sr-users August 2011

sr-users@lists.kamailio.org

70 participants
107 discussions

by Juha Heinanen

after reading mtree readme, it is not clear to me if mt_match() matches to longest or any matching prefix in the three. for example, if i have tree with prefixes 00 and 00358 and string to be matched is 0035892345670, will mt_match() set pv to value associated with 00 or 00358? also, if mt_ignore_duplicates is set, will 00358 be considered as duplicate of 00 or just prefixes that are exactly same. finally, what are possible mt_tree_type values and what types they stand for? -- juha

12 years, 8 months

kamailio - proxy/relay

by Sunny

Hi list, I'm trying to configure kamailio to work as a OBP (Outbound Proxy). Basically KAMAILIO receive requests from UAs, after that it acts like a client and send those requests to the IP-PBX. On the same way the responses are forwarded from IP-PBX to the UAs. [UA] --> [KAMAILIO] --> [IP-PBX ] [UA] <-- [KAMAILIO] <-- [IP-PBX ] Following possible call flow: [UA ] -> [KAMAILIO] SIP/SDP Request: INVITE sip:12345678909@[KAMAILIO], with session description [KAMAILIO] -> [UA ] SIP Status: 100 Trying [KAMAILIO] -> [IP-PBX ] SIP/SDP Request: INVITE sip:12345678909@[IP-PBX ], with session description [IP-PBX ] -> [KAMAILIO] SIP Status: 100 Trying [IP-PBX ] -> [KAMAILIO] SIP Status: 407 Proxy Authentication Required [KAMAILIO] -> [IP-PBX ] SIP Request: ACK sip:12345678909@[IP-PBX ] [KAMAILIO] -> [UA ] SIP Status: 407 Proxy Authentication Required [UA ] -> [KAMAILIO] SIP Request: ACK sip:12345678909@[KAMAILIO] [UA ] -> [KAMAILIO] SIP/SDP Request: INVITE sip:12345678909@[KAMAILIO], with session description [KAMAILIO] -> [UA ] SIP Status: 100 Trying [KAMAILIO] -> [IP-PBX ] SIP/SDP Request: INVITE sip:12345678909@[IP-PBX ], with session description [IP-PBX ] -> [KAMAILIO] SIP Status: 100 Trying [IP-PBX ] -> [KAMAILIO] SIP/SDP Status: 183 Session Progress, with session description [KAMAILIO] -> [UA ] SIP/SDP Status: 183 Session Progress, with session description [IP-PBX ] -> [KAMAILIO] SIP/SDP Status: 180 Ringing, with session description [KAMAILIO] -> [UA ] SIP/SDP Status: 180 Ringing, with session description (...) [UA ] -> [KAMAILIO] SIP Request: CANCEL sip:12345678909@[KAMAILIO] [KAMAILIO] -> [UA ] SIP Status: 200 OK [KAMAILIO] -> [IP-PBX ] SIP Request: CANCEL sip:12345678909@[IP-PBX ] [IP-PBX ] -> [KAMAILIO] SIP Status: 200 OK [IP-PBX ] -> [KAMAILIO] SIP Status: 487 Request Terminated [KAMAILIO] -> [IP-PBX ] SIP Request: ACK sip:12345678909@[IP-PBX ] [KAMAILIO] -> [UA ] SIP Status: 487 Request Terminated [UA ] -> [KAMAILIO] SIP Request: ACK sip:12345678909@[KAMAILIO] Wondering if anyone could help to make this configuration work? Thanks, Sunny

12 years, 8 months

time for acc records

by Daniel-Constantin Mierla

Hello, it seems that is some parts of the world it is required to have call duration with two decimals precision. By default acc module does the timestamp in seconds. Thinking of adding an option to save the milliseconds, I see two option: - a new column to store the milliseconds part as integer - a new column to store the seconds.milliseconds as double For those using various billing engines, what would be easier to digest when rating the CDRs? There is already the option of doing db extra column to store $TV(Sn), but this value is stored in a string (varchar) format, giving a little more troubles to convert to numeric format before being able to compute duration, so it might just worth doing the C coding to store in right data type. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

12 years, 8 months

Re: [SR-Users] Default kamailio installation, syntax errors with Defined Values

by Gertjan Wolzak

Sorry, my bad.... Missing dependencies Regards, Gertjan Wolzak

12 years, 8 months

Default kamailio installation, syntax errors with Defined Values

by Gertjan Wolzak

Hello, Just been installing Kamailio, version 3.1.x via Git, but when I use the default kamailio.cfg, Kamailio will not start and the debug gives the following errors. 0(25876) : <core> [cfg.y:3330]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 285, column 29-35: syntax error 0(25876) : <core> [cfg.y:3330]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 285, column 29-35: Invalid arguments 0(25876) : <core> [cfg.y:3333]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 285, column 36: ERROR: bad config file (4 errors) When I check the line, column it is the following line of the acc params: modparam("acc", "log_flag", FLT_ACC) I assume the variable FLT_ACC is not set or read correctly. But as it is the default kamailio.cfg that seems strange to me. The defined values are: ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://openser:openserrw@localhost/openser" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 Does anyone has an idea why te values are not loaded into the variables.? With regards, Gertjan Wolzak

12 years, 8 months

Using Kamailio to convert TCP to TLS

by aur.hoffmann＠free.fr

Hello, I would like to use Kamailio as a TCP <=> TLS convertor ? Do you think it's possible ? I have developed a software (which simulate a SIP phone) which can use UDP or TCP (not TLS). I need a SIP proxy to convert TCP to TLS. For example: UA => TCP OPTION (5060) => Kamailio => TLS OPTION (5061) => OXE UA <= TCP 200 OK (5060) <= Kamailio <= TLS 200 OK (5061) <= OXE Do you think Kamailio can do that ? Thanks in advance, Regards

12 years, 9 months

$hu is not documented

by Juha Heinanen

xcap_server examples refer to pseudo variable $hu, which is not documented on pseudo variables wiki page: http://sip-router.org/wiki/cookbooks/pseudo-variables/devel -- juha

12 years, 9 months

Cancel issue

by Carl Wagner

Hi, I am having an issue getting Kamailio to properly cancel a call. I initiate the call with an INVITE from Asterisk and then hang up. According to rfc3665 the call flow should be: 3.8. Unsuccessful No Answer Alice Proxy 1 Proxy 2 Bob | | | | | INVITE F1 | | | |--------------->| INVITE F2 | | | 100 F3 |--------------->| INVITE F4 | |<---------------| 100 F5 |--------------->| | |<---------------| | | | | 180 F6 | | | 180 F7 |<---------------| | 180 F8 |<---------------| | |<---------------| | | | CANCEL F9 | | | |--------------->| | | | 200 F10 | | | |<---------------| CANCEL F11 | | | |--------------->| | | | 200 F12 | | | |<---------------| | | | | CANCEL F13 | | | |--------------->| | | | 200 F14 | | | |<---------------| | | | 487 F15 | | | |<---------------| | | | ACK F16 | | | 487 F17 |--------------->| | |<---------------| | | | ACK F18 | | | 487 F19 |--------------->| | |<---------------| | | | ACK F20 | | | |--------------->| | | In my case I only have one proxy - the Kamailio box, and the call is coming from Asterisk instead of Alice. The is the call flow that I am seeing: Asterisk Kamailio SIP_phone | | | | INVITE | | |--------------->| INVITE | | 100 |--------------->| |<---------------| 100 | | |<---------------| | | | | | 180 | | 180 |<---------------| |<---------------| | | CANCEL | | |--------------->| | | | CANCEL | | |--------------->| | 200 Canceling | | |<---------------| | | | 200 | | |<---------------| | | | | | | | | 487 | | |<---------------| | | ACK | | 487 |--------------->| |<---------------| | | ACK | | |--------------->| | | CANCEL | | |--------------->| | | 200 Canceling | | |<---------------| | | CANCEL | | |--------------->| | | 200 Canceling | | |<---------------| | It looks like the problem is that that "200 Canceling" has a To: tag and it shouldn't. ====================================== # cancel from Asterisk: U 2011/05/24 15:18:56.123714 3.3.3.3:5060 -> 2.2.2.2:5060 CANCEL sip:SIP_13034445555@2.2.2.2 SIP/2.0. Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK51c9602a;rport. From: "+13032223333" <sip:+13032223333@3.3.3.3>;tag=as4c415d87. To: <sip:SIP_13034445555@2.2.2.2>. Call-ID: 605b8de40a05924177085d71154bda77(a)3.3.3.3. CSeq: 102 CANCEL. User-Agent: VerbalWorld. Max-Forwards: 70. Content-Length: 0. # 200 canceling from Kamailio to asterisk: U 2011/05/24 15:18:56.127349 2.2.2.2:5060 -> 3.3.3.3:5060 SIP/2.0 200 canceling. Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK51c9602a;rport=5060. From: "+13032223333" <sip:+13032223333@3.3.3.3>;tag=as4c415d87. To: <sip:SIP_13034445555@2.2.2.2>;tag=4ed8ab9f3ed29a77613bb14b3431ff43-2f34. Call-ID: 605b8de40a05924177085d71154bda77(a)3.3.3.3. CSeq: 102 CANCEL. Server: kamailio (3.1.3 (i386/linux)). Content-Length: 0. ====================================== Why is kamialio adding a To: tag? Portion of kamailio.cfg ===================== ... route { ... route(WITHINDLG); ### only initial requests (no To: tag) from here on if (is_method("CANCEL")) { if (t_check_trans()) { xlog("L_INFO", "MAIN: Recieved Cancel, t_check_trans is true\n"); ... t_relay(); exit; } ... Thanks! Carl

12 years, 9 months

Modifying $rU after lookup()

by Efelin Novak

Hi Folks, I would like to ask how can I change the $rU (user part of uri) in multiple appended branches? In my scenario I do lookup("location") and after this I try to modify an username using the $rU variable. This works fine until there are several records in the location table for the given user (registered from different locations). In this case only one branch is changed and unchanged uri is sent to an another location. Is there a way to change it in all branches together? Thanks for the answer. Regards, Efelin

12 years, 9 months

Re: [SR-Users] Using SR for Number Translation & Number Portabilty Applications

by Henning Westerholt

On Monday 08 August 2011, Yaron Nachum wrote: > I have seen the match_mode but I haven't seen how is it possible to set it > up per domain. > > I would appreciate if you could explain how. Hi Yaron, please always CC to the mailling list. Its not possible to set the match_mode per domain, its a global parameter. With regards to the domains, this was another option you could use, e.g. use domain 0 for calls without the 'D' prefix, and then domain D for calls with 'D' prefix. Best regards, Henning

12 years, 9 months

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sr-users August 2011