INVITE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.31dbafc4.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.21dbafc4.0 Record-Route: <sip:10.1.20.103;lr=on;did=768.90707151> Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.000384 10.1.20.103:5060 -> 10.1.20.103:5060 SIP/2.0 180 Ringing To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 10 INVITE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.21dbafc4.0 Record-Route: <sip:10.1.20.103;lr=on;did=768.90707151> Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +2.440228 10.1.3.15:5061 -> 10.1.20.103:5060 SIP/2.0 200 OK To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 10 INVITE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.31dbafc4.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.21dbafc4.0 Record-Route: <sip:10.1.20.103;lr=on;did=768.90707151> Contact: "111" <sip:111@10.1.3.15:5061> Server: Linksys/SPA941-5.1.5 Content-Length: 200 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 761338 761338 IN IP4 10.1.3.15 s=- c=IN IP4 10.1.3.15 t=0 0 m=audio 16434 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv # U +0.000315 10.1.20.103:5060 -> 10.1.20.103:5060 SIP/2.0 200 OK To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 10 INVITE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.21dbafc4.0 Record-Route: <sip:10.1.20.103;lr=on;did=768.90707151> Contact: "111" <sip:111@10.1.3.15:5061> Server: Linksys/SPA941-5.1.5 Content-Length: 200 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 761338 761338 IN IP4 10.1.3.15 s=- c=IN IP4 10.1.3.15 t=0 0 m=audio 16434 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv # U +0.001190 10.1.20.103:5060 -> 10.1.20.103:5060 ACK sip:111@10.1.3.15:5061 SIP/2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.21dbafc4.0 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 CSeq: 10 ACK Route: <sip:10.1.20.103;lr=on;did=768.90707151> User-Agent: kamailio (3.0.0 (i386/linux)) Content-Length: 0 # U +0.001249 10.1.20.103:5060 -> 10.1.3.15:5061 ACK sip:111@10.1.3.15:5061 SIP/2.0 Max-Forwards: 10 *Via: SIP/2.0/UDP 10.1.20.103;branch=0* Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.21dbafc4.0 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 CSeq: 10 ACK User-Agent: kamailio (3.0.0 (i386/linux)) Content-Length: 0 # U +0.000418 10.1.20.103:5060 -> 10.1.20.103:5060 REFER sip:111@10.1.3.15:5061 SIP/2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.0e0719e2.0 To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd CSeq: 11 REFER Call-ID: 3d3d4ede-7172(a)127.0.0.1 Route: <sip:10.1.20.103;lr=on;did=768.90707151> Content-Length: 0 User-Agent: kamailio (3.0.0 (i386/linux)) Referred-By: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> Refer-To: sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103> * No Contact!*! # U +0.005133 10.1.20.103:5060 -> 10.1.3.15:5061 REFER sip:111@10.1.3.15:5061 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.1e0719e2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.0e0719e2.0 To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd CSeq: 11 REFER Call-ID: 3d3d4ede-7172(a)127.0.0.1 Content-Length: 0 User-Agent: kamailio (3.0.0 (i386/linux)) Referred-By: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> Refer-To: sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103> # U +0.046925 10.1.3.15:5061 -> 10.1.20.103:5060 INVITE sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103> SIP/2.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-9ea2814a From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>> Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 101 INVITE Max-Forwards: 70 Contact: "111" <sip:111@10.1.3.15:5061> Expires: 240 Referred-By: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> User-Agent: Linksys/SPA941-5.1.5 Content-Length: 389 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 761586 761586 IN IP4 10.1.3.15 s=- c=IN IP4 10.1.3.15 t=0 0 m=audio 16434 RTP/AVP 8 0 2 4 18 96 97 98 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv # U +0.000520 10.1.20.103:5060 -> 10.1.3.15:5061 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-9ea2814a From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103> >;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.d44b Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 101 INVITE Proxy-Authenticate: Digest realm="10.1.20.103", nonce="4b7179d300000000c65419875a5dfc25067a1fc4b173b813" Server: kamailio (3.0.0 (i386/linux)) Content-Length: 0 Warning: 392 10.1.20.103:5060 "Noisy feedback tells: pid=7162 req_src_ip=10.1.3.15 req_src_port=5061 in_uri=sip:112@10.1.20.103<sip%3A112(a)10.1.20.103>out_uri= sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103> via_cnt==1" # U +0.004377 10.1.3.15:5061 -> 10.1.20.103:5060 SIP/2.0 202 Accepted To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 11 REFER Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.1e0719e2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.0e0719e2.0 Contact: "111" <sip:111@10.1.3.15:5061> Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.000193 10.1.20.103:5060 -> 10.1.20.103:5060 SIP/2.0 202 Accepted To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 11 REFER Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.0e0719e2.0 Contact: "111" <sip:111@10.1.3.15:5061> Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.000286 10.1.20.103:5060 -> 10.1.20.103:5060 BYE sip:111@10.1.3.15:5061 SIP/2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.2e0719e2.0 To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd CSeq: 11 BYE Call-ID: 3d3d4ede-7172(a)127.0.0.1 Route: <sip:10.1.20.103;lr=on;did=768.90707151> Content-Length: 0 User-Agent: kamailio (3.0.0 (i386/linux)) # U +0.003618 10.1.20.103:5060 -> 10.1.3.15:5061 BYE sip:111@10.1.3.15:5061 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.3e0719e2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.2e0719e2.0 To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd CSeq: 11 BYE Call-ID: 3d3d4ede-7172(a)127.0.0.1 Content-Length: 0 User-Agent: kamailio (3.0.0 (i386/linux)) # U +0.005223 10.1.3.15:5061 -> 10.1.20.103:5060 ACK sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103> SIP/2.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-9ea2814a From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103> >;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.d44b Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 101 ACK Max-Forwards: 70 Contact: "111" <sip:111@10.1.3.15:5061> User-Agent: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.010954 10.1.3.15:5061 -> 10.1.20.103:5060 INVITE sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103> SIP/2.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-ee6fa05e From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>> Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="111",realm="10.1.20.103",nonce="4b7179d300000000c65419875a5dfc25067a1fc4b173b813",uri=" sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103> ",algorithm=MD5,response="2baeb40875c0cc7040857daa6e8192fa" Contact: "111" <sip:111@10.1.3.15:5061> Expires: 240 Referred-By: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> User-Agent: Linksys/SPA941-5.1.5 Content-Length: 389 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 761586 761586 IN IP4 10.1.3.15 s=- c=IN IP4 10.1.3.15 t=0 0 m=audio 16434 RTP/AVP 8 0 2 4 18 96 97 98 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv # U +0.001353 10.1.20.103:5060 -> 10.1.3.15:5061 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-ee6fa05e From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>> Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 102 INVITE Server: kamailio (3.0.0 (i386/linux)) Content-Length: 0 Warning: 392 10.1.20.103:5060 "Noisy feedback tells: pid=7159 req_src_ip=10.1.3.15 req_src_port=5061 in_uri=sip:112@10.1.20.103<sip%3A112(a)10.1.20.103>out_uri= sip:112@10.1.3.18:5061 via_cnt==1" # U +0.000226 10.1.20.103:5060 -> 10.1.3.18:5061 INVITE sip:112@10.1.3.18:5061 SIP/2.0 Record-Route: <sip:10.1.20.103;lr=on;did=a4d.ef1941a3> Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bK7d0c.d071633.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-ee6fa05e From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>> Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 102 INVITE Max-Forwards: 69 Contact: "111" <sip:111@10.1.3.15:5061> Expires: 240 Referred-By: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> User-Agent: Linksys/SPA941-5.1.5 Content-Length: 389 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 761586 761586 IN IP4 10.1.3.15 s=- c=IN IP4 10.1.3.15 t=0 0 m=audio 16434 RTP/AVP 8 0 2 4 18 96 97 98 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv # U +0.007697 10.1.3.15:5061 -> 10.1.20.103:5060 SIP/2.0 500 Internal Server Error To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 11 BYE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.3e0719e2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.2e0719e2.0 Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.000280 10.1.20.103:5060 -> 10.1.20.103:5060 SIP/2.0 500 Internal Server Error To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 11 BYE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.2e0719e2.0 Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.012355 10.1.3.18:5061 -> 10.1.20.103:5060 SIP/2.0 100 Trying To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>> From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bK7d0c.d071633.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-ee6fa05e Record-Route: <sip:10.1.20.103;lr=on;did=a4d.ef1941a3> Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.034919 10.1.3.18:5061 -> 10.1.20.103:5060 SIP/2.0 180 Ringing To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>>;tag=d091cd3cd91903c5i1 From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bK7d0c.d071633.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-ee6fa05e Record-Route: <sip:10.1.20.103;lr=on;did=a4d.ef1941a3> Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.000260 10.1.20.103:5060 -> 10.1.3.15:5061 SIP/2.0 180 Ringing To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>>;tag=d091cd3cd91903c5i1 From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-ee6fa05e Record-Route: <sip:10.1.20.103;lr=on;did=a4d.ef1941a3> Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.032160 10.1.3.15:5061 -> 10.1.20.103:5060 NOTIFY sip:kamailio.org:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-77e77cb8 From: <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>>;tag=d45e4a92cc93b70fi1 To: <sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> >;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 101 NOTIFY Max-Forwards: 70 Route: <sip:10.1.20.103;lr=on;did=768.90707151> Event: refer User-Agent: Linksys/SPA941-5.1.5 Subscription-State: active Content-Length: 21 Content-Type: message/sipfrag;version=2.0 SIP/2.0 180 Ringing # U +0.002343 10.1.20.103:5060 -> 193.22.119.66:5060 NOTIFY sip:kamailio.org:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bK2455.f8178c15.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-77e77cb8 From: <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>>;tag=d45e4a92cc93b70fi1 To: <sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> >;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 101 NOTIFY Max-Forwards: 69 Event: refer User-Agent: Linksys/SPA941-5.1.5 Subscription-State: active Content-Length: 21 Content-Type: message/sipfrag;version=2.0 SIP/2.0 180 Ringing # U +0.461009 10.1.20.103:5060 -> 193.22.119.66:5060 NOTIFY sip:kamailio.org:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bK2455.f8178c15.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-77e77cb8 From: <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>>;tag=d45e4a92cc93b70fi1 To: <sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> >;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 101 NOTIFY Max-Forwards: 69 Event: refer User-Agent: Linksys/SPA941-5.1.5 Subscription-State: active Content-Length: 21 Content-Type: message/sipfrag;version=2.0 SIP/2.0 180 Ringing # U +0.028104 10.1.3.15:5061 -> 10.1.20.103:5060 NOTIFY sip:kamailio.org:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-77e77cb8 From: <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>>;tag=d45e4a92cc93b70fi1 To: <sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> >;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 101 NOTIFY Max-Forwards: 70 Route: <sip:10.1.20.103;lr=on;did=768.90707151> Event: refer User-Agent: Linksys/SPA941-5.1.5 Subscription-State: active Content-Length: 21 Content-Type: message/sipfrag;version=2.0 SIP/2.0 180 Ringing # U +0.760273 10.1.3.18:5061 -> 10.1.20.103:5060 SIP/2.0 200 OK To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>>;tag=d091cd3cd91903c5i1 From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bK7d0c.d071633.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-ee6fa05e Record-Route: <sip:10.1.20.103;lr=on;did=a4d.ef1941a3> Contact: "112" <sip:112@10.1.3.18:5061> Server: Linksys/SPA941-5.1.5 Content-Length: 200 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 761170 761170 IN IP4 10.1.3.18 s=- c=IN IP4 10.1.3.18 t=0 0 m=audio 16464 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv # U +0.000246 10.1.20.103:5060 -> 10.1.3.15:5061 SIP/2.0 200 OK To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>>;tag=d091cd3cd91903c5i1 From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-ee6fa05e Record-Route: <sip:10.1.20.103;lr=on;did=a4d.ef1941a3> Contact: "112" <sip:112@10.1.3.18:5061> Server: Linksys/SPA941-5.1.5 Content-Length: 200 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 761170 761170 IN IP4 10.1.3.18 s=- c=IN IP4 10.1.3.18 t=0 0 m=audio 16464 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv # U +0.022157 10.1.3.15:5061 -> 10.1.20.103:5060 ACK sip:112@10.1.3.18:5061 SIP/2.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-a8e0a92a From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>>;tag=d091cd3cd91903c5i1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 102 ACK Max-Forwards: 70 Route: <sip:10.1.20.103;lr=on;did=a4d.ef1941a3> Proxy-Authorization: Digest username="111",realm="10.1.20.103",nonce="4b7179d300000000c65419875a5dfc25067a1fc4b173b813",uri=" sip:112@10.1.3.18:5061 ",algorithm=MD5,response="3d72eeb2b6b5b2c161685ac3435b657c" Contact: "111" <sip:111@10.1.3.15:5061> User-Agent: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.009321 10.1.20.103:5060 -> 10.1.3.18:5061 ACK sip:112@10.1.3.18:5061 SIP/2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-a8e0a92a From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>>;tag=d091cd3cd91903c5i1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 102 ACK Max-Forwards: 69 Proxy-Authorization: Digest username="111",realm="10.1.20.103",nonce="4b7179d300000000c65419875a5dfc25067a1fc4b173b813",uri=" sip:112@10.1.3.18:5061 ",algorithm=MD5,response="3d72eeb2b6b5b2c161685ac3435b657c" Contact: "111" <sip:111@10.1.3.15:5061> User-Agent: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.447617 10.1.3.18:5061 -> 10.1.20.103:5060 BYE sip:111@10.1.3.15:5061 SIP/2.0 Via: SIP/2.0/UDP 10.1.3.18:5061;branch=z9hG4bK-73a35564 From: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>>;tag=d091cd3cd91903c5i1 To: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 101 BYE Max-Forwards: 70 Route: <sip:10.1.20.103;lr=on;did=a4d.ef1941a3> User-Agent: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.007101 10.1.20.103:5060 -> 10.1.3.15:5061 BYE sip:111@10.1.3.15:5061 SIP/2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKad0c.d0bfffc.0 Via: SIP/2.0/UDP 10.1.3.18:5061;branch=z9hG4bK-73a35564 From: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>>;tag=d091cd3cd91903c5i1 To: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 101 BYE Max-Forwards: 69 User-Agent: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.014754 10.1.3.15:5061 -> 10.1.20.103:5060 SIP/2.0 200 OK To: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 From: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>>;tag=d091cd3cd91903c5i1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 101 BYE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKad0c.d0bfffc.0 Via: SIP/2.0/UDP 10.1.3.18:5061;branch=z9hG4bK-73a35564 Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.000198 10.1.20.103:5060 -> 10.1.3.18:5061 SIP/2.0 200 OK To: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 From: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>>;tag=d091cd3cd91903c5i1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 101 BYE Via: SIP/2.0/UDP 10.1.3.18:5061;branch=z9hG4bK-73a35564 Server: Linksys/SPA941-5.1.5 Content-Length: 0

3 4

sipp and stateful transaction problem
by JR Richardson 25 May '10

25 May '10

Hi All, I'm doing some testing with kamailio 1.5: kamailio1:/etc/kamailio# kamailio -V version: kamailio 1.5.4-notls (i386/linux) flags: STATISTICS, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4194304 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: 2:6005M @(#) $Id: main.c 5608 2009-02-13 16:48:17Z henningw $ main.c compiled on 10:14:11 May 18 2010 with gcc 4.3.2 Using dispatcher module trying to load balance SIP calls across some Asterisk servers. I have it working fine when I test in this scenario: sip phone dial out><asterisk><kamailio><round robin to several asterisk servers This works stateful and stateless, handles everything gracefully. This scenario is giving me fits: sipp dial out><kamailio><round robin to several asterisk servers I get retransmits on every call back to sipp with errors like "Discarding message which can't be mapped to a known SIPp call" and "SIP/2.0 481 Call leg/transaction does not exist" This happens with kamailio setup stateful or stateless. I'm wondering if sipp is the problem or just doesn't play well with kamailio? I've kept the config as simple as possible for testing, it is listed here http://pastebin.com/BZ8hJvJv Here is my sipp usage: sipp -sn uac 10.10.12.53 -i 10.10.14.97 -s 55 -d 7000 -l 10 -r 1 -trace_err Any insight would be appriciated. Thanks. JR -- JR Richardson Engineering for the Masses

3 4

Need help with route statement
by Nelson Pereira 24 May '10

24 May '10

Hi all... Well I have made some progress... Bellow is my routing statement: route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); break; }; if (uri==myself) { if (method=="REGISTER") { save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { if (src_ip==10.98.6.5) { if (dst_port==5065) { t_relay_to_tcp("10.98.118.20", "5065"); } else if (dst_port==5066) { t_relay_to_tcp("10.98.118.20", "5066"); } else if (dst_port==5067) { t_relay_to_tcp("10.98.118.20", "5067"); } else { t_relay_to_tcp("10.98.118.20", "5060"); } } else { t_relay_to_udp("10.98.6.5", "5060"); }; } When asterisk sends a call to kamailio, Kamailio then sends the invite to 10.98.118.20 via TCP on port 5061. INVITE sip:1989@10.98.6.5:5061 SIP/2.0 Record-Route: <sip:10.98.6.5:5065;transport=tcp;r2=on;lr=on> Record-Route: <sip:10.98.6.5:5061;r2=on;lr=on> Via: SIP/2.0/TCP 10.98.6.5:5065;branch=z9hG4bK74fd.a4578a84.0 Via: SIP/2.0/UDP 10.98.6.5:5060;branch=z9hG4bK7b9bb22d;rport=5060 From: "1103" <sip:1103@10.98.6.5>;tag=as4ae41ccf To: <sip:1989@10.98.6.5:5061> Contact: <sip:1103@10.98.6.5> Call-ID: 44073b911e86b0a96c9104cb7a5ec389(a)10.98.6.5 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Fri, 21 May 2010 12:23:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 232 P-hint: usrloc applied The problem is that the Kamailio receives a 302 Moved Temporarily with a contact field of CONTACT: <sip:1989@10.98.6.5:5065;transport=TCP> I need to have Kamailio, use this contact field and re-send the invite. How can this be done? Nelson Pereira Senior Network Specialist Protus<http://www.protus.com/> npereira(a)protus.com<mailto:name@protus.com> phone: 613.733.0000 ext.528 MyFax: 613.822.5083 MyFax.com<http://www.myfax.com/> | my1voice.com<http://www.my1voice.com/> | Campaigner.com<http://www.campaigner.com/>

2 4

kamailioctl
by Jon Farmer 21 May '10

21 May '10

Hi I have just install 1.5 and I am trying to setup TLS I ran kamctl tls rootCA and it created ok However the docs at http://www.kamailio.org/docs/tls.htm next say to run kamailioctl tls userCERT However I don't have a kamailioctl on my system. Any ideas anyone? Regards Jon -- Jon Farmer Tel: 07795 118140 Email: viperdudeuk(a)gmail.com Twitter: @viperdudeuk

4 3

ser server does not start
by Sanjeev Singh 21 May '10

21 May '10

hi all, i am a new user of ser. PLease help me to start ser i installed ser rpm v0.9.6 on linux machine but ser is not starting. when i given command /etc/init.d/ser start - it is always stuck at Starting ser: as shown below ------------------- [root@linuxrnd1 ~]# /etc/init.d/ser start Starting ser: ----------------- now when i give command just ser then it is stuck at 0(13589) DEBUG: init_mod_child (1): usrloc as shown below --------------------------------------------------------- [root@linuxrnd1 ~]# ser 0(13589) loading module /usr/lib/ser/modules/sl.so 0(13589) loading module /usr/lib/ser/modules/tm.so 0(13589) loading module /usr/lib/ser/modules/rr.so 0(13589) loading module /usr/lib/ser/modules/maxfwd.so 0(13589) loading module /usr/lib/ser/modules/usrloc.so 0(13589) loading module /usr/lib/ser/modules/registrar.so 0(13589) set_mod_param_regex: usrloc matches module usrloc 0(13589) set_mod_param_regex: found <db_mode> in module usrloc [/usr/lib/ser/modules/usrloc.so] 0(13589) set_mod_param_regex: rr matches module rr 0(13589) set_mod_param_regex: found <enable_full_lr> in module rr [/usr/lib/ser/modules/rr.so] 0(13589) find_export: found <mf_process_maxfwd_header> in module maxfwd [/usr/lib/ser/modules/maxfwd.so] 0(13589) find_export: found <sl_send_reply> in module sl_module [/usr/lib/ser/modules/sl.so] 0(13589) find_export: found <sl_send_reply> in module sl_module [/usr/lib/ser/modules/sl.so] 0(13589) find_export: found <loose_route> in module rr [/usr/lib/ser/modules/rr.so] 0(13589) find_export: found <lookup> in module registrar [/usr/lib/ser/modules/registrar.so] 0(13589) find_export: found <lookup> in module registrar [/usr/lib/ser/modules/registrar.so] 0(13589) find_export: found <sl_send_reply> in module sl_module [/usr/lib/ser/modules/sl.so] 0(13589) find_export: found <t_relay> in module tm [/usr/lib/ser/modules/tm.so] 0(13589) find_export: found <sl_reply_error> in module sl_module [/usr/lib/ser/modules/sl.so] 0(13589) find_export: found <save> in module registrar [/usr/lib/ser/modules/registrar.so] 0(13589) find_export: found <sl_reply_error> in module sl_module [/usr/lib/ser/modules/sl.so] 0(13589) routing table 0: 0(13589) if ( 0(13589) NOT( 0(13589) external_module_call( 0(13589) f_ptr<0x4f3db0> 0(13589) , "10" 0(13589) ); 0(13589) ) 0(13589) ) { 0(13589) external_module_call( 0(13589) f_ptr<0x692790> 0(13589) , "483" 0(13589) , "Too Many Hops" 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) if ( 0(13589) UNKNOWN 0(13589) > 0(13589) 65535 0(13589) ) { 0(13589) external_module_call( 0(13589) f_ptr<0x692790> 0(13589) , "513" 0(13589) , "Message Overflow" 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) if ( 0(13589) external_module_call( 0(13589) f_ptr<0x830c30> 0(13589) ); 0(13589) ) { 0(13589) route( 0(13589) 1 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) if ( 0(13589) uri 0(13589) != 0(13589) _myself_ 0(13589) ) { 0(13589) route( 0(13589) 1 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) if ( 0(13589) method 0(13589) == 0(13589) "ACK" 0(13589) ) { 0(13589) route( 0(13589) 1 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) if ( 0(13589) method 0(13589) == 0(13589) "REGISTER" 0(13589) ) { 0(13589) route( 0(13589) 2 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) }; 0(13589) external_module_call( 0(13589) f_ptr<0xe86d60> 0(13589) , "aliases" 0(13589) ); 0(13589) if ( 0(13589) uri 0(13589) != 0(13589) _myself_ 0(13589) ) { 0(13589) route( 0(13589) 1 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) if ( 0(13589) NOT( 0(13589) external_module_call( 0(13589) f_ptr<0xe86d60> 0(13589) , "location" 0(13589) ); 0(13589) ) 0(13589) ) { 0(13589) external_module_call( 0(13589) f_ptr<0x692790> 0(13589) , "404" 0(13589) , "User Not Found" 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) route( 0(13589) 1 0(13589) ); 0(13589) if ( 0(13589) UNKNOWN 0(13589) > 0(13589) 65535 0(13589) ) { 0(13589) external_module_call( 0(13589) f_ptr<0x692790> 0(13589) , "513" 0(13589) , "Message Overflow" 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) if ( 0(13589) external_module_call( 0(13589) f_ptr<0x830c30> 0(13589) ); 0(13589) ) { 0(13589) route( 0(13589) 1 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) if ( 0(13589) uri 0(13589) != 0(13589) _myself_ 0(13589) ) { 0(13589) route( 0(13589) 1 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) if ( 0(13589) method 0(13589) == 0(13589) "ACK" 0(13589) ) { 0(13589) route( 0(13589) 1 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) if ( 0(13589) method 0(13589) == 0(13589) "REGISTER" 0(13589) ) { 0(13589) route( 0(13589) 2 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) }; 0(13589) external_module_call( 0(13589) f_ptr<0xe86d60> 0(13589) , "aliases" 0(13589) ); 0(13589) if ( 0(13589) uri 0(13589) != 0(13589) _myself_ 0(13589) ) { 0(13589) route( 0(13589) 1 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) if ( 0(13589) NOT( 0(13589) external_module_call( 0(13589) f_ptr<0xe86d60> 0(13589) , "location" 0(13589) ); 0(13589) ) 0(13589) ) { 0(13589) external_module_call( 0(13589) f_ptr<0x692790> 0(13589) , "404" 0(13589) , "User Not Found" 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) route( 0(13589) 1 0(13589) ); 0(13589) if ( 0(13589) external_module_call( 0(13589) f_ptr<0x830c30> 0(13589) ); 0(13589) ) { 0(13589) route( 0(13589) 1 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) if ( 0(13589) uri 0(13589) != 0(13589) _myself_ 0(13589) ) { 0(13589) route( 0(13589) 1 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) if ( 0(13589) method 0(13589) == 0(13589) "ACK" 0(13589) ) { 0(13589) route( 0(13589) 1 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) if ( 0(13589) method 0(13589) == 0(13589) "REGISTER" 0(13589) ) { 0(13589) route( 0(13589) 2 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) }; 0(13589) external_module_call( 0(13589) f_ptr<0xe86d60> 0(13589) , "aliases" 0(13589) ); 0(13589) if ( 0(13589) uri 0(13589) != 0(13589) _myself_ 0(13589) ) { 0(13589) route( 0(13589) 1 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) if ( 0(13589) NOT( 0(13589) external_module_call( 0(13589) f_ptr<0xe86d60> 0(13589) , "location" 0(13589) ); 0(13589) ) 0(13589) ) { 0(13589) external_module_call( 0(13589) f_ptr<0x692790> 0(13589) , "404" 0(13589) , "User Not Found" 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) route( 0(13589) 1 0(13589) ); 0(13589) if ( 0(13589) uri 0(13589) != 0(13589) _myself_ 0(13589) ) { 0(13589) route( 0(13589) 1 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) if ( 0(13589) method 0(13589) == 0(13589) "ACK" 0(13589) ) { 0(13589) route( 0(13589) 1 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) if ( 0(13589) method 0(13589) == 0(13589) "REGISTER" 0(13589) ) { 0(13589) route( 0(13589) 2 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) }; 0(13589) external_module_call( 0(13589) f_ptr<0xe86d60> 0(13589) , "aliases" 0(13589) ); 0(13589) if ( 0(13589) uri 0(13589) != 0(13589) _myself_ 0(13589) ) { 0(13589) route( 0(13589) 1 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) if ( 0(13589) NOT( 0(13589) external_module_call( 0(13589) f_ptr<0xe86d60> 0(13589) , "location" 0(13589) ); 0(13589) ) 0(13589) ) { 0(13589) external_module_call( 0(13589) f_ptr<0x692790> 0(13589) , "404" 0(13589) , "User Not Found" 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) route( 0(13589) 1 0(13589) ); 0(13589) if ( 0(13589) method 0(13589) == 0(13589) "ACK" 0(13589) ) { 0(13589) route( 0(13589) 1 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) if ( 0(13589) method 0(13589) == 0(13589) "REGISTER" 0(13589) ) { 0(13589) route( 0(13589) 2 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) }; 0(13589) external_module_call( 0(13589) f_ptr<0xe86d60> 0(13589) , "aliases" 0(13589) ); 0(13589) if ( 0(13589) uri 0(13589) != 0(13589) _myself_ 0(13589) ) { 0(13589) route( 0(13589) 1 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) if ( 0(13589) NOT( 0(13589) external_module_call( 0(13589) f_ptr<0xe86d60> 0(13589) , "location" 0(13589) ); 0(13589) ) 0(13589) ) { 0(13589) external_module_call( 0(13589) f_ptr<0x692790> 0(13589) , "404" 0(13589) , "User Not Found" 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) route( 0(13589) 1 0(13589) ); 0(13589) external_module_call( 0(13589) f_ptr<0xe86d60> 0(13589) , "aliases" 0(13589) ); 0(13589) if ( 0(13589) uri 0(13589) != 0(13589) _myself_ 0(13589) ) { 0(13589) route( 0(13589) 1 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) if ( 0(13589) NOT( 0(13589) external_module_call( 0(13589) f_ptr<0xe86d60> 0(13589) , "location" 0(13589) ); 0(13589) ) 0(13589) ) { 0(13589) external_module_call( 0(13589) f_ptr<0x692790> 0(13589) , "404" 0(13589) , "User Not Found" 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) route( 0(13589) 1 0(13589) ); 0(13589) if ( 0(13589) uri 0(13589) != 0(13589) _myself_ 0(13589) ) { 0(13589) route( 0(13589) 1 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) if ( 0(13589) NOT( 0(13589) external_module_call( 0(13589) f_ptr<0xe86d60> 0(13589) , "location" 0(13589) ); 0(13589) ) 0(13589) ) { 0(13589) external_module_call( 0(13589) f_ptr<0x692790> 0(13589) , "404" 0(13589) , "User Not Found" 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) route( 0(13589) 1 0(13589) ); 0(13589) if ( 0(13589) NOT( 0(13589) external_module_call( 0(13589) f_ptr<0xe86d60> 0(13589) , "location" 0(13589) ); 0(13589) ) 0(13589) ) { 0(13589) external_module_call( 0(13589) f_ptr<0x692790> 0(13589) , "404" 0(13589) , "User Not Found" 0(13589) ); 0(13589) drop( 0(13589) type<0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) route( 0(13589) 1 0(13589) ); 0(13589) route( 0(13589) 1 0(13589) ); 0(13589) 0(13589) routing table 1: 0(13589) if ( 0(13589) NOT( 0(13589) external_module_call( 0(13589) f_ptr<0x175ac0> 0(13589) ); 0(13589) ) 0(13589) ) { 0(13589) external_module_call( 0(13589) f_ptr<0x6927d0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) 0(13589) routing table 2: 0(13589) if ( 0(13589) NOT( 0(13589) external_module_call( 0(13589) f_ptr<0xe89810> 0(13589) , "location" 0(13589) ); 0(13589) ) 0(13589) ) { 0(13589) external_module_call( 0(13589) f_ptr<0x6927d0> 0(13589) ); 0(13589) } else { 0(13589) }; 0(13589) Listening on udp: 172.16.227.160 [172.16.227.160]:5060 tcp: 172.16.227.160 [172.16.227.160]:5060 Aliases: tcp: rajeevb.nj.us.utstar.com:5060 udp: rajeevb.nj.us.utstar.com:5060 WARNING: no fork mode 0(13589) fm_malloc_init: F_OPTIMIZE=16384, /ROUNDTO=2048 0(13589) fm_malloc_init: F_HASH_SIZE=2067, fm_block size=16548 0(13589) fm_malloc_init(0xb5f28000, 33554432), start=0xb5f28000 0(13589) shm_mem_init: success 0(0) core dump limits set to 4294967295 0(0) DEBUG: init_mod: sl_module stateless - initializing 0(0) DEBUG: register_fifo_cmd: new command (sl_stats) registered 0(0) unixsock_register_cmd: New command (sl_stats) registered 0(0) DEBUG: MD5 calculated: fa1d503d7fdb2604d1c6d01e337d6416 0(0) DEBUG: init_mod: tm 0(0) TM - (size of cell=2776, sip_msg=760) initializing... 0(0) Call-ID initialization: '5ff7c5ff' 0(0) DEBUG: register_fifo_cmd: new command (t_uac_dlg) registered 0(0) DEBUG: register_fifo_cmd: new command (t_uac_cancel) registered 0(0) DEBUG: register_fifo_cmd: new command (t_hash) registered 0(0) DEBUG: register_fifo_cmd: new command (t_reply) registered 0(0) unixsock_register_cmd: New command (t_uac_dlg) registered 0(0) unixsock_register_cmd: New command (t_uac_cancel) registered 0(0) unixsock_register_cmd: New command (t_hash) registered 0(0) unixsock_register_cmd: New command (t_reply) registered 0(0) DEBUG: lock_initialize: lock initialization started 0(0) DEBUG: register_fifo_cmd: new command (t_stats) registered 0(0) unixsock_register_cmd: New command (t_stats) registered 0(0) DEBUG: MD5 calculated: af355ad735cdf39a33ba44b1135ce624 0(0) DEBUG: MD5 calculated: 4ac82e9f97f459165de8eea783d7a053 0(0) DEBUG: init_mod: rr 0(0) rr - initializing 0(0) DEBUG: init_mod: maxfwd 0(0) Maxfwd module- initializing 0(0) DEBUG: init_mod: usrloc 0(0) usrloc - initializing 0(0) DEBUG: register_fifo_cmd: new command (ul_stats) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_rm) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_rm_contact) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_dump) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_flush) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_add) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_show_contact) registered 0(0) unixsock_register_cmd: New command (ul_stats) registered 0(0) unixsock_register_cmd: New command (ul_rm) registered 0(0) unixsock_register_cmd: New command (ul_rm_contact) registered 0(0) unixsock_register_cmd: New command (ul_dump) registered 0(0) unixsock_register_cmd: New command (ul_flush) registered 0(0) unixsock_register_cmd: New command (ul_add) registered 0(0) unixsock_register_cmd: New command (ul_show_contact) registered 0(0) DEBUG: init_mod: registrar 0(0) registrar - initializing 0(0) find_export: found <sl_send_reply> in module sl_module [/usr/lib/ser/modules/sl.so] 0(0) find_export: found <ul_bind_usrloc> in module usrloc [/usr/lib/ser/modules/usrloc.so] 0(0) find_export: found <ul_register_udomain> in module usrloc [/usr/lib/ser/modules/usrloc.so] 0(0) find_export: found <ul_get_all_ucontacts> in module usrloc [/usr/lib/ser/modules/usrloc.so] 0(0) find_export: found <ul_insert_urecord> in module usrloc [/usr/lib/ser/modules/usrloc.so] 0(0) find_export: found <ul_delete_urecord> in module usrloc [/usr/lib/ser/modules/usrloc.so] 0(0) find_export: found <ul_get_urecord> in module usrloc [/usr/lib/ser/modules/usrloc.so] 0(0) find_export: found <ul_lock_udomain> in module usrloc [/usr/lib/ser/modules/usrloc.so] 0(0) find_export: found <ul_unlock_udomain> in module usrloc [/usr/lib/ser/modules/usrloc.so] 0(0) find_export: found <ul_release_urecord> in module usrloc [/usr/lib/ser/modules/usrloc.so] 0(0) find_export: found <ul_insert_ucontact> in module usrloc [/usr/lib/ser/modules/usrloc.so] 0(0) find_export: found <ul_delete_ucontact> in module usrloc [/usr/lib/ser/modules/usrloc.so] 0(0) find_export: found <ul_get_ucontact> in module usrloc [/usr/lib/ser/modules/usrloc.so] 0(0) find_export: found <ul_update_ucontact> in module usrloc [/usr/lib/ser/modules/usrloc.so] 0(0) find_export: found <ul_register_watcher> in module usrloc [/usr/lib/ser/modules/usrloc.so] 0(0) find_export: found <ul_unregister_watcher> in module usrloc [/usr/lib/ser/modules/usrloc.so] 0(0) find_export: found <ul_register_ulcb> in module usrloc [/usr/lib/ser/modules/usrloc.so] 0(0) fixing /usr/lib/ser/modules/maxfwd.so mf_process_maxfwd_header 0(0) fixing /usr/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/lib/ser/modules/rr.so loose_route 0(0) fixing /usr/lib/ser/modules/registrar.so lookup 0(0) fixing /usr/lib/ser/modules/registrar.so lookup 0(0) fixing /usr/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/lib/ser/modules/tm.so t_relay 0(0) fixing /usr/lib/ser/modules/sl.so sl_reply_error 0(0) fixing /usr/lib/ser/modules/registrar.so save 0(0) fixing /usr/lib/ser/modules/sl.so sl_reply_error 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) DEBUG: udp_init: trying SO_RCVBUF: 221184 0(0) DEBUG: setting SO_RCVBUF; set=221184,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 223232 0(0) DEBUG: setting SO_RCVBUF; set=223232,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 225280 0(0) DEBUG: setting SO_RCVBUF; set=225280,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 227328 0(0) DEBUG: setting SO_RCVBUF; set=227328,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 229376 0(0) DEBUG: setting SO_RCVBUF; set=229376,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 231424 0(0) DEBUG: setting SO_RCVBUF; set=231424,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 233472 0(0) DEBUG: setting SO_RCVBUF; set=233472,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 235520 0(0) DEBUG: setting SO_RCVBUF; set=235520,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 237568 0(0) DEBUG: setting SO_RCVBUF; set=237568,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 239616 0(0) DEBUG: setting SO_RCVBUF; set=239616,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 241664 0(0) DEBUG: setting SO_RCVBUF; set=241664,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 243712 0(0) DEBUG: setting SO_RCVBUF; set=243712,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 245760 0(0) DEBUG: setting SO_RCVBUF; set=245760,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 247808 0(0) DEBUG: setting SO_RCVBUF; set=247808,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 249856 0(0) DEBUG: setting SO_RCVBUF; set=249856,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 251904 0(0) DEBUG: setting SO_RCVBUF; set=251904,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 253952 0(0) DEBUG: setting SO_RCVBUF; set=253952,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 256000 0(0) DEBUG: setting SO_RCVBUF; set=256000,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 258048 0(0) DEBUG: setting SO_RCVBUF; set=258048,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 260096 0(0) DEBUG: setting SO_RCVBUF; set=260096,verify=262142 0(0) DEBUG: udp_init: trying SO_RCVBUF: 262144 0(0) DEBUG: setting SO_RCVBUF; set=262144,verify=262142 0(0) DEBUG: setting SO_RCVBUF has no effect 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 0(0) DBG: open_uac_fifo: opening fifo... 0(0) DEBUG: FIFO created @ /tmp/ser_fifo 0(0) DEBUG: fifo /tmp/ser_fifo opened, mode=432 0(0) init_unixsock_socket: No unix domain socket will be opened 1(0) DEBUG: init_mod_child (-1): tm 1(0) DEBUG: callid: '5ff7c5ff-0(a)172.16.227.160' 1(0) DEBUG: init_mod_child (-1): usrloc 2(0) INFO: fifo process starting: 13591 2(0) DEBUG: init_mod_child (-2): tm 2(0) DEBUG: callid: '5ff7c5ff-0(a)172.16.227.160' 2(0) DEBUG: init_mod_child (-2): usrloc 2(0) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 2(0) DEBUG: register_fifo_cmd: new command (print) registered 2(0) DEBUG: register_fifo_cmd: new command (uptime) registered 2(0) DEBUG: register_fifo_cmd: new command (version) registered 2(0) DEBUG: register_fifo_cmd: new command (pwd) registered 2(0) DEBUG: register_fifo_cmd: new command (arg) registered 2(0) DEBUG: register_fifo_cmd: new command (which) registered 2(0) DEBUG: register_fifo_cmd: new command (ps) registered 2(0) DEBUG: register_fifo_cmd: new command (kill) registered 2(0) WARNING: no fifo_db_url given - fifo DB commands disabled! 0(13589) DEBUG: init_mod_child (1): tm 0(13589) DEBUG: callid: '5ff7c5ff-13589(a)172.16.227.160' 0(13589) DEBUG: init_mod_child (1): usrloc now if i press ctrl+c then below prints come but ser does not start 1(13590) INFO: signal 2 received 1(13590) Memory status (pkg): 1(13590) fm_status (0x8107e40): 1(13590) heap size= 1048576 1(13590) dumping free list: 1(13590) hash = 16 fragments no.: 4, unused: 0 bucket size: 128 - 128 (first 128) 1(13590) hash = 281 fragments no.: 1, unused: 0 bucket size: 2248 - 2248 (first 2248) 1(13590) hash = 800 fragments no.: 1, unused: 0 bucket size: 6400 - 6400 (first 6400) 1(13590) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 1015408) 1(13590) TOTAL: 7 free fragments = 1024568 free bytes 1(13590) ----------------------------- 0(13589) 2(13591) INT received, program terminates 2(13591) INFO: signal 15 received 2(13591) Memory status (pkg): 2(13591) fm_status (0x8107e40): 2(13591) heap size= 1048576 2(13591) dumping free list: 2(13591) hash = 7 fragments no.: 1, unused: 0 bucket size: 56 - 56 (first 56) 2(13591) hash = 16 fragments no.: 2, unused: 0 bucket size: 128 - 128 (first 128) 2(13591) hash = 281 fragments no.: 1, unused: 0 bucket size: 2248 - 2248 (first 2248) 2(13591) hash = 800 fragments no.: 1, unused: 0 bucket size: 6400 - 6400 (first 6400) 2(13591) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 1015408) 2(13591) TOTAL: 6 free fragments = 1024368 free bytes 2(13591) ----------------------------- 0(13589) DEBUG: tm_shutdown : start 0(13589) DEBUG: unlink_timer_lists : emptying DELETE list 0(13589) DEBUG: tm_shutdown : emptying hash table 0(13589) DEBUG: tm_shutdown : releasing timers 0(13589) DEBUG: tm_shutdown : removing semaphores 0(13589) DEBUG: tm_shutdown : destroying tmcb lists 0(13589) DEBUG: tm_shutdown : done 0(13589) Memory status (pkg): 0(13589) fm_status (0x8107e40): 0(13589) heap size= 1048576 0(13589) dumping free list: 0(13589) hash = 2 fragments no.: 35, unused: 0 bucket size: 16 - 16 (first 16) 0(13589) hash = 3 fragments no.: 2, unused: 0 bucket size: 24 - 24 (first 24) 0(13589) hash = 11 fragments no.: 1, unused: 0 bucket size: 88 - 88 (first 88) 0(13589) hash = 16 fragments no.: 3, unused: 0 bucket size: 128 - 128 (first 128) 0(13589) hash = 281 fragments no.: 1, unused: 0 bucket size: 2248 - 2248 (first 2248) 0(13589) hash = 800 fragments no.: 1, unused: 0 bucket size: 6400 - 6400 (first 6400) 0(13589) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 1015408) 0(13589) TOTAL: 44 free fragments = 1025136 free bytes 0(13589) ----------------------------- 0(13589) Memory status (shm): 0(13589) fm_status (0xb5f28000): 0(13589) heap size= 33554432 0(13589) dumping free list: 0(13589) hash = 1 fragments no.: 8, unused: 0 bucket size: 8 - 8 (first 8) 0(13589) hash = 2 fragments no.: 3, unused: 0 bucket size: 16 - 16 (first 16) 0(13589) hash = 4 fragments no.: 3, unused: 0 bucket size: 32 - 32 (first 32) 0(13589) hash = 5 fragments no.: 3, unused: 0 bucket size: 40 - 40 (first 40) 0(13589) hash = 40 fragments no.: 1, unused: 0 bucket size: 320 - 320 (first 320) 0(13589) hash = 90 fragments no.: 1, unused: 0 bucket size: 720 - 720 (first 720) 0(13589) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) 0(13589) hash = 512 fragments no.: 2, unused: 0 bucket size: 4096 - 4096 (first 4096) 0(13589) hash = 1024 fragments no.: 2, unused: 0 bucket size: 8192 - 8192 (first 8192) 0(13589) hash = 2055 fragments no.: 1, unused: 0 bucket size: 1048576 - 2097152 (first 1572864) 0(13589) hash = 2059 fragments no.: 1, unused: 0 bucket size: 16777216 - 33554432 (first 31937728) 0(13589) TOTAL: 26 free fragments = 33537656 free bytes 0(13589) ----------------------------- 0(13589) shm_mem_destroy 0(13589) destroying the shared memory lock 0(13589) Thank you for flying ser [root@linuxrnd1 ~]#

2 1

Add Header on cancel
by Alex Rendour 21 May '10

21 May '10

Hi, Is it possible to add an header reason in a cancel? We receive a CANCEL without header reason and we need to forward to another SIP proxy with the reason header? We use this code in my conf: if(is_method("CANCEL")) { if(!is_present_hf("Reason:")) { xlog("L_ERR", "reason missing"); append_hf("reason: Q.850; cause=31\r\n"); xlog("L_ERR", "add raison"); } } Thanks, regards, -- Alexandre Rendour Acropolis Telecom <http://www.acropolistelecom.net> Direct: +33 (0) 181813201 Support: +33 (0) 811 851 851 rendour(a)acropolistelecom.net <mailto:rendour@acropolistelecom.net> Adresse : 161-163 avenue Gallieni Paris - Porte de Bagnolet 93170 Bagnolet

4 5

break seems to work also inside while loop
by Juha Heinanen 21 May '10

21 May '10

according to core cookbook http://sip-router.org/wiki/cookbooks/core-cookbook/devel#break 'break' only works inside switch statement, but i just tried and it seemed to work also inside while statement. is that an accident or on purpose? -- juha

2 2

Need help in routing configuration
by Nelson Pereira 20 May '10

20 May '10

I have the bellow setup for openser, yet this wont work in Kamilio. Wondering if anyone could help to make this routing code work? What Im looking to do is receive Invites in UDP and relaying it via TCP.... route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); }; route(1); } route[1] { if (src_ip==10.98.6.5) { if (dst_port==5060) { t_relay("tcp:10.98.118.20:5060"); } if (dst_port==5065) { t_relay("tcp:10.98.118.20:5065"); } else if (dst_port==5066) { t_relay("tcp:10.98.118.20:5066"); } else if (dst_port==5067) { t_relay("tcp:10.98.118.20:5067"); } # else { # t_relay("tcp:10.98.118.20:5060"); # } } else { t_relay("udp:10.98.6.5:5060"); }; } Nelson Pereira Senior Network Specialist Protus<http://www.protus.com/> npereira(a)protus.com<mailto:name@protus.com> phone: 613.733.0000 ext.528 MyFax: 613.822.5083 MyFax.com<http://www.myfax.com/> | my1voice.com<http://www.my1voice.com/> | Campaigner.com<http://www.campaigner.com/>

4 12

allow_trusted() use.
by Vicente 20 May '10

20 May '10

Good afternoon, I'm trying to configure permissions module to allow calls from an IP address. I've the module loaded, the entry on the data base table trusted. But according to the examples I've seen on the list and other sources, the call to allow_trusted() has to be placed just before the INVITE authorization section like: if (is_from_local()){ if(!allow_trusted()){ ...authorization... In my scenario allow_tusted() is never reached and thus Kamailo allows the call regardless the table trusted is populated with the the allowed proxy IP or not. The INVITEs that I want to allow come from another proxy and then I understand is_from_local() is always evaluated false for those messages. And allow_trusted isn't reached. Where on the script should the condition to allow only the calls from the external proxy IP be placed? Thanks in advance. Vicente.

2 3

Routing problem
by alet＠librelogiciel.com 19 May '10

19 May '10

Hi, First I'm a complete SIP newbie. We've purchased a new pabx which is a complete piece of crap (Alcatel OmniPCX OXO) : no LDAP support, no registrar functionnality, and so on... and it seems there's no way to upgrade. This pabx is on a subnet different from the one used by desktop computers. What I'd like to achieve is to be able to install SIP soft phones onto desktop computers (like ekiga for example), allow them to call all non-IP phones already configured in the pabx, and be called by any non-IP phone through the pabx. Since the pabx doesn't have a registrar mode, it can't associate IP addresses with phone numbers. So in ekiga, I can configure the pabx as a SIP proxy, and my ekiga can call any normal (non-IP) phone connected to the pabx. This works fine, but is only one half of what I want, since I want these SIP phones to be callable from non-IP phones. Enters openser, which I plan to use as a registrar and router. So I've installed openser 1.3.2 onto a Debian Squeeze box. I can successfully register SIP soft phones to it, and they can call each other just fine calling (number(a)sip.example.com). This openser box is on the same subnet as the Alcatel pabx (so different from the one used by SIP soft phones). All our "normal" phone numbers are 6 digits long and begin with 290, so I want to route all call numbers beginning with "290" to the Alcatel pabx. So I've modified the default openser.cfg to contain : --- CUT --- port=5060 listen=udp:10.125.0.1:5060 ... route { ... if (is_method("INVITE")) { setflag(1); if (uri=~"sip:290[0-9]{3}@sip.example.com") { xlog("#### CALL TO NON-IP PHONE ####\n"); route(2); } } ... } ... # route[1] was in original openser.cfg route[1] { # for INVITEs enable some additional helper routes if (is_method("INVITE")) { t_on_branch("2"); t_on_reply("2"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); }; exit; } # # I've added the following route[2] { rewritehostport("pabx-alcatel.example.com:5060"); xlog("#### Server and port rewritten ####\n"); route(1); }... --- CUT --- Now in ekiga if I dial sip:number@pabx-alcatel.example.com:5060 it works fine as it always did, because this call is direct. However if I dial sip:number@sip.example.com:5060 the host and port in URL are correctly rewritten in sip:number@pabx-alcatel.example.com:5060 but nothing happens : the call doesn't ring and fails immediately. In the logs when dialing sip:290081@sip.example.com:5060 I've got this, so I suppose it *SHOULD* work : --- CUT --- /usr/sbin/openser[759]: #### CALL TO NON-IP PHONE #### /usr/sbin/openser[759]: #### Server and port rewritten #### /usr/sbin/openser[759]: new branch at sip:290081@pabx-alcatel.example.com:5060 /usr/sbin/openser[760]: incoming reply /usr/sbin/openser[758]: incoming reply --- CUT --- but it doesn't work as expected. any idea why ? NB : I understand this would only solve half of my problem, the other part will be solved (I hope) with the help of the guy who installed the pabx. Thanks in advance Jerome Alet

3 5

Siremis - missing cdr calcs and repeats
by Anders 19 May '10

19 May '10

Hi, I'm having some challenges with Siremis, and it says on the project website that this list is the place to ask - so here are the two issues: 1. Repeats: I had problems with repeats in the cdrs table. This was possible, because I had forgotten to set the correct unique index (call_id + from_tag + to_tag). Once this was set, it was obviously not possible to make the repeats anymore - so problem solved. But I still wonder: why was it trying to calculate calls that had already been calculated? Shouldn't it ignore these and only treat new ones? 2. Missing calcs After an unsuccessful attempt to upgrade MySql from 5.0 to 5.1, it seems that not all calls are being calculated and put in the cdrs table. The CRON job was stopped due to the failed upgrade of MySql, but I started it again later - MySql was not down. However, I thought it would 'catch up' with the transaction that had not been calculated while the CRON job for the stored procedure had been off, but it did not. In fact, it seems that not all calls are being calculated and put inthe cdrs table - I'm missing almost half the calls. Any idea what this can be? Thanks!! //Anders

1 1

Re: [SR-Users] Kamailio 3.0.1 output debug to terminal
by Nelson Pereira 19 May '10

19 May '10

Thanks for the response, although I'm not seing anything coming out when tail'ing the syslog file. I only see when kamailio starts: May 19 10:22:14 qa-qsuitePBX-01 kamailio: INFO: <core> [tcp_main.c:4150]: init_tcp: using epoll_lt as the io watch method (auto detected) May 19 10:22:14 qa-qsuitePBX-01 /usr/sbin/kamailio[1784]: INFO: usrloc [hslot.c:53]: locks array size 512 May 19 10:22:14 qa-qsuitePBX-01 /usr/sbin/kamailio[1784]: INFO: <core> [udp_server.c:166]: INFO: udp_init: SO_RCVBUF is initially 124928 May 19 10:22:14 qa-qsuitePBX-01 /usr/sbin/kamailio[1784]: INFO: <core> [udp_server.c:217]: INFO: udp_init: SO_RCVBUF is finally 262142 May 19 10:22:14 qa-qsuitePBX-01 /usr/sbin/kamailio[1784]: INFO: <core> [udp_server.c:166]: INFO: udp_init: SO_RCVBUF is initially 124928 May 19 10:22:14 qa-qsuitePBX-01 /usr/sbin/kamailio[1784]: INFO: <core> [udp_server.c:217]: INFO: udp_init: SO_RCVBUF is finally 262142 May 19 10:22:14 qa-qsuitePBX-01 /usr/sbin/kamailio[1784]: INFO: <core> [udp_server.c:166]: INFO: udp_init: SO_RCVBUF is initially 124928 May 19 10:22:14 qa-qsuitePBX-01 /usr/sbin/kamailio[1784]: INFO: <core> [udp_server.c:217]: INFO: udp_init: SO_RCVBUF is finally 262142 May 19 10:22:14 qa-qsuitePBX-01 /usr/sbin/kamailio[1784]: INFO: <core> [udp_server.c:166]: INFO: udp_init: SO_RCVBUF is initially 124928 May 19 10:22:14 qa-qsuitePBX-01 /usr/sbin/kamailio[1784]: INFO: <core> [udp_server.c:217]: INFO: udp_init: SO_RCVBUF is finally 262142 May 19 10:22:14 qa-qsuitePBX-01 /usr/sbin/kamailio[1804]: INFO: ctl [io_listener.c:224]: io_listen_loop: using epoll_lt io watch method (config) Do I need anything else to enable? Nelson Pereira -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, May 19, 2010 10:08 AM To: Nelson Pereira Subject: Re: [SR-Users] Kamailio 3.0.1 output debug to terminal Kamailio default uses syslog logging. I suggest to start a second console and use: tail -f /var/log/syslog you can redirect syslog to a dedicated file using, see: http://www.kamailio.org/dokuwiki/doku.php/tutorials:debug-syslog-messages#c… regards klaus Am 19.05.2010 15:50, schrieb Nelson Pereira: > How do you output debug info to the terminal? > > I don't understand this : > > # Several features can be enabled using '#!define WITH_FEATURE' directives: > > # > > # *** To run in debug mode: > > # - define WITH_DEBUG > > How do I get debug and output to screen? > > *Nelson Pereira* > > > > _______________________________________________ > 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

1 0

Kamailio 3.0.1 output debug to terminal
by Nelson Pereira 19 May '10

19 May '10

How do you output debug info to the terminal? I don't understand this : # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG How do I get debug and output to screen? Nelson Pereira

1 0

Re: [SR-Users] Acnnot get Kamailio to start ... GRRR
by Nelson Pereira 19 May '10

19 May '10

This is what Im getting at the end of the make deb, and it's not making the debian package...: Makefile.defs defs skipped make[3]: Entering directory `/usr/src/kamailio-3.0.1/modules/tls' mkdir -p /usr/src/kamailio-3.0.1/debian/kamailio-tls-modules/etc/kamailio MAIN_NAME=kamailio ./sip-router_cert.sh -d /usr/src/kamailio-3.0.1/debian/kamailio-tls-modules/etc/kamailio/ Could not find openssl command Set OPENSSL environment variable properly (see -h for more info) make[3]: *** [install-tls-cert] Error 1 make[3]: Leaving directory `/usr/src/kamailio-3.0.1/modules/tls' make[2]: *** [install-modules] Error 1 make[2]: Leaving directory `/usr/src/kamailio-3.0.1' make[1]: *** [install] Error 2 make[1]: Leaving directory `/usr/src/kamailio-3.0.1' dpkg-buildpackage: failure: fakeroot debian/rules binary gave error exit status 2 make: [deb] Error 2 (ignored) Nelson Pereira Senior Network Specialist Protus npereira(a)protus.com phone: 613.733.0000 ext.528 MyFax: 613.822.5083 MyFax.com | my1voice.com | Campaigner.com -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, May 19, 2010 4:20 AM To: Nelson Pereira Subject: Re: [SR-Users] Acnnot get Kamailio to start ... GRRR Am 18.05.2010 21:10, schrieb Nelson Pereira: > I tried make deb and was getting other errors... Could it be because im running 64bit? That shouldn't be the problem. Usually you just execute "make deb". The first time you will get an error message about missing dependencies. Install the missing dependencies (apt-get install ...) and then execute "make deb" again. regards klaus > > Nelson Pereira > Senior Network Specialist > > Protus > npereira(a)protus.com > phone: 613.733.0000 ext.528 > MyFax: 613.822.5083 > > MyFax.com | my1voice.com | Campaigner.com > > > -----Original Message----- > From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Sent: Tuesday, May 18, 2010 2:56 PM > To: Nelson Pereira > Cc: sr-users(a)lists.sip-router.org > Subject: Re: [SR-Users] Acnnot get Kamailio to start ... GRRR > > Nelson Pereira wrote: >> Do I absolutely need Mysql DB for this to work? > > No. > > If you use "make deb", then the dependencies needs to be fulfilled for > generating the modules (e.g. mysql-client). But there is no need to > install the kamailio-mysql module. > > regards > klaus > >> I'm simply using this to do UDP to TCp conversion, nothing else. >> >> NP >> >> -----Original Message----- >> From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Edson - Lists >> Sent: Tuesday, May 18, 2010 1:19 PM >> To: sr-users(a)lists.sip-router.org >> Subject: Re: [SR-Users] Acnnot get Kamailio to start ... GRRR >> >> It can be a stupid question, but did the directory exists? Has the user >> who runs Kamailio rights to it? >> >> BDW, which version of Kamailio are you testing? Why not the last SR? >> >> Edson. >> >> Em 18/05/2010 13:55, Nelson Pereira escreveu: >>> Basic install with source using the provided script: >>> >>> make_and_install install "standard" >>> >>> then try to do : >>> >>> /usr/local/sbin/kamctl start >>> >>> INFO: Starting Kamailio : >>> >>> ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start >>> failed >>> >>> root@qa-qsuitePBX-01:/usr/local/sbin# >>> >>> What am I doing wrong? >>> >>> *Nelson Pereira* >>> >>> >>> >>> _______________________________________________ >>> 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 >> >> _______________________________________________ >> 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 >> >> _______________________________________________ >> 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 >

1 2

Acnnot get Kamailio to start ... GRRR
by Nelson Pereira 19 May '10

19 May '10

Basic install with source using the provided script: make_and_install install "standard" then try to do : /usr/local/sbin/kamctl start INFO: Starting Kamailio : ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed root@qa-qsuitePBX-01:/usr/local/sbin# What am I doing wrong? Nelson Pereira

6 10

Kamailio 1.5 stops processing messages
by Santiago Gimeno 19 May '10

19 May '10

Hi, I'm working with Kamailio 1.5 (rev 5968) and from time to time kamailio freezes and stops processing messages. Kamailio is working only in UDP. I really don't know in which conditions this problem is happening: it happens both in high and low load. Here is a backtrace of one of the UDP working processes when kamailio is frozen (all the other UDP process backtraces look similar) : #0 0xb7fdf410 in ?? () #1 0xbfbfec28 in ?? () #2 0x00000001 in ?? () #3 0xa7392ad0 in ?? () #4 0xb7f3b4ac in sched_yield () from /lib/tls/i686/cmov/libc.so.6 #5 0xb7ba9463 in lock_hash (i=10361) at ../../mem/../fastlock.h:182 #6 0xb7bc4587 in t_lookup_request (p_msg=0x823fc70, leave_new_locked=1) at t_lookup.c:468 #7 0xb7bc54ae in t_newtran (p_msg=0x823fc70) at t_lookup.c:1124 #8 0xb7bb740c in t_relay_to (p_msg=0x823fc70, proxy=0x0, flags=1) at t_funcs.c:212 #9 0xb7bb8f6c in t_replicate (p_msg=0x823fc70, dst=0x817cdf8, flags=1) at t_fwd.c:763 #10 0xb7bc7fce in w_t_replicate (p_msg=0x823fc70, dst=0x817cdf8 "\230ê\030\b\016", flags=0x0) at tm.c:957 #11 0x0805301c in do_action (a=0x818eb20, msg=0x823fc70) at action.c:874 #12 0x080557aa in run_action_list (a=0x818eb20, msg=0x823fc70) at action.c:145 #13 0x08054491 in do_action (a=0x818ec78, msg=0x823fc70) at action.c:746 #14 0x080557aa in run_action_list (a=0x818c940, msg=0x823fc70) at action.c:145 #15 0x08053efb in do_action (a=0x81b38e0, msg=0x823fc70) at action.c:120 #16 0x080557aa in run_action_list (a=0x81b3878, msg=0x823fc70) at action.c:145 #17 0x08054491 in do_action (a=0x81b5f00, msg=0x823fc70) at action.c:746 #18 0x080557aa in run_action_list (a=0x81b5f00, msg=0x823fc70) at action.c:145 #19 0x08054f2d in do_action (a=0x81b5f68, msg=0x823fc70) at action.c:752 #20 0x080557aa in run_action_list (a=0x81b5f68, msg=0x823fc70) at action.c:145 #21 0x08054f2d in do_action (a=0x81b5fd0, msg=0x823fc70) at action.c:752 #22 0x080557aa in run_action_list (a=0x81aefd0, msg=0x823fc70) at action.c:145 #23 0x08053efb in do_action (a=0x818bc08, msg=0x823fc70) at action.c:120 #24 0x080557aa in run_action_list (a=0x8187910, msg=0x823fc70) at action.c:145 #25 0x08055b43 in run_top_route (a=0x8187910, msg=0x823fc70) at action.c:120 #26 0x0808c659 in receive_msg ( buf=0x8158040 "REGISTER sip:xxxxxxxxxxx.com SIP/2.0\r\nFrom: < sip:xxxxxx@xxxxxxxxxxxx.com <sip%3Axxxxxx(a)xxxxxxxxxxxx.com>>;tag=as516fcce2\r\nTo: <sip:xxxxxxx@xxxxxxxxxxxx.com <sip%3Axxxxxxx(a)xxxxxxxxxxxx.com>>\r\nCall-ID: 01e4ea8e185c85332ea68874543c810e(a)xxxxxxxxxxxxxx.com\r\nCSeq: 383 REGISTER\r\nVia: SIP"..., len=794, rcv_info=0xbfc004d4) at receive.c:175 #27 0x080c3ea3 in udp_rcv_loop () at udp_server.c:449 #28 0x0806e394 in main (argc=9, argv=0xbfc006b4) at main.c:774 Any idea of what might be happening? Best regards, Santi

3 10

Robust header compression (RFC 3095) with Kamailio + rtpproxy
by Vikram Ragukumar 18 May '10

18 May '10

Hello, We are looking into bandwidth conservation by implementing RTP/UDP/IP header compression. Has anybody implemented ROHC or another header compression scheme in combination with kamailio + rtpproxy ? Could you please point us to online documentation or other useful resources ? Thanks and Regards, Vikram.

4 3

Trying to find the built-in password for the database....
by SIP 18 May '10

18 May '10

So... I've been working on installing Kamailio 3.0.1 on a test server to begin migration from SER 0.9.6 (yes, I know it will be a bit of a migration). The INSTALL docs are... not perfect if working with the source (and not a package). Following the instructions did not get me mysql support. There were files that didn't install where they needed to be that I had to move over by hand -- none of which is mentioned in the docs. Even after removing the db_mysql from the list of excluded modules, it BUILT the module, but didn't install it. I also used a different DB name, username, and password from the default (to match more closely the environment we'll be migrating from). After tinkering, I got the DB scripts to run with the right username/pass, and have now edited the kamctlrc and kamailio.cfg to have the proper db name, username, pass, etc. However, when I try and start kamailio, it fails, and the log spits out a bunch of access denied messages using the original, default username/pass (openserro@localhost). My question is, where is that password located? What file have I missed? N.

2 2

Kamailio as a UAC
by Ricardo Martinez 18 May '10

18 May '10

Hello. Is possible use some module in kamailio to authenticate against a Asterisk Box for example?. What I need is to have a Sip Client Registered in Kamailio, but when this client is called I need to send the call to Asterisk which will ask for authentication (user/pass). Is possible to do this with kamailio?. If so, can someone give an example? Thanks in advance, Regards, Ricardo Martinez.-

2 4

Dispatcher send OPTION Interval
by François BERGANZ 18 May '10

18 May '10

Hello all, In dispatcher, I know how to send OPTIONS, modify the interval when a gateway is down... I can't see how to change the Ping option interval when a gateway is up. how can I do? If it is in the sources, I can't find it. thank you, regards, François BERGANZ Ingénieur/Chef de projets VoIP Acropolis Telecom

2 1

Question on install of Kamailio 3.0.1 from source on Ubuntu 8.04
by Nelson Pereira 18 May '10

18 May '10

I downlaoded the source of 3.0.1 and untar'ed. Then executed as per docs (INSTALL file): make prefix=/ make prefix=/ install Yet, when I start kamailio from /sbin/kamailio, I get this: root@qa-qsuitePBX-01:/usr/src# kamailio loading modules under //lib/kamailio/modules_k/://lib/kamailio/modules/ Listening on udp: 10.98.6.6 [10.98.6.6]:5060 udp: 10.98.6.6 [10.98.6.6]:5065 udp: 10.98.6.6 [10.98.6.6]:5066 udp: 10.98.6.6 [10.98.6.6]:5067 Aliases: It's loading modules from //lib....? What is this? And is kamailio installed correctly? Nelson Pereira Senior Network Specialist Protus<http://www.protus.com/> npereira(a)protus.com<mailto:name@protus.com> phone: 613.733.0000 ext.528 MyFax: 613.822.5083 MyFax.com<http://www.myfax.com/> | my1voice.com<http://www.my1voice.com/> | Campaigner.com<http://www.campaigner.com/>

2 2

installing Kamilio onto Ubuntu 8.04
by Nelson Pereira 18 May '10

18 May '10

Is there a apt-get package to install kamilio? If so, how do I do that? NP

2 3

Re: [SR-Users] SIP REFER support...
by Jignesh Gandhi 18 May '10

18 May '10

Hello, Does OpenSER/Kamailio support REFER ? thanks, --Jignesh

4 11

SIP Router Devel Meeting, June 8, 2010, Berlin
by Daniel-Constantin Mierla 17 May '10

17 May '10

Hello, the SIP Router Devel Meeting days was set to June 8, in Berlin - event is open for anyone interested to discuss current development state and future directions. Following days, some of us will be present at LinuxTag, where we have a booth. More details at: http://www.kamailio.org/w/2010/05/sip-router-devel-meeting-june-8-2010-berl… Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

1 0

apt repositories for kamailio 3.0
by Daniel-Constantin Mierla 17 May '10

17 May '10

Hello, I setup an apt repository for kamailio 3.0.x, holding now debs for lenny i386 and ubuntu 9.10 i386. Add to your source.list: - for debian: deb http://kamailio.org/pub/apt/debian lenny main contrib non-free - for ubuntu deb http://kamailio.org/pub/apt/ubuntu karmic main contrib non-free then apt-get update and apt-get install kamailio. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

1 0

ds_ping_method from dispatcher
by François BERGANZ 17 May '10

17 May '10

Hello, I could see in dispatcher module documentation in| ds_ping_method "|With this Method you can define, with which method you want to probe the failed gateways. This method is only available, if compiled with the probing of failed gateways enabled. |" How to enable it before the compilation? I can't find it. Thank you | -- François BERGANZ Ingénieur/Chef de projets VoIP**

2 1

make: *** No rule to make target `deb-lenny'. Stop.
by Toyima Dias 15 May '10

15 May '10

Hello sip-router community... I'm having some problems with my Kamailio installation (kamailio-1.5.0-notls_src.tar.gz)...after downloading and decompress i do the following: kamailio:/usr/src/kamailio-1.5.0-notls# make deb-lenny bison -d -b cfg cfg.y cfg.y: conflicts: 1 shift/reduce flex cfg.lex make: *** No rule to make target `deb-lenny'. Stop. What sould i do to solve this problem? i really want to solve this but it's quite dificult, could somebody help me please? i've checked on the web, but no answers about this....does this has something to be with the Makefile under kamailio directory? Regards

3 5

Invitación a conectarnos en LinkedIn
by Johnny Laura 13 May '10

13 May '10

LinkedIn ------------ Me gustaría añadirte a mi red profesional en LinkedIn. -Johnny Johnny Laura Propietario en ITUS - GISS Perú Confirma que conoces a Johnny Laura https://www.linkedin.com/e/isd/1299499162/fi9HEYnJ/ ------ (c) 2010, LinkedIn Corporation

2 1

carrierroute cr_route fails with pvars
by Ole Kaas 13 May '10

13 May '10

Hello, Any idea why this fails: $var(carrier) = "carriername"; $var(domainname) = "domain"; if(cr_route( "$var(carrier)", "$var(domainname)", "$rU", "$rU", "call_id")) with tis in the log: INFO: carrierroute [cr_func.c:598]: uri domain was rewritten to sip:domain, carrier 2, domain 2 While this: if(cr_route( "carriername", "$var(domainname)", "$rU", "$rU", "call_id")) and this: if(cr_route( "$var(carrier)", "domain", "$rU", "$rU", "call_id")) works as expected? Kamailio 3.0.0 /Ole

2 3

[NATHELPER] bug in SDP parsing when dealing with multi-part bodies ?
by Pascal Maugeri 13 May '10

13 May '10

Hi I got the following error when dealing with SIP INVITE with multi-part when the client is behind NAT: May 12 15:27:04 test /usr/sbin/kamailio[1792]: ERROR: nathelper [nathelper.c:2954]: can't extract body from the message The SIP INVITE message looks like: INVITE sip:toto@test.net <sip%3Atoto(a)test.net> SIP/2.0 [...] Content-Type: multipart/mixed;boundary=gnkr Content-Length: 659 --gnkr Content-Type: application/sdp Content-Length: 330 v=0 o=pascal 1273565967736 1273565967737 IN IP4 10.1.1.226 [...] It looks the Content-length within the multi-part is not accepted. I fixed it changing the source code of parser/sdp/sdp_helpr_funcs.c, in get_sdp_hdr_field() function: switch(hdr->type){ case HDR_CONTENTTYPE_T: case HDR_CONTENTDISPOSITION_T: case HDR_CONTENTLENGTH_T: replaced with: switch(hdr->type){ case HDR_CONTENTTYPE_T: case HDR_CONTENTDISPOSITION_T: (I'm using kamailio 3.0.1) Cheers Pascal

3 3

db_bind_mod() function...
by Bruno Bresciani 13 May '10

13 May '10

Hi people, in kamailio the function bind_dbmod() was replaced by the db_bind_mod() ok? I didn't understand very well the db_func_t *mydbf parameter that I have to provide for this new function. In ser I used bind_dbmod() to check if database module has been loaded and I want make this in kamailio now using the db_bind_mod() function but I don't know what provide to db_func_t *mydbf parameter. Someone can help me? Cheers, Bruno

1 0

lookup(), received an $du
by Andreas Granig 13 May '10

13 May '10

Hi, Some interesting behavior with kam-3.0, which I haven't experienced with <= 1.5: Let's say I have two subscribers A and B registered. A has a received-value in location table, B has it set to NULL. If I call lookup() for A, $ru is set to the registered contact, and $du is set to the received-value. When for example A is busy, I catch it with a failure route and rewrite $ru to B. Calling lookup() again sets my $ru to the contact of B, but since the received-value of B is NULL in the DB, the $du is still pointing to the received-value of A, instead of being set to null. A quick-fix is to set $du to null before calling lookup(), but I'm wondering if this is a bug or intended behavior? Thanks, Andreas

4 8

Re: [SR-Users] function bind_dbmod()
by Alex Balashov 12 May '10

12 May '10

No problem. It is all courtesy of my friend 'grep', which has following options: -H print filename -R search recursively -i case-insensitive matching So, for example, one can do something like this using 'egrep', a variant of grep that supports regular expressions: egrep -HRi ".*bind.*db.*" . It has been very fruitful for me and probably can be for you as well. :) > oh yes, thank's for your help... > > > > 2010/5/12 Alex Balashov <abalashov(a)evaristesys.com> > >> >> > But I'm using the kamailio 1.5.0 version and don't exist the lib >> > directory, only in the 3.0.0 version. >> >> Oh, I see. In that case, it's the same function (db_bind_mod()), but in >> db/db.c instead. >> >> -- >> Alex Balashov - Principal >> Evariste Systems LLC >> 1170 Peachtree Street >> 12th Floor, Suite 1200 >> Atlanta, GA 30309 >> Tel: +1-678-954-0670 >> Fax: +1-404-961-1892 >> Web: http://www.evaristesys.com/ >> >> > -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

1 0

function bind_dbmod()
by Bruno Bresciani 12 May '10

12 May '10

Hi people, Someone know tell me if in the kamailio the bind_dbmod() function doesn't exist anymore? If doesn't, exist other similar function? Cherrs, Bruno

2 2

Asterisk call features
by toqeer ali 12 May '10

12 May '10

Currently i have a setup kamailio/asterisk. Is this possible in opensips to pass standard asterisk feature codes to asterisk from openips so the users of asterisk can use of message center, call forwarding, call transfer, 3 way calling, return call, and other standard calling features available at the dial prompt. (1) Opensips ALL ENPOINTS REGISTER WITH OPENSIPS (2) Asterisk Media Server For now i can register and make calls but have no features at the dial prompt. i want to use the features below of asterisk. - 69 - Last Number Called - 70 - Activate Call Waiting - 71 - Deactivate Call Waiting - 72 - Call Forwarding - 73 - Deactivate Call Forwarding - 78 - Enable Do not disturb - 79 - Disable Do not disturb - 90 - Call Forward Busy - 91 - Disable call forward busy - 97 - Message Center - 98 - Enter Message Center Please help me if there is any possibility. Any help will highly appreciated. -- Toqeer Ali Syed Red Hat Certified Engineer mob: +92 321 9059916

2 1

Referring to a subnet
by Jon Farmer 12 May '10

12 May '10

Hi I need to trust a subnet and was wondering if something like src_ip = x.x.x.x/27 where x.x.x.x is the network address of the subnet is valid in the config. I am using OpenSER 1.1.0 in this instance. -- Jon Farmer Tel: 07795 118140 Email: viperdudeuk(a)gmail.com Twitter: @viperdudeuk

4 3

v3.0 cfg tips
by Daniel-Constantin Mierla 12 May '10

12 May '10

Hello, some time ago, I started a series of short blog posts with tips valid for kamailio/ser v3.0.x config file, not finished by has already 10 articles. You can access it at: http://by-miconda.blogspot.com/2010/04/kamailio-30x-dyk-toc.html Hope is useful for some of you! Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla

3 2

How to insert modules in kamailio?
by Bruno Bresciani 12 May '10

12 May '10

How to insert modules in kamailio? Hi people, I worked with ser but now I'm migrating to kamailio... I'd like to know how to insert a ser module in the kamailio? Example: In the ser modules exist the route module and I want to use it in the kamailio. It's possible? Cheers, Bruno

4 11

Configuring TCP keep alive and connection lifetime
by Pascal Maugeri 11 May '10

11 May '10

"kamailio" <users(a)lists.kamailio.org>, Hi I am sorry to ask again about this topic but I don't understand yet how the TCP keep-alive are sent. With Kamailio 3.0.1 and the following configuration: tcp_keepalive=yes tcp_keepidle=10 tcp_connection_lifetime=3600 The TCP connection is kept alive and TCP Keep-Alive packets are sent when the TCP connection is idle. But I don't understand why the time between KA is not linear: first KA is sent 10 seconds after last last TCP message. This is logical and connected with the tcp_keepidle=10 configuration. then 2nd KA is sent 75 seconds after first KA Then my client sends some traffic and 23 seconds after last message, a new KA is sent. Another time it was 54 seconds. etc. etc. Could you explain me where do come from these values of 75 seconds, 23, 54 ? Everything works well: my client behind NAT is kept connected but I'd like to understand why it does work :-) Thanks in advance, Pascal

4 6

[LCR] Why 'gw_name' column is an UNIQUE index in 'gw' table?
by Iñaki Baz Castillo 08 May '10

08 May '10

Why is it an UNIQUE key? It makes things complex when I want to create another gateway with same IP and port (but belonging to a different grp_id). I still want to name it with the same name (an internal domain for example). Regards. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 5

Wrong handling CANCEL message
by pars3c 08 May '10

08 May '10

Hi, i have a problem about the handling of the "cancel" message. The call flow is this: A -----------------> (Invite) Proxy (P) B (100 Tryng) <--------------------------- ----------------->(Invite) B (100 Tryng) <--------------------------- (183 Progress) <--------------------------- (183 Progress) <--------------------------- (200 OK) <--------------------------- (200 OK) <--------------------------- --------------------------------->(CANCEL) (200 canceling) <--------------------------- (200 OK) <--------------------------- (200 OK) <--------------------------- ----------------->(ACK) ----------------->(ACK) (BYE) <--------------------------- (BYE) <--------------------------- ----------------->(481 Call Leg/Transaction Does Not Exist) ----------------->(481 Call Leg/Transaction Does Not Exist) The B side answer with OK, after a while , a send a CANCEL. I don't know why Kamailio don't forward this message to the B side. B retry to send the OK message, then A send the ACK. At the end , B send BYE , but A don't have the transactin. In this situation, kamailio should deliver the "CANCEL" to the B side ? (even if, before B send the OK ) Or the proxy should not consider the CANCEL because B has answered with "OK" ? Where is the error on my configuration ? Thanks I have Kamailio 1.5.3 and my configuration is this: modparam("siptrace", "trace_on", 0) modparam("carrierroute", "config_source", "db") modparam("auth_db", "password_column", "ha1") modparam("auth_db", "use_domain", 1) modparam("usrloc", "db_mode", 2) #modparam("usrloc","nat_bflag", 8) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "db_mode", 1) modparam("permissions", "peer_tag_avp", "$avp(i:707)") modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") modparam("avpops","avp_table","usr_preferences") modparam("avpops","attribute_column","attribute") modparam("avpops","value_column","value") modparam("uac","from_restore_mode","auto") modparam("tm", "fr_timer", 12) modparam("tm", "wt_timer", 32) modparam("rr", "append_fromtag", 1) #modparam("rr", "enable_full_lr", 1) modparam("userblacklist", "use_domain", 0) #modparam("dialog", "dlg_flag", 2) modparam("registrar|nathelper", "received_avp", "$avp(i:42)") #modparam("registrar", "min_expires", 17000) modparam("acc", "db_flag", 2) modparam("acc", "db_missed_flag", 3) modparam("acc", "db_table_acc", "acc") modparam("acc", "db_table_missed_calls", "missed_calls") modparam("acc", "detect_direction", 1) route{ # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); exit; }; # ----------------------------------------------------------------- # Preprocessing # ----------------------------------------------------------------- if (!method=="REGISTER") record_route(); # ----------------------------------------------------------------- # NAT Detection # ----------------------------------------------------------------- force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(7); } # ----------------------------------------------------------------- # # ----------------------------------------------------------------- sip_trace(); ##Loose_route packets if (has_totag()) { if (loose_route()) { if(method=="BYE") { #Account BYE transactions setflag(2); }; route(2); } else { 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"); exit; } } sl_send_reply("404","Not here"); } exit; } # ----------------------------------------------------------------- # CANCEL processing # ----------------------------------------------------------------- if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; }; t_check_trans(); # ----------------------------------------------------------------- # # ----------------------------------------------------------------- if (method =="OPTIONS" || method=="SUBSCRIBE") {exit;}; if (method =="REGISTER") {route(3);}; route(1); } route[2] { t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; exit; } .. failure_route[2] { if (t_was_cancelled()) { # xlog("INFO:cancelled transaction--\n"); exit; } revert_uri(); .. t_on_failure("2"); append_branch(); if (!t_relay()) { exit; }; }

6 13

auth_db authorize realm
by Alex Balashov 06 May '10

06 May '10

There is a peculiar and confusing aspect to the documented significance of the "realm" argument to www_authorize(), and presumably proxy_authorize() as well. The documentation says that if this value is empty, the digest realm will be generated from the domain part of the To or From URI, whichever is applicable to the given situation (REGISTER vs. any other request). This is the way *_authorize() is invoked in most cases, and works fine. However, we recently ran into a situation where www_authorize() would always fail and claim that it could not find the user in 'subscriber' despite being provided correct username and domain, with the appropriate options -- return value -1. We were sending the public host IP as the domain of the To URI, using it as the realm, and setting it in the domain column of the 'subscriber' table. The problem was, the public IP of the host was not in /etc/hosts -- /etc/hosts consisted solely of: 127.0.0.1 localhost.localdomain localhost For some reason, it wasn't until I added the public IP into it that www_authorize() started working properly: 127.0.0.1 localhost.localdomain xxx.xxx.xxx.xxx public_host.domain.tld public_host I don't see anything different in the anatomy of the 401 Unauthorized challenges; the realm is still xxx.xxx.xxx.xxx in both cases. It just seems that unless Kamailio detects a DNS reverse alias for the domain, it won't properly authenticate requests. This aspect of the behaviour is not documented, and I am also confused as to why it happens this way. Any clarification would be appreciated. -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

2 1

segfault kamailio 3.01 avp_copy flag g
by Jon Bonilla 06 May '10

06 May '10

Hi all I have experienced segfaults when using avp_copy in Kamailio 3.01. This happens when using the "g" flag and the avp is not null (no problem if it has no content) avp_copy("$avp(s:block_out_list)", "$avp(s:caller_block_out_list)/gd"); Removing the "g" flags works ok (but does not copy all the content of course): avp_copy("$avp(s:block_out_list)", "$avp(s:caller_block_out_list)/d"); Directly copying the avp works fine: $avp(s:caller_block_out_list) = $avp(s:block_out_list); Logs: kernel: [1503237.523015] kamailio[8156]: segfault at 1008eb91c ip 505344 sp 7fffffffad10 error 4 in kamailio[400000+1c9000] WARNING: <core> [usr_avp.c:483]:AVP specified with index, but not used for search ALERT: <core> [main.c:722]: child process 8156 exited by a signal 11 ALERT: <core> [main.c:725]: core was not generated INFO: <core> [main.c:737]: INFO: terminating due to SIGCHLD INFO: <core> [main.c:788]: INFO: signal 15 received

2 10

Testing at SIPit 26
by Daniel-Constantin Mierla 06 May '10

06 May '10

Hello, anyone of you at SIPit 26 in about two weeks? http://sip-router.org/2010/05/05/testing-at-sipit-26/ http://www.kamailio.org/w/2010/05/testing-at-sipit-26/ I will be there ready to test 3.0 and devel for Kamailio and SER. If you want to test together or meet there, drop me an email. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

2 1

RPID Mod Question
by dotnetdub 06 May '10

06 May '10

Hi List, For certain Asterisk customers I want them to be allowed to manage and send their own CLI - we currently pass this onto providers as RPID. For example we also run a multi tenant pbx that uses kamailio as its proxy so it makes sense that we can send the callerID from this box to kamailio. Basically I will setup a prefix, check for that and if it matches I want to rewrite the RPID using the From"" from the SIP message.. From: "16172345671" <sip:username@domain>;tag=xxxxxxxxxxxx In this case I would send the RPID as <sip:16172345671@domain;party=calling;id-type=subscriber;privacy=off;screen=yes>" to our providers rather than reading the default value from the database for rpid. Where in the Kamailio / SR docs should I be looking for examples that would be similar / manipulating this type of info. Thanks, S.

3 3

Rtpproxy behind the NAT
by indiver 05 May '10

05 May '10

Hello Every one, My network setup is one to one nating of public and private ip. I want to run rtpproxy behind the NAT with in the local IP,but in vain. can't create listener error appears when i run rtpproxy on local ip. Can we run rtpproxy behind the NAT with local IP. If possible can any one specify the method. Regards, Indiver. -- View this message in context: http://old.nabble.com/Rtpproxy-behind-the--NAT-tp28458859p28458859.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 1

Kamailio SCTP support
by Francisco José Méndez Cirera 05 May '10

05 May '10

Hello, I've downloaded Kamailio 3.0.0 (the last release) and I´ve seen it´s possible downloading a "binary tar.gz" or the source to compile directly. I would like to know if the binary has enabled support for SCTP by default. If it´s enabled by default, how can I activate it (I can´t find any option related to sctp in kamailio.cfg) ? If it isn´t enabled by default, which are the dependencies? Is there a document or something with at least the main dependencies? Thank you very much. Bye!

3 2

new features: extended cfg preprocessor directives
by Daniel-Constantin Mierla 05 May '10

05 May '10

Hello, just to let you know about some new features related to config preprocessing. 3.0 added config preprocessor directives such as 'include_file' and 'define'. But the define could be used only to control which parts of config file are active. Several days ago I committed the code to allow you 'define' values for IDs, like: #!define MYINT 123 #!define MYSTR "xyz" The defined IDs are replaced at startup, during config parsing, e.g.,: $var(x) = 100 + MYINT; - is read as: $var(x) = 100 + 123; Moreover, you can have multi-line defined IDs: #!define IDLOOP $var(i) = 0; \ while($var(i)<5) { \ xlog("++++ $var(i)\n"); \ $var(i) = $var(i) + 1; \ } - then in routing block route { ... IDLOOP ... } A completely new thing is the substitution preprocessor directive: #!subst "/regexp/subst/" Defined IDs are not replaced within string values (because a cfg IDs is a standalone alphanumeric token, not enclosed in single or double quotes), so if you need to change something inside a string within config file, then use substitution. For example, if you want to replace the password in all db_url module parameters: #!subst "/DBPASSWD/xyz/" modparam("acc", "db_url", "mysql://user:DBPASSWD@localhost/db") modparam("auth_db", "db_url", "mysql://user:DBPASSWD@localhost/db") The value of a 'subst' directive can be any valid perl-like substitution expression, you can use wildcards, group tokens, backreferences, .... Hope this will make your cfg scripting easier. Testing and feedback is appreciated. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla

2 2

Simple Trunkig with Kamailio
by Jared Martin 05 May '10

05 May '10

Ok, this is probably an easy question: Say I have my kamailio server set up and all of my voip clients are gleefully calling each other... but now I want to connect them to the PSTN. Can I set up kamailo to trunk calls using a few grandstream gateways? or is Asterisk neccesary? Sorry if the answers out there, but I haven't found anything that answered the question directly. I'm a linux admin, but I'm new to PBX's and the like. I've installed kamailio in the past using instructions specific to that situation, but haven't configured trunking (or I didn't recognize thats what it was) references to howtos or tutorials would also be greatly appreciated Thanks for your help, Jared Martin

2 1

Error with 'Contact' field replacing
by Konstantin Shpinev 05 May '10

05 May '10

Hi! I have a problem with Kamailio. Sometimes (can not catch the moment and conditions of) situations arise when, calling from AAA to uri XXX call comes to uri YYY. The call comes from PSTN to the gray subnet of user-agents registered in kamailio. As the PSTN we using Audiocodes mediant. Kamailio ver. 1.5.3 Mediant host: 172.19.32.2, Kamailio host: 172.19.32.3 XXX host: 172.19.32.33, YYY host: 192.168.1.203 I put a line in the log section route_branch: > Apr 28 12:59:20 sipproxy /ks/sbin/kamailio[18294]: [INFO] new branch at sip:XXX@172.19.32.33:5060;user=phone Then in the log module acc wrote: > Apr 28 12:59:21 sipproxy /ks/sbin/kamailio[18302]: ACC: transaction answered: timestamp=1272473961;method=BYE;from_tag=1e76ae41;to_tag=1c1795565414;call_id=17 955646162422000234627(a)172.19.32.2;code=481;reason=Call/Transaction Does Not Exist;from_uri=sip:XXX@172.19.32.3 <sip%3AXXX(a)172.19.32.3>;from_username=XXX;from_name=;from_domain= 172.19.32.3;to_uri=sip:AAA@172.19.32.2 <sip%3AAAA(a)172.19.32.2>;to_username=AAA;to_name=;to_domain=172.19.32.2;request_uri=sip:AAA@172.19.32.2 <sip%3AAAA(a)172.19.32.2>;request_username=AAA;route_id=;route_name=;route_type_id=;destination=;calllist_id= Despite the error, the call passes, but to the another uri. Here's the log of the mediant: 1: > Apr 28 12:52:31 172.19.32.2 ( lgr_flow)(44244101 ) ---- Outgoing SIP Message to 172.19.32.3:5060 ---- Apr 28 12:52:31 172.19.32.2 ACK sip:XXX@172.19.32.33:5060;user=phone SIP/2.0^M Via: SIP/2.0/UDP 172.19.32.2;branch=z9hG4bKac1897823584^M Max-Forwards: 70^M From: <sip:AAA@172.19.32.2 <sip%3AAAA(a)172.19.32.2>>;tag=1c1795565414^M To: <sip:XXX@172.19.32.3 <sip%3AXXX(a)172.19.32.3>>;tag=1e76ae41^M Call-ID: 17955646162422000234627(a)172.19.32.2^M CSeq: 1 ACK Contact: <sip:AAA@172.19.32.2 <sip%3AAAA(a)172.19.32.2>>^M Route: <sip:172.19.32.3;lr;ftag=1c1795565414;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA->^M Supported: em,timer,replaces,path,early-session,resource-priority^M Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE^M *User-Agent: Audiocodes-Sip-Gateway-Mediant 2000/v.5.00A.045.003*^M Content-Length: 0 As you can see, in 1. all is well. But then: 2: > ..... ---- Incoming SIP Message from 172.19.32.3:5060 ---- ....From: <sip:AAA@172.19.32.2 <sip%3AAAA(a)172.19.32.2>>;tag=1c1795565414^M > Call-ID: 17955646162422000234627(a)172.19.32.2^M CSeq: 1 INVITE^M Allow: > INVITE, ACK, CANCEL,OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO^M > Content-Type: application/sdp^M *User-Agent: Zoiper for Windows rev.1105*^M > Content-Length: 232^M ^M v=0^M *o=Zoiper_user 0 0 IN IP4 192.168.1.203^M > s=Zoiper_session^M c=IN IP4 192.168.1.203^*M t=0 0^M m=audio 8000 RTP/AVP > 8 0 101^M a=rtpmap:8 PCMA/8000^M a=rtpmap:0 PCMU/8000^M a=rtpmap:101 > telephone-event/8000^M a=fmtp:101 0-15^M a=sendrecv And that is where the substitution occurred at the data fields 'Contact' an entirely different user-agent (uri YYY), which is also registered on Kamailio. That it eventually comes to call. *Repeat: In second SIP message from Kamailio 'Contact' field is replaced by 'Contact' of another (!) user-agent (YYY).* There is a suspicion that an error occurs in a block: > if (!lookup("location")) { switch ($retcode) { case -3: ... case -1: ... case -2: ... } } } I myself anywhere 'Contact' field are not affected, so all suspected to *locate() *from module Registrar.

2 1

kamailio and presence_xml parse error
by Francesco 05 May '10

05 May '10

Hi to all. I'm new here. I'm getting crazy with presence_xml error. I'm using kamailio 1.5.4 notls with xlite 3.0 (winXP client) softphone and postgresql 8.4 as db backend. Seems that presence works, but i have error in parsing xml: those are the relevant log: Apr 29 16:42:54 ocs /usr/sbin/kamailio[2462]: INFO:presence:send_notify_request: NOTIFY sip:user1@ocs.ita.domain.net<sip%3Auser1(a)ocs.ita.domain.net>via sip:user1@10.44.17.145:24258 on behalf of sip:user2@ocs.ita.domain.net<sip%3Auser2(a)ocs.ita.domain.net>for event presence Apr 29 16:42:54 ocs /usr/sbin/kamailio[2463]: ERROR:presence_xml:agregate_xmls: while parsing xml body message Apr 29 16:42:54 ocs /usr/sbin/kamailio[2463]: ERROR:presence_xml:pres_agg_nbody: while aggregating body When i login on kamailio, the users are not immediatly notified to my presence...but, when there's a change of status, the client are notified and on softphone the status change. Other that, if the client close softphone, i can watch this on the log: Apr 29 16:42:41 ocs /usr/sbin/kamailio[2461]: ERROR:presence_xml:offline_nbody: while parsing xml memory Apr 29 16:42:41 ocs /usr/sbin/kamailio[2461]: ERROR:presence_xml:pres_agg_nbody: while constructing offline body Apr 29 16:42:41 ocs /usr/sbin/kamailio[2461]: INFO:presence:send_notify_request: NOTIFY sip:user1@ocs.ita.domain.net<sip%3Auser1(a)ocs.ita.domain.net>via sip:user1@10.44.17.145:24258 on behalf of sip:user3@ocs.ita.domain.net<sip%3Auser3(a)ocs.ita.domain.net>for event presence And the client, on the other softphone, results on line, rather then offline. I'm thinking about encoding database, i've tryied with SQL ASCII and UTF-8 too. Same behaviour. I try different releases (1.5.3 no tls) on different distribution (debian, lucid, alpine linux). Same behaviour. Obiouvsly, the softphone are right configured (presence agent is enabled). Any hints? Thanks for you kind attention. :: Francesco Colista :: Email : francesco.colista(a)gmail.com :: Jabber: francesco(a)jabber.org

3 5

Re: [SR-Users] sr-users Digest, Vol 60, Issue 6
by Rothe, Marcus, D22-WHV 05 May '10

05 May '10

"sr-users-request(a)lists.sip-router.org" <sr-users-request(a)lists.sip-router.org> schrieb: Send sr-users mailing list submissions to sr-users(a)lists.sip-router.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users or, via email, send a message with subject or body 'help' to sr-users-request(a)lists.sip-router.org You can reach the person managing the list at sr-users-owner(a)lists.sip-router.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Authentication SER + RADIUS + LDAP (Pablo Ros) 2. Kamailio SCTP support (Francisco Jos? M?ndez Cirera) 3. Re: NAT Traversal IPTel example (Daniel-Constantin Mierla) 4. Re: Kamailio SCTP support (Daniel-Constantin Mierla) ---------------------------------------------------------------------- Message: 1 Date: Tue, 4 May 2010 10:03:26 +0200 From: Pablo Ros <prf1987(a)gmail.com> Subject: [SR-Users] Authentication SER + RADIUS + LDAP To: sr-users(a)lists.sip-router.org Message-ID: <m2i46878ffc1005040103m27d01ccfo9c84d8bba7dc7896(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Hello, We have a LDAP database with many users information and this is the one we use to implement most of our services; on the contrary we have SER working with a SQL data base. Our intention was to make also the authentication against LDAP. After some research, we've seen there's no specific module for SER to work with LDAP and we have considered some alternatives among them there was the "module" from ETH world<http://www.ethworld.ethz.ch/technologies/sipeth/ser_modules/ldap>. However, we didn't manage to make it work (if it's an advisable choice we'd appreciate some clues). So, we decided to make the authentication through the RADIUS server. Nevertheless, we are having some problems with the way data is sent. When doing the user authentication there's no problem as it is sent in plain text and we modified to do it against the email attribute as it's this what we want. It makes it perfectly. But it turns out that when we try to make the password authentication, as the data sent from SER comes in a hash (user:realm:password) as long as we know, we don't really know how to make it compare with the password field in LDAP (under MD5 algorithm as well). When we make a test over Radius by sending plain text it works perfectly so it shouldn't be a problem by searching the attributes over LDAP. We have tried to follow instructions to set the digest section properly but there's something we definitely miss. Attached there's a log from the radius when trying to log with SER and the Register section from SER. log: 03 User-Name = "my.user(a)i2cat.net" Digest-Attributes = 0x0a0b7061626c6f2e726f73 Digest-Attributes = 0x010b69326361742e6e6574 Digest-Attributes = 0x022a34626466643065343837303734363630626261366134363437663730313034343639 663532306532 Digest-Attributes = 0x040f7369703a69326361742e6e6574 Digest-Attributes = 0x030a5245474953544552 Digest-Response = "6c95bcba1fca30e976fa9295025b1bf4" Service-Type = Sip-Session Sip-Uri-User = "my.user" NAS-Port = 5060 NAS-IP-Address = 127.0.0.1 +- entering group authorize ++[preprocess] returns ok ++[chap] returns noop ++[mschap] returns noop rlm_digest: Adding Auth-Type = DIGEST ++[digest] returns ok rlm_realm: Looking up realm "i2cat.net" for User-Name = " my.user(a)i2cat.net" rlm_realm: No such realm "i2cat.net" ++[suffix] returns noop rlm_eap: No EAP-Message, not doing EAP ++[eap] returns noop ++[unix] returns notfound ++[files] returns noop rlm_ldap: - authorize rlm_ldap: performing user authorization for my.user(a)i2cat.net WARNING: Deprecated conditional expansion ":-". See "man unlang" for details expand: (mail=%{Stripped-User-Name:-%{User-Name}}) -> (mail= my.user(a)i2cat.net) expand: ou=activat,ou=personal,dc=i2cat,dc=net -> ou=activat,ou=personal,dc=i2cat,dc=net rlm_ldap: ldap_get_conn: Checking Id: 0 rlm_ldap: ldap_get_conn: Got Id: 0 rlm_ldap: attempting LDAP reconnection rlm_ldap: (re)connect to ldap.i2cat.net:389, authentication 0 rlm_ldap: bind as cn=anonim,dc=i2cat,dc=net/i2mngr to ldap.i2cat.net:389 rlm_ldap: waiting for bind result ... rlm_ldap: Bind was successful rlm_ldap: performing search in ou=activat,ou=personal,dc=i2cat,dc=net, with filter (mail=pablo.ros(a)i2cat.net) rlm_ldap: No default NMAS login sequence rlm_ldap: looking for check items in directory... rlm_ldap: LDAP attribute userPassword as RADIUS attribute Digest-HA1 == "{md5}nCK4tZ5NNP48oT0wlXX+Jw==" rlm_ldap: looking for reply items in directory... WARNING: No "known good" password was found in LDAP. Are you sure that the user is configured correctly? rlm_ldap: user pablo.ros(a)i2cat.net authorized to use remote access rlm_ldap: ldap_release_conn: Release Id: 0 ++[ldap] returns ok ++[expiration] returns noop ++[logintime] returns noop rlm_pap: WARNING! No "known good" password found for the user. Authentication may fail because of this. ++[pap] returns noop rad_check_password: Found Auth-Type DIGEST auth: type "digest" +- entering group authenticate rlm_digest: Digest-HA1 has invalid length, authentication failed. ++[digest] returns invalid auth: Failed to validate the user. Login incorrect: [my.user(a)i2cat.net/<via Auth-Type = DIGEST>] (from client localhost port 5060) Found Post-Auth-Type Reject +- entering group REJECT expand: %{User-Name} -> my.user(a)i2cat.net attr_filter: Matched entry DEFAULT at line 11 ++[attr_filter.access_reject] returns updated Delaying reject of request 0 for 1 seconds SER register -> User Authentication part #------------------------------------------------------------------------ # Comprovacio de credencials per als usuaris. #------------------------------------------------------------------------ if (!is_user_in("From", "noauth")) { xlog("L_NOTICE", "SER-INFO: challenging user...\n"); # IMPORTANTE: radius_www_authorize solo toma un par?metro! if(!radius_www_authorize("")) { # L'usuari NO esta registrat correctament o les # credencials no son valides! www_challenge("i2cat.net","0"); xlog("L_ALERT","SER-ALERT r[4]-Bad Auth from <%fu>:(%is) [403 Forbiden]\n"); sl_send_reply("403", "Forbiden!, Bad Credentials"); break; #tallem la comunicacio }; #-------------------------------------------------------------------- # check_to #-------------------------------------------------------------------- if(!check_to()) { xlog("L_ALERT","SER-ALERT: check_to(): REG Spoofed attempt <%fu>:(%is)\n"); sl_send_reply("403", "Use To=id la proxima vegada :@"); consume_credentials(); # fem que caduqui la sessio break; }; } -- Pablo Ros

1 0

Authentication SER + RADIUS + LDAP
by Pablo Ros 05 May '10

05 May '10

Hello, We have a LDAP database with many users information and this is the one we use to implement most of our services; on the contrary we have SER working with a SQL data base. Our intention was to make also the authentication against LDAP. After some research, we've seen there's no specific module for SER to work with LDAP and we have considered some alternatives among them there was the "module" from ETH world<http://www.ethworld.ethz.ch/technologies/sipeth/ser_modules/ldap>. However, we didn't manage to make it work (if it's an advisable choice we'd appreciate some clues). So, we decided to make the authentication through the RADIUS server. Nevertheless, we are having some problems with the way data is sent. When doing the user authentication there's no problem as it is sent in plain text and we modified to do it against the email attribute as it's this what we want. It makes it perfectly. But it turns out that when we try to make the password authentication, as the data sent from SER comes in a hash (user:realm:password) as long as we know, we don't really know how to make it compare with the password field in LDAP (under MD5 algorithm as well). When we make a test over Radius by sending plain text it works perfectly so it shouldn't be a problem by searching the attributes over LDAP. We have tried to follow instructions to set the digest section properly but there's something we definitely miss. Attached there's a log from the radius when trying to log with SER and the Register section from SER. log: 03 User-Name = "my.user(a)i2cat.net" Digest-Attributes = 0x0a0b7061626c6f2e726f73 Digest-Attributes = 0x010b69326361742e6e6574 Digest-Attributes = 0x022a34626466643065343837303734363630626261366134363437663730313034343639663532306532 Digest-Attributes = 0x040f7369703a69326361742e6e6574 Digest-Attributes = 0x030a5245474953544552 Digest-Response = "6c95bcba1fca30e976fa9295025b1bf4" Service-Type = Sip-Session Sip-Uri-User = "my.user" NAS-Port = 5060 NAS-IP-Address = 127.0.0.1 +- entering group authorize ++[preprocess] returns ok ++[chap] returns noop ++[mschap] returns noop rlm_digest: Adding Auth-Type = DIGEST ++[digest] returns ok rlm_realm: Looking up realm "i2cat.net" for User-Name = " my.user(a)i2cat.net" rlm_realm: No such realm "i2cat.net" ++[suffix] returns noop rlm_eap: No EAP-Message, not doing EAP ++[eap] returns noop ++[unix] returns notfound ++[files] returns noop rlm_ldap: - authorize rlm_ldap: performing user authorization for my.user(a)i2cat.net WARNING: Deprecated conditional expansion ":-". See "man unlang" for details expand: (mail=%{Stripped-User-Name:-%{User-Name}}) -> (mail= my.user(a)i2cat.net) expand: ou=activat,ou=personal,dc=i2cat,dc=net -> ou=activat,ou=personal,dc=i2cat,dc=net rlm_ldap: ldap_get_conn: Checking Id: 0 rlm_ldap: ldap_get_conn: Got Id: 0 rlm_ldap: attempting LDAP reconnection rlm_ldap: (re)connect to ldap.i2cat.net:389, authentication 0 rlm_ldap: bind as cn=anonim,dc=i2cat,dc=net/i2mngr to ldap.i2cat.net:389 rlm_ldap: waiting for bind result ... rlm_ldap: Bind was successful rlm_ldap: performing search in ou=activat,ou=personal,dc=i2cat,dc=net, with filter (mail=pablo.ros(a)i2cat.net) rlm_ldap: No default NMAS login sequence rlm_ldap: looking for check items in directory... rlm_ldap: LDAP attribute userPassword as RADIUS attribute Digest-HA1 == "{md5}nCK4tZ5NNP48oT0wlXX+Jw==" rlm_ldap: looking for reply items in directory... WARNING: No "known good" password was found in LDAP. Are you sure that the user is configured correctly? rlm_ldap: user pablo.ros(a)i2cat.net authorized to use remote access rlm_ldap: ldap_release_conn: Release Id: 0 ++[ldap] returns ok ++[expiration] returns noop ++[logintime] returns noop rlm_pap: WARNING! No "known good" password found for the user. Authentication may fail because of this. ++[pap] returns noop rad_check_password: Found Auth-Type DIGEST auth: type "digest" +- entering group authenticate rlm_digest: Digest-HA1 has invalid length, authentication failed. ++[digest] returns invalid auth: Failed to validate the user. Login incorrect: [my.user(a)i2cat.net/<via Auth-Type = DIGEST>] (from client localhost port 5060) Found Post-Auth-Type Reject +- entering group REJECT expand: %{User-Name} -> my.user(a)i2cat.net attr_filter: Matched entry DEFAULT at line 11 ++[attr_filter.access_reject] returns updated Delaying reject of request 0 for 1 seconds SER register -> User Authentication part #------------------------------------------------------------------------ # Comprovacio de credencials per als usuaris. #------------------------------------------------------------------------ if (!is_user_in("From", "noauth")) { xlog("L_NOTICE", "SER-INFO: challenging user...\n"); # IMPORTANTE: radius_www_authorize solo toma un parámetro! if(!radius_www_authorize("")) { # L'usuari NO esta registrat correctament o les # credencials no son valides! www_challenge("i2cat.net","0"); xlog("L_ALERT","SER-ALERT r[4]-Bad Auth from <%fu>:(%is) [403 Forbiden]\n"); sl_send_reply("403", "Forbiden!, Bad Credentials"); break; #tallem la comunicacio }; #-------------------------------------------------------------------- # check_to #-------------------------------------------------------------------- if(!check_to()) { xlog("L_ALERT","SER-ALERT: check_to(): REG Spoofed attempt <%fu>:(%is)\n"); sl_send_reply("403", "Use To=id la proxima vegada :@"); consume_credentials(); # fem que caduqui la sessio break; }; } -- Pablo Ros

2 1

NAT Traversal IPTel example
by Andy Savage 05 May '10

05 May '10

Hi there, I've been setting up SER for a testing server on the local network behind a NAT. I'm having problems understanding how to setup the NAT routing in the configuration file. My understanding is that IPTel has already setup advanced NAT routing on the free service (as per the website). Is it possible to get a copy of the configuration file (atleast the part that has the NAT routing stuff). This would help me immensely in getting proper nat traversal setup as it already works great on the IPTel server. Not sure who I would contact in regards to this, but this seemed like a good place to start. Kind regards, Andy Savage -- "The greatest challenge to any thinker is stating the problem in a way that will allow a solution" - Bertrand Russell Andy Savage Cell Phone: +852 936 34341 Skype ID: andy_savage Linked In: http://www.linkedin.com/in/andysavage

2 1

RTPproxy in bridge mode question
by Uriel Rozenbaum 04 May '10

04 May '10

Guys, I'm successfully using a Kamailio + RTPproxy setup in bridge mode with most of my Gateways. My setup includes two different interfaces one with a public IP and teh other with the private IP. Now I'm facing some slight issue. Some providers won't accept my calls (or calls will have some strange behavior) if the Contact header has an IP out of immediate range. I tried to use fix_nated_contact() function but as per my topology, this function will not change the contact header because the IP is already the one on the interface. Example: U 192.168.200.X:5060 -> 192.168.200.Y:5060 INVITE sip:111160911097@192.168.200.Y SIP/2.0. Via: SIP/2.0/UDP 192.168.200.X:5060;branch=z9hG4bK096baacc;rport. From: "Uriel Rozenbaum" <sip:60911100@192.168.200.X>;tag=as32794d5e. To: <sip:111160911097@192.168.200.Y>. Contact: <sip:60911100@*192.168.200.X*>. U 200.A.A.A:5060 -> 200.B.B.B:5060 INVITE sip:898960911097@200.B.B.B SIP/2.0. Record-Route: <sip:200.A.A.A;r2=on;lr=on;ftag=as32794d5e>. Record-Route: <sip:192.168.200.Y;r2=on;lr=on;ftag=as32794d5e>. Via: SIP/2.0/UDP 200.A.A.A;branch=z9hG4bK5222.14fbf4f7.0. Via: SIP/2.0/UDP 192.168.200.X:5060;received=192.168.200.X;branch=z9hG4bK096baacc;rport=5060. From: "Uriel Rozenbaum" <sip:60911100@192.168.200.X>;tag=as32794d5e. To: <sip:111160911097@192.168.200.Y>. Contact: <sip:60911100@*192.168.200.X*>. Is there any way to let know Kamailio the outgoing IP I'll be using and fix the contact accordingly? I can trigger this change after I know the destination IP. Thanks! Uriel

4 7

kamailio and presence_xml parse error
by Francesco 03 May '10

03 May '10

Thx for reply. This is what i get with patch applied (is not pretty readable, but what i can see is "^M" char that could be the cause of the problem...or in rpid field the status "unknown") May 3 15:50:36 ocs /usr/sbin/kamailio[1858]: ERROR:presence_xml:agregate_ xmls: while parsing xml body message [[<?xml version='1.0' encoding='UTF-8'?><presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' entity=' sip:amusso@ocs.ita.wtbts.net <sip%3Aamusso(a)ocs.ita.wtbts.net>'><tuple id='t7164a248'><status><basic>open</basic></status></tuple><dm:person id='pf31ded47'><rpid:activities><rpid:unknown/></rpid:activities></dm:person></presence>P Server^M Content-Length: 0^M ^M ]] May 3 15:50:36 ocs /usr/sbin/kamailio[1858]: ERROR:presence_xml:pres_agg_nbody: while aggregating body May 3 15:51:22 ocs /usr/sbin/kamailio[1855]: ERROR:presence_xml:pres_agg_nbody: while aggregating body May 3 15:51:22 ocs /usr/sbin/kamailio[1856]: ERROR:presence_xml:offline_nbody: while parsing xml memory May 3 15:51:22 ocs /usr/sbin/kamailio[1856]: ERROR:presence_xml:pres_agg_nbody: while constructing offline body May 3 15:51:51 ocs /usr/sbin/kamailio[1857]: ERROR:presence_xml:agregate_xmls: while parsing xml body message [[<?xml version='1.0' encoding='UTF-8'?><presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' entity=' sip:fcolista@ocs.ita.wtbts.net <sip%3Afcolista(a)ocs.ita.wtbts.net>'><tuple id='tdd57124f'><status><basic>open</basic></status></tuple><dm:person id='pe22c066e'><rpid:activities><rpid:unknown/></rpid:activities></dm:person></presence>resence>]] May 3 15:51:51 ocs /usr/sbin/kamailio[1857]: ERROR:presence_xml:pres_agg_nbody: while aggregating body May 3 15:52:32 ocs /usr/sbin/kamailio[1856]: ERROR:presence_xml:agregate_xmls: while parsing xml body message [[<?xml version='1.0' encoding='UTF-8'?><presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' entity=' sip:amusso@ocs.ita.wtbts.net <sip%3Aamusso(a)ocs.ita.wtbts.net>'><tuple id='t7164a248'><status><basic>open</basic></status></tuple><dm:person id='pf31ded47'><rpid:activities><rpid:unknown/></rpid:activities></dm:person></presence>resence>P Server^M Content-Length: 0^M ^M 74fe7e7c241d"^M Server: AlpineLinux SIP Server^M Content-Length: 0^M ^M ]] May 3 15:52:32 ocs /usr/sbin/kamailio[1856]: ERROR:presence_xml:pres_agg_nbody: while aggregating body May 3 15:53:26 ocs /usr/sbin/kamailio[1858]: ERROR:presence_xml:agregate_xmls: while parsing xml body message [[<?xml version='1.0' encoding='UTF-8'?><presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' entity=' sip:amusso@ocs.ita.wtbts.net <sip%3Aamusso(a)ocs.ita.wtbts.net>'><tuple id='t7164a248'><status><basic>open</basic></status></tuple><dm:person id='pf31ded47'><rpid:activities><rpid:unknown/></rpid:activities></dm:person></presence>e> tbts.net", nonce="4bded515000000284f85e37237864bdc5da8d6836da5a769"^M Server: AlpineLinux SIP Server^M Content-Length: 0^M ^M ]] May 3 15:53:26 ocs /usr/sbin/kamailio[1858]: ERROR:presence_xml:pres_agg_nbody: while aggregating body :: Francesco Colista :: Email : francesco.colista(a)gmail.com :: Jabber: francesco(a)jabber.org

1 0

[Kamailio-Users] re : openserctl statistics..
by Jignesh Gandhi 03 May '10

03 May '10

Hello, What does the following stat mean when you run the " openserctl moni " sl: received_ACKs = xxxx I did some digging and I got the following explanation... But am not sure if this is an error in operser or an error on a message received by openser or what? " sl: received_ACKs - number of received ACKs due sending negative replies. " any help is appreciated. thanks, --Jignesh

6 11

Add header on failure Route
by Alex Rendour 03 May '10

03 May '10

Hi, We need to add header on the failure route. Exemple: On failure 404 Not found, we need add header Reason: Q.850; cause=1 We tried this in failure_route: if(t_check_status("404")){ xlog("L_ERR", "404 Not found\n"); append_to_reply("Reason: Q.850; cause=1\r\n"); xlog("L_ERR", "404 Not found apres\n"); exit; } and if(t_check_status("404")){ xlog("L_ERR", "404 Not found\n"); append_hf("Reason: Q.850; cause=1\r\n"); xlog("L_ERR", "404 Not found apres\n"); exit; } The twice doesn't work. Someone can help me? Thanks, -- Alexandre Rendour Acropolis Telecom <http://www.acropolistelecom.net> <mailto:rendour@acropolistelecom.net> Adresse : 161-163 avenue Gallieni Paris - Porte de Bagnolet 93170 Bagnolet

2 2

Simple Trunking with Kamailio
by Jared Martin 01 May '10

01 May '10

Ok, this is probably an easy question: Say I have my kamailio server set up and all of my voip clients are gleefully calling each other... but now I want to connect them to the PSTN. Can I set up kamailo to trunk calls using a few grandstream gateways? or is Asterisk neccesary? Sorry if the answers out there, but I haven't found anything that answered the question directly. I'm a linux admin, but I'm new to PBX's and the like. I've installed kamailio in the past using instructions specific to that situation, but haven't configured trunking (or I didn't recognize thats what it was) references to howtos or tutorials would also be greatly appreciated Thanks for your help, Jared Martin

2 1

sr-users = ser + kamailio users mailing lists
by Daniel-Constantin Mierla 30 Apr '10

30 Apr '10

Hello, a short note that the merging of SIP Express Routers (SER) users mailing list into sr-users is completed from subscriber point of view: - posts to serusers(a)lists.iptel.org are directed to sr-users(a)lists.sip-router.org - posts to sr-users(a)lists.sip-router.org are received by subscribers of serusers(a)lists.iptel.org Just a bit of time ago, kamailio (openser) users' mailing list was merged as well. Since the release 3.0.x is practically the same source code, no mater the binary is compiled as ser or kamailio, and the development is in the same place (code in same repository), a combined mailing lists environment is the right for providing the best answers in shortest time for both communities. Therefore, now there are two mailing lists: - sr-users(a)lists.sip-router.org - to be used for discussions about stable versions and generic project/community topics - sr-dev(a)lists.sip-router.org - to be used for discussions about devel version and future development topics If you were subscribed to both with same email address, you won't get duplicates. If you had some settings that are no longer in place, please contact one of devels on sr-dev lists or write to me. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla

1 0

Bug with avp's or feature ?
by Tristan Mahé 30 Apr '10

30 Apr '10

Hi guys, I was debugging an install on kamailio 3.0, and was asked to add per user custom outbound proxies. To do so, I used load_credentials from auth_db to store the custom route in an avp, as you can see in the relevant part below. I noticed a strange thing, POINT 1 show "OK", but POINT 2 shows "KO". Is it a normal behaviour that avp got deleted after the Dialog PART ? Regards, Gled --------------------------------------------------------------------------------- modparam("auth_db", "load_credentials", "$avp(s:proxy)=outbound_override;$avp(s:rights)=rights;$avp(s:max_out)=max_out") route[INVITE_FROM_UAC] { if (!www_authorize("", "my_user_view")) { www_challenge("", "1"); } if ( !($tU =~ $avp(s:rights_ereg)) ) { sl_send_reply("403","Forbidden"); t_release(); exit(); } /* POINT 1 */ if ( is_avp_set("$avp(s:proxy)") ) { xlog("OK"); } else { xlog("KO"); } get_profile_size("outbound","$fU","$avp(calls_out)"); if ( $avp(s:max_out) != 0 && $avp(calls_out) >= $avp(s:max_out) ) { t_reply("403","Too Many calls"); t_release(); exit(); } /* Dialog PART */ if (!is_in_profile("outbound")) { dlg_manage(); set_dlg_profile("outbound","$fU"); } /* POINT 2 */ if ( is_avp_set("$avp(s:proxy)") ) { xlog("OK"); } else { xlog("KO"); } } ---------------------------------------------------------------------------------

2 1

[Kamailio-Users] PCRE Regex / Transformations Bug?
by Brandon Armstead 30 Apr '10

30 Apr '10

Hello, I'm attempting to use the following in Kamailio: $(hdr(from){re.subst,/(.*?);(.*)/\1/}) --- however it does not match / replace, with this from header sample: From: <sip:714364XXXX@XX.XX.XX.XX ;pstn-params=808282808882;cpc=unknown>;tag=gK0b53f6d9 However if this is fully pcre compatible, then I am a bit lost as I've checked the results in php, perl, and python produces: From: <sip:714364XXXX@XX.XX.XX.XX (expected result) However, Kamailio does not match / replace / etc using the following: xlog("L_INFO", "Test From Header Regex: $(hdr(from){re.subst,/(.*?);(.*)/\1/})"); Code Samples / Regex Pattern Replacements: PHP: http://codepad.org/CUttQH1r Perl: http://codepad.org/ZfAO4X5v Python: http://codepad.org/T14SYWkZ Any comments / help / etc is always appreciated, look forward to hearing back from you all! Sincerely, Brandon Armstead

4 9

Re: [SR-Users] [sr-dev] Hello, please unsubscribe my email from the mail list, thank you.
by Iñaki Baz Castillo 30 Apr '10

30 Apr '10

2010/4/30 Lee Jacky <jacky.lee888(a)yahoo.com.cn>: > Hello,please unsubscribe my email:jacky.lee888@yahoo.com.cn from the mail list,thank you. List-Unsubscribe: <http://lists.sip-router.org/cgi-bin/mailman/options/sr-dev>, <mailto:sr-dev-request@lists.sip-router.org?subject=unsubscribe> -- Iñaki Baz Castillo <ibc(a)aliax.net>

1 0

[Kamailio-Users] Exotic case (but real) in which RtpProxy doesn't work
by Iñaki Baz Castillo 28 Apr '10

28 Apr '10

Hi, I've a problem with an Alcatel PBX in the way it performs transference: - Alcatel (behind NAT) sends INVITE to 111 and Kamailio forces RtpProxy. Let's assume the selected UDP port for RtpProxy is 1000. - Alcatel sends INVITE to 222 and Kamailio forces RtpProxy. Let's assume the selected UDP port for RtpProxy is 2000. Then Alcatel performs the transference as follows: - It sends a re-INVITE for 111 to Kamailio by setting the SDP to the IP or RtpProxy and port 2000. - It also sends a re-INVITE for 222 to Kamailio by setting the SDP to the IP or RtpProxy and port 1000. This is, Alcatel wants that the provider (me) sends the RTP to itself, while mantaining the original SIP dialogs established (so it's ok at signalling level, but at RTP level it cannot work as RtpProxy shoud send RTP to itself). I'm thinking on how to solve it but find no solution. Any suggestion? Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

3 7

[permissions] 'address' table and mask priorirty
by Iñaki Baz Castillo 28 Apr '10

28 Apr '10

Hi, some time ago I asked if allow_source_address_group() gives preference to the entries in 'address' table with lowest mask. I was replied that such case is not analyzed. But the fact is that in my tests it just works: Entry 1: - grp = 1 - ip_addr = 9.9.9.9 - mask = 32 Entry 2: - grp = 2 - ip_addr = 9.9.9.0 - mask = 24 I send a request from IP 9.9.9.9 and the function allow_source_address_group() always returns "1", it doesn't matter if the entry 1 appears before or after the entry 2, neither their "id" value. So it works as I expected (even if it's not explained in the documentation). However after checking the module code it seems that there are just two cases: - IP addresses (mask = 32). - Subnets (mask != 32). So first the source address is always matched against he address hash, and if it doesn't match then it is matched against the subnet hash, but in this last case it is not taken into account any kind of priority based on mask value, am I right? Could somebody confirm it please? Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

3 3

[LCR openser 1.2] Strange behavior in "lcr" table
by Iñaki Baz Castillo 28 Apr '10

28 Apr '10

Hi, I have configured LCR in an existing OpenSER 1.2 installation by filling the "lcr" table as follows: id prefix from_uri grp_id priority 1 NULL NULL 1 1 2 NULL NULL 2 2 In "gw" table there are some entries with "grp_id=1" and some others with "grp_id=2". Then in the script I just call "load_gws()" as usual, expecting that first is ALWAYS selected a gateway with grp_id=1. In fact I have ths *same* configuration working as expected in Kamailio 1.5. But for some reason, in OpenSer 1.2 for some calls "load_gws()" selects gateways with "grp_id=2", which IMHO doesn't make sense at all as grp_id 2 has priority 2 and its gateways would be selected always after those belonging to grp_id 1. Is there any bug or different behavior in LCR in openser 1.2? or do I miss something? Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 2

old mailing lists serusers
by Pavel Kasparek 28 Apr '10

28 Apr '10

Hello all, the email address for old iptel.org mailing list serusers(a)lists.iptel.org (and serusers(a)iptel.org) is now forwarded to the new sip-router project mailing list sr-users(a)lists.sip-router.org. Pavel iptel.org

1 0

old mailing lists serusers
by Pavel Kasparek 28 Apr '10

28 Apr '10

Hello all, the email address for old iptel.org mailing list serusers(a)lists.iptel.org (and serusers(a)iptel.org) is now forwarded to the new sip-router project mailing list sr-users(a)lists.sip-router.org. Pavel iptel.org

1 0

old SER bugtracker turned off
by Pavel Kasparek 27 Apr '10

27 Apr '10

Hello all, the old SER bug tracking system http://tracker.iptel.org/ was shut down. There is now a small page at http://tracker.iptel.org/ with links to: - the sip-router project bugtracker http://sip-router.org/tracker/ - the new iptel.org bugtracker http://bugtracker.iptel.org/, where Sems and Serweb projects are now and which is used also for reporting problems with iptel.org public SIP service There is also a read-only archive of old SER bugs (exported from the old bugtracker), just for reference: http://tracker.iptel.org/jira-ser/ser-oldbugs.html Pavel

1 0

[Serusers] old SER bugtracker turned off
by Pavel Kasparek 27 Apr '10

27 Apr '10

Hello all, the old SER bug tracking system http://tracker.iptel.org/ was shut down. There is now a small page at http://tracker.iptel.org/ with links to: - the sip-router project bugtracker http://sip-router.org/tracker/ - the new iptel.org bugtracker http://bugtracker.iptel.org/, where Sems and Serweb projects are now and which is used also for reporting problems with iptel.org public SIP service There is also a read-only archive of old SER bugs (exported from the old bugtracker), just for reference: http://tracker.iptel.org/jira-ser/ser-oldbugs.html Pavel

1 0

termination with user/pwd
by Anders 27 Apr '10

27 Apr '10

I'm sure this has been asked before, but I couldn't find the answer anywhere when I googled it. I have a temporary situation where I need to terminate calls to a carrier with user/pwd authentication, and not through the IP authentication. As I understand it this is not a normal setup situation for a gw in Kamailio - but is there a way I can do this? Thanks, Anders

4 4

Sintax errors can stop the machine?
by Bruno Machado 26 Apr '10

26 Apr '10

We are facing a periodically problem with a server that is running Kamailio 1.5.2. I think that it is not the case, but I will ask anyway. Some clients, sometimes, send us messages with sintax errors. This machine needs to be rebooted every week. It stops working completely. I don't think that SIP messages with this kind of errors can lead the machine to its 'death'. This proxy is very simple: it just receives and forwards the calls to pstn carriers. The routing rules are defined in the script, so we dont use modules to take the IPs and all other information about the carriers. The database is used just for ACC. Nothing more. Also, it is not receiving a flood atack. We have several servers running at this time with a huge number of clients and we never had this problem. This machine is the only one that does not want to work without a weekly break :) The causes, in my opnion, will be with the operational system or something physical. But, I would like to know if somebody already saw this behavior with SIP messages. tks a lot bruno machado

2 1

B2BUA removing a=nortpproxy:yes (rtpproxy-1 <---RTP---> rtpproxy-2)
by Iñaki Baz Castillo 26 Apr '10

26 Apr '10

Hi, I must deal with a B2BUA which keeps the original SDP unchanged except the fact that it removes the "a=nortpproxy:yes" line added by RtpProxy. The B2BUA intercommunicates two Kamailio, both forcing its own RtpProxy server. I've not tested it again but expect there will be no RTP as each RtpProxy will wait for RTP coming from the other. Am I right? Of course I could force RtpProxy just in NAT cases and so, but I prefer to force it always so the clients of each Kamailio can configure QoS for the media addresses (different in each server). Is there some workaround for it? In the other hand: is it possible two different Kamailio's to share the same RtpProxy server? I've never tryied it. If it's valid, would it solve the B2BUA issue? (I don't think so as there are still different media sessions created by different kamailio instances). Any thought? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

6 24

htable size
by Alex Balashov 26 Apr '10

26 Apr '10

The htable size is static; it doesn't dynamically resize itself up to the next power of 2 after a certain utilisation threshold, right? -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

2 5

Re-invite problems with t.38?
by Espen Berg 26 Apr '10

26 Apr '10

Hello all! I finally have something that I could call a working config (almost that is), the proxy works fine with voice and fax sent over g.711 (but I guess re-invites will cause problems in this scenario also). The problem occurs when I try to switch to T.38. Since it works with voice my guess is that the problem is caused by the re-invite to t.38. I guess it has to be solved by a if (loose_route()) section, but I am little bit clueless right now, I have tried multiple variants but with no success. RTP proxy is started with the -l flag to distinguish between local and public traffic. -l PUBLIC.IP.ETH0/LOCAL.IP.ETH1 The setup: Sonus UAC/Trunk <-private ip-> (eth1)Kamailio/RTPProxy(eth0) <-public ip-> Asterisk IP.TO.TRUNK = Sonus IP, public address range, but only reachable by private net (from the range of LOCAL.IP.ETH1). LOCAL.IP.ETH1 = Local net, static route to IP.TO.TRUNK. PUBLIC.IP.ETH0 = Public routable address. IP.TO.ASTERISK = Public routable address. The purpose of Kamailio/RTPProxy are to proxy everything between the routable and the private net. Does anyone here have any idea of how I'm going to solve my issue? It would also be nice if anyone here have better solutions to my approach, and there could also be some other errors in my config that causes this behaviour. ####### Routing ######## route { if (msg:len >= 2048) { sl_send_reply("513", "Message too big"); exit; } if (method != "INVITE" && method != "ACK" && method != "BYE" && method != "CANCEL") { sl_send_reply("503", "Service Unavailable"); exit; } if (!has_totag()) { record_route(); route("1"); } if (has_totag()) { if (!loose_route()) { if (t_check_trans()) { t_relay(); exit; } exit; } } if (has_body("application/sdp")) { if (method == "INVITE") { route("2"); t_on_reply("2"); } else if (method == "ACK") { rtpproxy_answer(); } } else if (method == "INVITE") { t_on_reply("2"); } if (method == "BYE" || method == "CANCEL") { unforce_rtp_proxy(); } t_relay(); } #RELAY BY SOURCE ROUTE route[1] { if (src_ip == "IP.TO.TRUNK" || dst_ip == "LOCAL.IP.ETH1") { force_send_socket("PUBLIC.IP.ETH0"); rewritehost("IP.TO.ASTERISK"); } else { force_send_socket("LOCAL.IP.ETH1"); rewritehost("IP.TO.TRUNK"); } } # RTP OFFER ROUTE route[2] { if (src_ip == "IP.TO.TRUNK" || dst_ip == "LOCAL.IP.ETH1") force_rtp_proxy("eir"); else force_rtp_proxy("ier"); } # RTP OFFER ONREPLY onreply_route[1] { if (has_body("application/sdp")) route("2"); } # RTP ANSWER ONREPLY onreply_route[2] { if (has_body("application/sdp")) route("2"); } ## END OF CONFIG Sorry for the long post, but I hope someone outthere are able to help med with this issue. -- Espen C. Berg, Driftstekniker - Monsternett AS (www.monsternett.no) Violgata 3A, N-1776 HALDEN, NORWAY Sentralbord: 69701800 - Direkte: 69701810 - Faks: 69701801

2 6

meeting in Berlin, June 7-8
by Daniel-Constantin Mierla 25 Apr '10

25 Apr '10

Hello, several developers will be around Berlin in June 7-8 time frame and we think to organize a face to face meeting. It is a good time frame if you want to attend Amoocon in Rostock (June 4-6, aprox 230km north of Berlin) or LinuxTag (June 9-12, in Berlin). http://www.linuxtag.org/2010/ http://www.amoocon.de/2010 I will have two presentations at Amoocon, about Asynchronous processing in SIP Router / Kamailio and extending your routing logic with Lua (via the new app_lua module). Hennig has one talk at LinuxTag about Geographical redundant VoIP systems with Kamailio. So far, interest in event and participation was shown by: - Andrei Pelinescu-Onciul - Jiri Kuthan - Inaki Baz Castillo - Jesus Rodriguez - Elena-Ramona Modroiu - Daniel-Constantin Mierla - Henning Westerholt - Marcus Hunger A good number to move it into public space and see if others from community are interested in such event. Based on that we will decide the type of event and look for a proper place to host it. Among topics could be: hands on hacking or docs update sessions, quick start guide to start writing your own extensions as module, open debates about current devel state and future directions... Please drop an email if you want to participate to the event - there is no participation fee - however, late registrations may be rejected if the capacity of location is filled - priority is first come first served. You can email me at miconda(a)gmail.com or to registration(a)lists.sip-router.org Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla

1 0

[OT] Hitler should use Kamailio
by Iñaki Baz Castillo 24 Apr '10

24 Apr '10

http://www.youtube.com/watch?v=4nKh8bg3eHo -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 1

LCR in openser 1.2 - is it possible to disable a gateway?
by Iñaki Baz Castillo 24 Apr '10

24 Apr '10

Hi, I need to add LCR support to an existing OpenSer 1.2 server (not possible to upgrade it for now). I've realized that 1.2 doesn't include gateways monitorize feature. Ok, I can do the check with an external tool, but I would like some way to dissable such gw from the LCR module, something like upgrading the gw table and performing the MI reload function. But I don't see how to set in the gw table that such gateway is disabled. Is it possible? Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 1

Solutions to missing BYEs, accounting for them
by Alex Balashov 23 Apr '10

23 Apr '10

Hi all, Please forgive the slightly long post, but if you have anything to contribute on this topic, please consider giving it a read as I could really use your input. :-) As I'm sure many others of you running proxy-based service delivery platforms of some description also, I am faced with the problem of trying to account for calls with missing BYEs in a realistic way. There is no shortage of mailing list posts over the years on this topic. Inevitably, in a platform with sufficient call volume, with some NAT'd and/or endpoint diversity and other technical causes, there will be some calls that are never officially terminated from the point of view of a proxy. The ability of the 'dialog' module to spoof bidirectional BYEs on timeout[1] goes a long way toward addressing this problem theoretically. However, there are practical obstacles to relying on it solely as a solution, mainly because there is not an acceptable timeout value to use as a trade-off. If the timeout period is set to a very low value, users will obviously complain, and in any case, depending on the destination, the worst-case scenario for maximum call billing may still be far too high. If the timeout period is set high--perhaps something like 5-8 hours--then all calls that fail to end in the normal way will be billed some excessively large amount that certainly will not sit well with users either. If either the core delivery element of the platform or the user agent is tightly controlled by the operator of the proxy from an administrative point of view, it is indeed probably possible to rely on RTP timeouts or SIP Session Timers (SSTs) on one of the endpoints. That doesn't create a satisfying resolution for those of us dealing with indeterminate call completion scenarios with a great deal of user and vendor diversity, though. For instance, I route to about 15 ITSPs and carriers; I think maybe one of them does 15-minute SSTs, and the rest are certainly not going to turn them on just for me, even if their SBCs/switches/things have the capability. The user endpoints are mostly Asterisk and do RTP timeout, of course, and in most cases I do get the resulting BYE. However, this discussion is about the minute but nontrivial percentage of cases in which I do not get the BYE, whether because of NAT statekeeping problems or network reachability or whatever underlying causes--in truth, I cannot accurately characterise these. So, it seems to me that from a theoretical point of view, there are basically two directions someone in this position can go from here: 1) Inline B2BUA in the signaling path of all calls; 1a) Make it do SSTs; or 1b) Make it relay media, too, and hang up the call (bidirectional BYE) on RTP receive timeout; 2) Couple the proxy to an RTP relay and provide some mechanism by which the proxy can be made aware, in an asynchronous fashion, that an RTP timeout was detected by the relay. It seems to me from a brief and informal survey of prior mailing list literature that #1 is the usually recommended option here. If #1 is pursued, what is the best tool to use in the Kamailio/SIP-Router-oriented ecosystem? My default instinct would say SEMS; I really like SEMS, and use it a lot for various related chores. The problem is that the pre-built modules and examples for SEMS mostly center on application-level functionality, while low-level documentation of its powerful C++ API is a bit impoverished, so this would take a lot of work. Needless to say, I am interested in the option that requires the least work but still solves the problem in an elegant way from a technical and--dare I say--aesthetic perspective. For instance, it seems clear from looking at the SEMS-1.1.1 sources that SSTs are supported in principle in core/plug-in/session_timer. But unless I am missing something, I cannot find anywhere in the sources or examples where it is actually used. So, I suppose one option is to figure out how to make this stuff work in SEMS, and make it work. But for some reason who is not attune to the universe of its C++ API, it is a rather formidable chore. I think the same would hold true of making it observe bidirectional RTP timeout. Turning attention to option #2, I have looked at rtpproxy (my preferred default), iptrtpproxy, and mediaproxy modules but have not found any evidence that the control protocols Kamailio/SR uses to engage them support any notion of backward asynchronous feedback in case of RTP timeout. It would be really nice if one of these stream control protocols was augmented to kick back a packet to Kamailio that can be caught in a special event_route, like event_route[nathelper:rtp-stream-timeout], but that is clearly not the case today. To be honest, I would not use MediaProxy even if it had this feature, because, well, let's be bluntly honest and acknowledge what the more politically aware presumably already conjecture: in light of AG Projects' zealous OpenSIPS partnership, it's difficult to muster confidence in future compatibility of MediaProxy with Kamailio. The module is there, it works, and I'm sure its maintainers are dedicated to doing whatever it takes to reverse engineer and keep it working, lift patches from OpenSIPS as necessary, etc., but who wants to be on the wrong side of the project ecosystem fence? Not I. That leaves iptrtpproxy, whose 'switchboard' concept I do not fully comprehend due to lack of experience with it, but which holds a potentially viable, if slightly kludgy/Rube Goldbergian answer. Of the three RTP proxies, it is the only one that provides a ready means of exporting a list of media streams it is currently tracking, together with statistics on how many packets have been received, etc. It is not inconceivable to cook up an external process that will frequently check this 'switchboard', as it were, and incite Kamailio/SR to do dlg_bye() via MI if it appears that the media stream has disappeared from either side; the dialog module helpfully exports the MI command dlg_end_dlg. Still, this does not seem nearly as parsimonious and reliable a solution as simply building some kind of RTP stream leg timeout notification into the control socket. After all, the control socket is open persistently, right, not on-demand? The various RTP proxies all seem to have some kind of dead peer detection internally in order to have some means of gracefully expiring resources allocated to media streams that have gone away, so it would just be a matter of passing a control frame up the socket to Kamailio/SR and wiring that to a custom event_route or a more static callback in the code. By the way, I should mention that I am aware of and historically very sympathetic to the perspective that this kind of call control is alien to the nature of a proxy, and an appropriate job for UAs and not proxies at all. However, we all have to make pragmatic concessions to the realities of real-world operation, which I assume is the motivation for dialog timeouts, dlg_bye(), and other perversions from the point of view of a purist. :-) I welcome your thoughts and suggestions about the easiest and most technically meritorious approach. Thanks, -- Alex [1] Enabled via $dlg_ctx(timeout_bye) = 1 -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

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MI functions
by Hector.Ortiz＠swisscom.com 23 Apr '10

23 Apr '10

Hi everyone, In the page where a module is described, there is a list of dependencies that have to be met in order to use the module, the list of exported parameters that might need to be set in the configuration file and a list of exported functions which can be used in the configuration script. I've noticed as well, that there are some "Exported MI functions" but I was wondering if any of you know what the purpose of those functions is, where and how such functions could be used? Thanks Héctor

2 1

Error with Presence: no shmem
by Tristan Mahé 23 Apr '10

23 Apr '10

Hi List, I'm writing a little message because I encountered today a strange error on my Kamailio 3.0 proxy/presence server ( and can't ask on IRC there's nobody atm ;p ) I get these errors in the logs since a few hours: Apr 23 14:31:41 proxy-b /proxy/bin/kamailio/sbin/kamailio[29008]: ERROR: tm [t_msgbuilder.c:1371]: build_uac_req(): no shmem Apr 23 14:31:41 proxy-b /proxy/bin/kamailio/sbin/kamailio[29008]: ERROR: tm [uac.c:327]: t_uac: Error while building message Apr 23 14:31:41 proxy-b /proxy/bin/kamailio/sbin/kamailio[29008]: ERROR: presence [notify.c:1577]: in function tmb.t_request_within Apr 23 14:31:41 proxy-b /proxy/bin/kamailio/sbin/kamailio[29008]: ERROR: presence [notify.c:1673]: sending Notify not successful Apr 23 14:31:41 proxy-b /proxy/bin/kamailio/sbin/kamailio[29008]: ERROR: presence [notify.c:1348]: Could not send notify for dialog That prevents NOTIFYs from being sent, but I can also see sometimes NOTIFYs get sent without theses errors. The strange thing is that the error indicate no shared memory, but on the server, I'm far from being memless (16G total RAM, 4G free not even used for cache ). I have presence, presence_xml, presence_dialoginfo, pua, pua_dialoginfo and pua_bla loaded. Anyone got an idea on where to look ? Regards, Gled.

2 1

[Kamailio-Users] How can i install and configuration CDRTool + Freeradius + mediaproxy + kamailio
by RAJNIKANT VANZA 23 Apr '10

23 Apr '10

Hi All, I would like to use CDRTool for accounting and integrate with kamailio + mediaproxy + freeradius. so, please let me know about configuration or installation steps for CDRToolon centos-5.X os. I have followed this link for CDRTool configuration http://cdrtool .ag-projects.com/wiki/Install but not success and confused. I have seen above url for CDRTool that provides information for deban os. but i want to install on centos. I am installed and run kamailio and mediaproxy successful. I am confused about compile or install about CDRTool and i want to help about Freeradius integration with kamailo and CDRTool. Thanks in advance. -- Best Regards, Rajnikant

3 2

possible bug with dialog module
by Kelvin Chua 23 Apr '10

23 Apr '10

currently using kamailio 3.0 with around 1000 users. Apr 21 10:19:57 kamvm-1 /usr/local/sbin/kamailio[28511]: ERROR: dialog [dlg_handlers.c:218]: bad sip message or missing Contact hdr Apr 21 10:19:57 kamvm-1 /usr/local/sbin/kamailio[28511]: ERROR: dialog [dlg_handlers.c:350]: could not add further info to the dialog Apr 21 10:19:57 kamvm-1 kernel: kamailio[28511]: segfault at 0000000000000088 rip 00002ab61b62779a rsp 00007fffec157440 error 4 Apr 21 10:19:57 kamvm-1 /usr/local/sbin/kamailio[28507]: ALERT: <core> [main.c:722]: child process 28511 exited by a signal 11 Apr 21 10:19:57 kamvm-1 /usr/local/sbin/kamailio[28507]: ALERT: <core> [main.c:725]: core was generated Kelvin Chua

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