sr-users January 2009

sr-users@lists.kamailio.org

85 participants
110 discussions

[Kamailio-Users] Internet drops and ATA don´t register

by MagoDaViola

Untitled Document Hello all .... I'm having a problem with our server or customers. Most of our clients access the server using the combination ATA and ADSL Modem, but when the ADSL drops and returns in a few minutes, the ATA can no longer make the register on SIP server. Sometimes passing through the modem or the requests of the ATA. I identified that this happens a lot with the D-Link modems DSL-500B and siemens SpeedStream 4200. If someone went through this I would your help ... thanks -------------------------------------------------------------------------- Alcindo Schleder Sistema Processa Acessoria e Comunicações Ltda. a.. MSN: a_schleder(a)hotmail.com b.. Skipe: alcindo_schleder Demétrio P. dos Santos, 705 - 95670-000 - Gramado-RS-Brasil Celular + 55 54 9966 7591 alcindo(a)processa.org Fone +55 54 3286-1738 comercial(a)processa.org

15 years, 3 months

[Kamailio-Users] Perl var main script access

by Luis Guaman

Hi , How can I get access in the onreply_route main script section for a variable set in a perl which is called from a previous route[] section?? Thanks in advance Luis Guaman Interlancompu

15 years, 3 months

[Kamailio-Users] I need termine a call then timeout.

by igor

I need termine a call then timeout. but, return a message WARNING and do not Hangup! help me! WARNING:dialog:get_expired_dlgs: start with tl=0xb5a11a20 tl->prev=0xb59cf550 tl->next=0xb59cf550 (101) at 101 and end with end=0xb59cf550 end->prev=0xb5a11a20 end->next=0xb5a11a20 WARNING:dialog:get_expired_dlgs: getting tl=0xb5a11a20 tl->prev=0xb59cf550 tl->next=0xb59cf550 with 101 WARNING:dialog:get_expired_dlgs: end with tl=0xb59cf550 tl->prev=0xb5a11a20 tl->next=0xb5a11a20 and d_timer->first.next->prev=(nil) WARNING:dialog:dlg_ontimeout: timeout for dlg with CallID 'T8x-xgYUoIe3707wDPg1qwq-AX4o3m' and tags '9697t1cm3thc6l2207g9' '7jju88u987' loadmodule "dialog.so" modparam("dialog", "enable_stats", 1) modparam("dialog", "db_url", "mysql://user:pass@localhost/openser") modparam("dialog", "db_mode", 1) modparam("dialog", "dlg_flag", 4) modparam("dialog", "timeout_avp", "$avp(s:dlgtimeout)") modparam("dialog", "dlg_extra_hdrs", "Hint: credit expired\r\n") modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "default_timeout", 10) if(is_method("INVITE")) { $avp(s:dlgtimeout)=5; $avp(s:ringtimeout)=10; setflag(4); route(4); } Atenciosamente, Igor Marques celular: 55 21 7858 7499 nextel: 87*24074 e-mail: igor(a)carneiro.cc web: www.carneiro.cc -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

15 years, 3 months

[Kamailio-Users] Accounting: How to avoid a fraudulent BYE with lower CSeq?

by Iñaki Baz Castillo

Hi, first of all I'm sorry for this cross-posting, but I think it could be interesting for any SIP proxy with accounting capabilities. I'm thinking in the following flow in which the caller/attacker would get an unlimited call (but a limited CDR duration): -------------------------------------------------------------------------- attacker Kamailio (Acc) gateway INVITE (CSeq 12) ------> <-------- 407 Proxy Auth INVITE (CSeq 13) ------> INVITE (CSeq 13) ------> <------------------- 200 Ok <------------------- 200 Ok << Acc START >> ACK (CSeq 13) -----------> ACK (CSeq 13) -----------> <******************* RTP ************************> # Fraudulent BYE !!! BYE (CSeq 10) -----------> << Acc STOP >> BYE (CSeq 10) -----------> <-- 500 Req Out of Order <-- 500 Req Out of Order -------------------------------------------------------------------------- The call hasn't finished, but Kamailio has ended the accounting for this call since it received a BYE. And this BYE will generate a correct ACC Stop action (since it matches From_tag, To_tag and Call-ID). I think this is *VERY* dangerous and I hope I'm wrong. Would help the dialog module here? does the dialog module check the CSeq of the BYE in some way and could it prevent Kamailio from generating the ACC STOP action? (I don't think so). I've also asked in SIP-implementors and an idea could be generating the ACC STOP action when receiving the 200 OK for the BYE (and not when receiving the BYE itself). Of course this will be valid when the gateway is the recipient of the BYE (and we know the gateway is not an "attacker"), but this is not valid when the recipient of the BYE is an user since it could send no reply for the BYE. The only solution I see is: - Using the dialog module, Kamailio should check the CSeq value in the BYE. 1) Kamailio should forward the BYE just in case the CSeq is higher than the actual CSeq for this dialog direction. 2) Kamailio should generate the ACC STOP action just in case the CSeq is higher than the actual CSeq for this dialog direction. Both 1) and 2) are needed since a gateway could accept a BYE with wrong CSeq. In this case the call is ended but the accounting STOP action doesn't exist (infinite call). But I think this is too complex, isn't it? -- Iñaki Baz Castillo <ibc(a)aliax.net>

15 years, 3 months

[Kamailio-Users] (no subject)

by igor

WARNING:dialog:get_expired_dlgs: start with tl=0xb5a11a20 tl->prev=0xb59cf550 tl->next=0xb59cf550 (101) at 101 and end with end=0xb59cf550 end->prev=0xb5a11a20 end->next=0xb5a11a20 Jan 8 11:52:22 voip01 /usr/sbin/kamailio[12596]: WARNING:dialog:get_expired_dlgs: getting tl=0xb5a11a20 tl->prev=0xb59cf550 tl->next=0xb59cf550 with 101 Jan 8 11:52:22 voip01 /usr/sbin/kamailio[12596]: WARNING:dialog:get_expired_dlgs: end with tl=0xb59cf550 tl->prev=0xb5a11a20 tl->next=0xb5a11a20 and d_timer->first.next->prev=(nil) Jan 8 11:52:22 voip01 /usr/sbin/kamailio[12596]: WARNING:dialog:dlg_ontimeout: timeout for dlg with CallID 'T8x-xgYUoIe3707wDPg1qwq-AX4o3m' and tags '9697t1cm3thc6l2207g9' '7jju88u987' Atenciosamente, Igor Marques celular: 55 21 7858 7499 nextel: 87*24074 e-mail: igor(a)carneiro.cc web: www.carneiro.cc -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

15 years, 3 months

[Serusers] SER server reload

by Leon Li

Dear all, I am new to SER and on a project to reload company's SER server for some SIP testing. However, I couldn't start SER after compiling. Every time I run "/usr/local/sbin/ser -f /PATH/ser.cfg", I got "ERROR: bad config file (1 errors)". Does this mean my ser.cfg has something wrong or where can I trace the logs for errors? Also, how do I uninstall the old SER and run a clean installation? Any suggestion would be appreciated. Best regards, Leon Leon Li | IT Engineer | Application & Services | AARNet Pty Ltd ________________________________ street address: Level 2, Binary Centre, Bldg1, 3 Richardson Place, North Ryde, NSW 2113 t. +61 2 9779 6937 m. 0403 603 473 e. leon.li(a)aarnet.edu.au w. www.aarnet.edu.au <http://www.aarnet.edu.au/> ________________________________ important This email and any files transmitted with it are confidential, and the rights of confidentiality in such information are not waived or lost by its mistaken delivery to you. Any dissemination, copying, use or disclosure of the email and/or such files without the permission of AARNet, or the sender, is strictly prohibited. If you have received this email in error, please contact the sender immediately and delete all copies of this transmission.

15 years, 3 months

[Kamailio-Users] prepaid system

by igor

I need implement a prepaid system in my kamilio ! anybody help me !? any information ! Atenciosamente, Igor Marques celular: 55 21 7858 7499 nextel: 87*24074 e-mail: igor(a)carneiro.cc web: www.carneiro.cc -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

15 years, 3 months

Re: [Kamailio-Users] how can I add .sar file as application to Wesip application server

by skumar453＠gmail.com

Hi thanku for reply while i am running the click2dial application i am getting the message like bellow ConvergedHostDeployer [main] - Installing web application at context path /click2call-08_11_06 from URL file:/usr/wesip/wesipapps/click2call-08_11_06 the log file geting while running the application is StandardAppSessionManager [main] - Force random number initialization completed processJars: cannot find /WEB-INF/lib ConvergedContext [main] - ConvergedContext [Welcome] has been started StandardService [main] - > Default Wesip Application ==> http://localhost:5075/ > Manager Wesip Application ==> http://localhost:5075/manager/html? StandardService [main] - > In order to access the manager application use default credentials: - username: wesip - password: wesip > This configuration is defined in wesip-users.xml inside conf folder: - file: ./conf/wesip-users.xml here i am not clear how to test this application ..can u please suggest me about this On Jan 7, 2009 8:27pm, Ginés Gómez <gines(a)voztele.com> wrote: > Hi, > > > > > > you can simply copy the .sar file at the wesipapps directory > > > > > > Regards > > > > > > Gines > > > > > > > > hi all > > > > > > I am workin on the "WeSIP" application server. I installed opensips as proxy with SEAS module and wesip as Application server. > > > these two are successfully connected ...but here i have no clue on how to add application "click2dial" to WeSIP. > > > > > > 1. please let me know how can I add .sar file as application. > > > 2. http://localhost:5075/manager/html? is showig a blank page. how can I get this page displayed? > > > > > > regards > > > suresh_______________________________________________ > > > Users mailing list > > > Users(a)lists.kamailio.org > > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > > > > > >

15 years, 3 months

[Kamailio-Users] carrierroute prioritize gateways

by Juan Asencio

Hi list, I'm reading some old post in order to implement carrierroute. I found this in a post that I include at the end of this message, but the part that I want to ask about is this one: "You can put all the carriers under the same table and prioritize the gateway based on domains. Like tis you don't need to hardcode anything in the config file" If I do this, do I need to prioritize the gateways on the subscriber table and if is there, how do I do it for the domain, when it is only a cr_preferred_carrier field? Cheers, Juan.- [OpenSER-Users] Carrierroute *Ovidiu Sas* osas at voipembedded.com <mailto:users%40lists.kamailio.org?Subject=%5BOpenSER-Users%5D%20Carrierrout…> /Wed Mar 26 04:57:54 CET 2008/ * Previous message: [OpenSER-Users] Carrierroute <http://lists.opensips.org/pipermail/users/2008-March/016657.html> * Next message: [OpenSER-Users] [FEATURE REQ] allow does_uri_exist() to test against other variables besides $ru <http://lists.opensips.org/pipermail/users/2008-March/016658.html> * *Messages sorted by:* [ date ] <http://lists.opensips.org/pipermail/users/2008-March/date.html#16659> [ thread ] <http://lists.opensips.org/pipermail/users/2008-March/thread.html#16659> [ subject ] <http://lists.opensips.org/pipermail/users/2008-March/subject.html#16659> [ author ] <http://lists.opensips.org/pipermail/users/2008-March/author.html#16659> ------------------------------------------------------------------------ Hello Douglas, I am using this module in production with more then 400k routes loaded for 8 different providers and it is working like a charm. As Henning mentioned, the interface is a little bit cumbersome but usable. You can put all the carriers under the same table and prioritize the gateway based on domains. Like tis you don't need to hardcode anything in the config file. Carrierroute was designed to deal with a big amount of routing rules and it is doing an amazing job. LCR was designed to deal with a limited number of routes/gateways but it has a different level of flexibility and a more mature interface. Carrierroute is a brand new module and for sure it will improve in flexibility as times go by (just like the lcr did along releases). Hope this helps, Ovidiu Sas

15 years, 3 months

[Kamailio-Users] dialplan and carrierroute

by victorpy

hello, sorry for the other message. i want to know if there is a way of routing based on patern matching using caller-id. Basically, i want to route through a carrier if the call-id have some pattern Eg.: international is 002-(country code)-(area code)-697699. Long distance is (country code) -(area code) -123456 Local is 123456 The international and long distance change If international then route through carrier1 else if long distance the route through carrier 2 else if local then route through carrier 3 -- View this message in context: http://www.nabble.com/dialplan-and-carrierroute-tp21331266p21331266.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

15 years, 4 months

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sr-users January 2009