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42 participants
37517 discussions

time for acc records
by Daniel-Constantin Mierla 11 Aug '11

11 Aug '11

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

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Re: [SR-Users] Default kamailio installation, syntax errors with Defined Values
by Gertjan Wolzak 11 Aug '11

11 Aug '11

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

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Default kamailio installation, syntax errors with Defined Values
by Gertjan Wolzak 11 Aug '11

11 Aug '11

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

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Using Kamailio to convert TCP to TLS
by aur.hoffmann＠free.fr 10 Aug '11

10 Aug '11

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

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$hu is not documented
by Juha Heinanen 10 Aug '11

10 Aug '11

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

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Cancel issue
by Carl Wagner 09 Aug '11

09 Aug '11

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

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Modifying $rU after lookup()
by Efelin Novak 09 Aug '11

09 Aug '11

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

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Re: [SR-Users] Using SR for Number Translation & Number Portabilty Applications
by Henning Westerholt 09 Aug '11

09 Aug '11

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

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Re: [SR-Users] Using SR for Number Translation & Number Portabilty Applications
by Henning Westerholt 08 Aug '11

08 Aug '11

> Yaron wrote: > Thanks for the quick response. I have already played with the sqlops - > this is an obvious way to do it. I will further investigate cr and sqlops > and decide which is best for my purposes. Hello Yaron, > I liked the idea of having several instances of SIP-ROUTERs which are > independent of each other- I understand that cr answers it. Indeed, this is one advantage of using an in-memory cache. But you probably need some more DB access on the servers for things like authentication. > I have another question - I was trying to add alpha-numeric character > into the cr prefix and it seems it is not supported. We are using 'D' prefixes > in our network in order to route between different elements in the network. > Is there a way to support it? Maybe another routing module? The cr module support also a extended matching mode, but it will be probably not really feasible with millions of routes: http://kamailio.org/docs/modules/stable/modules/carrierroute.html#id2651893 Another thing you could do is use another "domain" for the D prefix, e.g. one for the default routing, and then another one for the D case. The module support matching for the domain as well. You could maybe also use flags for matching. Other possible modules for PSTN routing are lcr and mtree. Regards, Henning

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Using SR for Number Translation & Number Portabilty Applications
by Yaron Nachum 08 Aug '11

08 Aug '11

Hi Everyone, I have downloaded SR r 3.1.4 and have been playing around with it. I am considering implementing Number Translation services (1800...) and Number Portability Services on it. We have around 20K entries on the Number Translation Service and around 3M entries on the Number portability Service. The volume is around 100 caps. I have started working with the carrierroute module - it seems perfect for the Number translation Service, but I am not sure about the Number portability. I have several questions: 1. Is it possible to use this module with several millions of entries? 2. What is the impact of loading the table? Is it possible to do it during traffic hours? I have read about PDB module, but I haven't understood how to setup the PDB server. I would appreciate for any guidance. Would you recommend using the PDB module for the Number Portability or the carrierroute? Thanks.

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