October 2010 - sr-users

Re: [SR-Users] setdebug() and 3.X
by Daniel-Constantin Mierla 30 Oct '10

30 Oct '10

On 10/29/10 5:40 PM, Sergey Okhapkin wrote: > I can't test now - high volume traffic is running. But I looked at the diff, it > seems to me OK, I see no reason why it could not work:-) Because of the law, the Murphy law ;-) Probably I will test sometime next week if you don't do it before and then backport. Cheers, Daniel > On Friday 29 October 2010, you wrote: >> On 10/29/10 4:33 PM, Daniel-Constantin Mierla wrote: >>> On 10/29/10 4:09 PM, Sergey Okhapkin wrote: >>>> I just realized that 3.1 executes main routing block when broken SIP >>>> message >>>> is received! But sanity_check() function returns 0 on failed checks >>>> and script >>>> execution stops, my favorite xlog() is not executed. >>> OK, that can be fixed very easy. Iirc, in some cases you don't get the >>> sip message in the config if it is really broken, but I am not sure if >>> error_route was executed for that case anyhow. >> I committed on git master, id: >> >> 39a8108c62a2beafba18417613b9660a8abef86a >> >> >> Can you take the patch and test it on 3.1? If all is fine, then I can >> backport - even the default cfg expects non-drop behavior since it >> attempts to print a log message there. >> >> Cheers, >> Daniel >> >>> Cheers, >>> Daniel >>> >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: ERROR:<core> >>>> [parser/msg_parser.c:177]: ERROR: get_hdr_field: bad to header >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: INFO:<core> >>>> [parser/msg_parser.c:351]: ERROR: bad header field [To: Chi Nhung<sip: >>>> ] >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: ERROR:<core> >>>> [parser/msg_parser.c:177]: ERROR: get_hdr_field: bad to header >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: INFO:<core> >>>> [parser/msg_parser.c:351]: ERROR: bad header field [To: Chi Nhung<sip: >>>> ] >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: ERROR:<core> >>>> [parser/msg_parser.c:177]: ERROR: get_hdr_field: bad to header >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: INFO:<core> >>>> [parser/msg_parser.c:351]: ERROR: bad header field [To: Chi Nhung<sip: >>>> ] >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: ERROR:<core> >>>> [msg_translator.c:1946]: ERROR: build_res_buf_from_sip_req: alas, parse >>>> _headers failed >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: WARNING: sanity >>>> [sanity.c:253]: sanity_check(): check_required_headers(): failed to s >>>> end 400 via sl reply >>>> >>>> On Friday 29 October 2010, Daniel-Constantin Mierla wrote: >>>>> On 10/29/10 11:53 AM, Sergey Okhapkin wrote: >>>>>> Daniel, I think this could be implemented without re-introduction of >>>>>> error_route, >>>>> it will not be re-introduced for sure, we have now a different core >>>>> framework which care reuse event_route mechanism. >>>>> >>>>> In the past the error route was executed when the error was discovered, >>>>> so it several places of the code, and I am not sure that the core can >>>>> continue up to config execution of something is really broken. >>>>> >>>>> I will have it in mind when I get some spare time. >>>>> >>>>> Cheers, >>>>> Daniel >>>>> >>>>>> what about core parameter route_on_error=yes/no (default no)? If >>>>>> route on error is enabled, then sanity module will do the trick: >>>>>> >>>>>> route{ >>>>>> if(!sanity_check("1159")) { >>>>>> xlog("L_INFO","Bad message from >>>>>> $proto:$si:$sp\n$mb\n"); >>>>>> break; >>>>>> } >>>>>> .... >>>>>> >>>>>> It will be up to script writer what to do in main routing block if >>>>>> message parsing fails. >>>>>> >>>>>> On Friday 29 October 2010, Sergey Okhapkin wrote: >>>>>>> It would be nice to get error_route back. The route was trivial in my >>>>>>> cfg file: >>>>>>> >>>>>>> error_route { >>>>>>> xlog("L_INFO","Bad message from $proto:$si:$sp\n$mb\n"); >>>>>>> drop(); >>>>>>> } >>>>>>> >>>>>>> On Friday 29 October 2010, Daniel-Constantin Mierla wrote: >>>>>>>> On 10/28/10 1:41 PM, Sergey Okhapkin wrote: >>>>>>>>> I've found one more feature missed in 3.1 - error_route is >>>>>>>>> eliminated >>>>>>>>> and sanity module added. But if received SIP message is >>>>>>>>> malformed and >>>>>>>>> can't be parsed, routing script is not executed and I have no >>>>>>>>> way to >>>>>>>>> log the message to kamailio log. I did it in error_route before. >>>>>>>>> >>>>>>>>> This logging is important to me to support customers. SIP ALG in >>>>>>>>> routers often breaks Via line :-( >>>>>>>> Since I developed error_route, I remember that it handled just >>>>>>>> sporadic >>>>>>>> cases, but I can add an event_route to pass the buffer and some >>>>>>>> hints >>>>>>>> to configuration file in case the initial parsing fails and the >>>>>>>> message >>>>>>>> is dropped before getting to config file. >>>>>>>> >>>>>>>> It is good to know that someone use such case, introduction of >>>>>>>> sanity >>>>>>>> was clearly mentioned and so far there was no complain. >>>>>>>> >>>>>>>> Cheers, >>>>>>>> Daniel >>>>>>>> >>>>>>>>> On Thursday 28 October 2010, Daniel-Constantin Mierla wrote: >>>>>>>>>> On 10/26/10 2:22 PM, Sergey Okhapkin wrote: >>>>>>>>>>> setdebug() function is no longer provided by core. How to change >>>>>>>>>>> debug level dynamically in the script to trace execution of a few >>>>>>>>>>> statements only? >>>>>>>>>> seems was lost in transition to 3.0, I will check to see what >>>>>>>>>> can be >>>>>>>>>> done with the new architecture. >>>>>>>>>> >>>>>>>>>> Thanks, >>>>>>>>>> Daniel >>>>>>>>> _______________________________________________ >>>>>>>>> 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 >>>> _______________________________________________ >>>> 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 -- Daniel-Constantin Mierla http://www.asipto.com

2 1

NAT ping replies
by Sergey Okhapkin 29 Oct '10

29 Oct '10

Kamailio 3.1 log is filled with lines like Oct 29 03:32:46 west /usr/local/sbin/kamailio[632]: INFO: <script>: incoming reply from udp:188.62.4.186:65333 SIP/2.0 404 Not Found Via: SIP/2.0/UDP 204.74.213.5:5060;branch=0 From: <sip:pinger@sip.callwithus.com>;tag=3993f0c To: <sip:188.62.4.186:65333> Call-ID: aff32da5-89ef0136-71dc8c5(a)204.74.213.5 CSeq: 1 OPTIONS Content-Length: 0 Oct 29 03:32:46 west /usr/local/sbin/kamailio[632]: INFO: <core> [forward.c:774]: broken reply to forward - no 2nd via Shouldn't replies to NAT ping messages be silently dropped like in 1.4?

2 2

Billing party and radius accounting
by Efelin Novak 29 Oct '10

29 Oct '10

Hi folks, I have a problem with setting billing party in my radius messages. My accounting is done by CDRTool via radius messages to radius server. The thing I don`t know is how to set billing party variable (by default $avp(s:billing_party) with reliable value. I can try adding Diversion header and fill the variable with it, but can I trust the Diversion header? When UAC sets redirect(302) and Invite comes, he replies 302 and Kamailio passes this message to origin. However is it possible to change Diversion header by UAC? This could be way how to make free calls and bill it to someone else. I was thinking of uac_redirect module. If I get it right this module saves original number in memory and does not use message itself to store it. In documentation there is written that accounting can by done by setting modparam("uac_redirect", "acc_function", "acc_rad_request"). Even when I use exported function set_accept_filter() the accounting itself isn't done. I`d like to get original number from module uac_redirect and set it to $avp(s:billing_party). This would solve the problem of billing forwarded calls. Am I getting it right? Is there anything else what can be passed to billing party variable to make correct bililng? I'm using kamailio 3.1.x and CDRTool v 8.0.6 Thanks, Efelin

4 12

tm.t_uac_start examples from CLI
by Alex Balashov 29 Oct '10

29 Oct '10

Can someone post an example of usage of tm.t_uac_start from the command line using sercmd over a UNIX domain socket (e.g. /tmp/ser_ctl)? I am quite unclear on how to pass minimally necessary header values to it in this form and cannot seem to get the syntax right. Thanks! -- 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

problem getting siriemis to login
by Gerry Kernan 29 Oct '10

29 Oct '10

Hi I am trying to get siremis working with kamailio, butwhen i open the login page and enter the username and password nothing happens. I have followed the instruction on the siremis home page. Created mysql db called openser and inserted username admin& password admin. mysql> use openser; Reading table information for completion of table and columnnames You can turn off this feature to get a quicker startup with-A Database changed mysql> select * from ob_users; +-------+--------+----------+ | SYSID | USERID | PASSWORD | +-------+--------+----------+ | 1 | admin |admin | +-------+--------+----------+ 1 row in set (0.00 sec) mysql> edited /var/www/html/siremis-1.0.1/siremis/metadata/Config.xml <?xml version="1.0"standalone="no"?> <Config> <DataSource> <DatabaseName="Default" Driver="Pdo_Mysql"Server="localhost" DBName="openser" User="admim"Password="admin" /> <DatabaseName="Seremis" Driver="Pdo_Mysql"Server="localhost" DBName="openser" User="admin"Password="admin" /> </DataSource> </Config> Ifsomeone could point out to me where i have gone wrong i would really appreciateit. Regards, Gerry Kernan InfinityIT Suite 17 The Mall, Beacon court, Sandyford, Dublin 18. p:+353-1-2930090 f:+353-1-2930137

1 0

Kamailio Business Directory
by Daniel-Constantin Mierla 28 Oct '10

28 Oct '10

Hello, I am please to announce the launch of Kamailio Business Directory. This is a web page hosted by the project site that will list companies and individuals offering products, services or solutions based on Kamailio or SER. It is an open directory that tries to enable more visibility to the business market around the project and help finding quickly the best service for specific market demands. We invite everyone doing such business to apply to be listed, please follow the instructions at: http://www.kamailio.org/w/business-directory-rules/ We have already several companies listed, couple of them to still in review process, therefore check it again soon, the directory is available at: http://www.kamailio.org/w/business-directory/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

About "reply_code" in 'carrierroute' module
by Iñaki Baz Castillo 28 Oct '10

28 Oct '10

Hi, how is the syntax of the "reply_code" column in 'carrierroute' module? In the doc I some examples: 408 404 50. Obviously "50." means any code starting by "50", but is it a typical regular expression? For example, does it allow something like "500|503" or "50[03]"? Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 4

Asterisk, Kamailio, Openfire and Social Media Integration
by Daniel-Constantin Mierla 28 Oct '10

28 Oct '10

Hello, courtesy of Kelvin Chua, I received the slides he presented yesterday at Astricon 2010 to the session "Asterisk, Kamailio, Openfire and Social Media Integration". More details at: http://www.kamailio.org/w/2010/10/asterisk-kamailio-openfire-and-social-med… Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

PSTN call
by michel freiha 28 Oct '10

28 Oct '10

Dear All, Can someone help me to connect my kamilio in order to make a PSTN call by rewriting host tp PSTN gateway? I changed my config in a manner to do that...The line is ringing but as soon as I open the line on other side the call will hangup...Please find the piece of code # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ rewritehost("XX.XX.XX.XX"); force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } Regards

2 1

SIP application type
by Mino Haluz 28 Oct '10

28 Oct '10

Hi, I'm using radius server to do an accounting, but it does not store an application type (audo/video). Audio/video calls has to be rated with cdrtool, that is why I need this information. It is not even sent through radius protocol, so the radius server cannot get it. My extra-params are: modparam("acc", "radius_extra", "User-Name=$Au; \ Calling-Station-Id=$from; \ Called-Station-Id=$to; \ Sip-Translated-Request-URI=$avp(s:translated_uri); \ Sip-RPid=$avp(s:rpid); \ Source-IP=$si; \ Source-Port=$sp; \ Canonical-URI=$avp(s:can_uri); \ Billing-Party=$avp(s:billing_party); \ Divert-Reason=$avp(s:divert_reason); \ X-RTP-Stat=$hdr(X-RTP-Stat); \ Contact=$hdr(contact); \ Event=$hdr(event); \ SIP-Proxy-IP=$avp(s:sip_proxy_ip); \ ENUM-TLD=$avp(s:enum_tld)") Suggestions? Thanks.

4 4