sr-users November 2010

sr-users@lists.kamailio.org

75 participants
112 discussions

Re: [SR-Users] iphone
by oscar ying 19 Nov '10

19 Nov '10

Daniel, I can log into serweb.iptel.org with my username 'netlink83' and my mobile number is saved in my account. I hope that is enough to have my mobile phone registered. Maybe I should register to iptel again. What numbers can I call via iptel? How do I configure my own pstn gateway? Is there instructions on iptel.org? Thanks, Oscar Ying 650 776 9821 oscar.ying(a)yahoo.com ________________________________ From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: oscar ying <oscar.ying(a)yahoo.com> Cc: kamailio <sr-users(a)lists.sip-router.org> Sent: Thu, November 18, 2010 2:02:18 AM Subject: Re: [SR-Users] iphone On 11/18/10 4:13 AM, oscar ying wrote: >I've tried two different numbers and they both fail. I can make regular >mobile calls but not VoIP calls via my SIP softphone (media5fone). What >do you mean if my phone is registered? > A sip phone register to SIP server with a username and password - that is what I mean with registered. Note that iptel.org does not allow you to call to pstn, unless you have your own pstn gateway configured. You can do voip calls for free. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Trainings Nov 22-25, 2010, Berlin, Germany Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

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Storing multiple SQL results in htable/mtree
by Robert McGilvray 19 Nov '10

19 Nov '10

Hello, I'm interested in using the htable and/or mtree modules to keep my aliases and did lookups in memory to minimize the amount of SQL lookups that are being performed. It's easy enough to use htable to cache a single value after it is retrieved but what I'm not clear on is how I can do it when SQL returns multiple destinations. Consider the following query and returned results: SELECT username,domain,CAST(q_value AS CHAR) FROM aliases where alias_username = '10001' AND alias_domain = 'us.sip.globeop.com' AND active = '1' ORDER by q_value; +----------+--------------------+-----------------------+ | username | domain | CAST(q_value AS CHAR) | +----------+--------------------+-----------------------+ | 8500 | us.sip.globeop.com | 1.000 | | 3874 | us.sip.globeop.com | 1.000 | | 7050 | us.sip.globeop.com | 1.000 | | 2100 | us.sip.globeop.com | 1.000 | In my existing script I just append the results as branches and all works well but I end up with a lot of unnecessary SQL traffic. I see the htable supports array but I'm not sure how to go about assigning multiple values to a single key to use later. How can I take the above results and store them in memory for use by other transactions? Thanks, Bob -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto.

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SIP Scanning Attacks Experiences
by Daniel-Constantin Mierla 19 Nov '10

19 Nov '10

Hello, during the testing period of Kamailio 3.1.0, while running it at voipuser.org, I had the chance to watch live and analyze a SIP scanning attack. Yesterday I noticed another one by looking at Siremis 2.0 charts, therefore I wrote an article with some hints about what you can use to protect your SIP services within Kamailio configuration file. You can read it at: * http://asipto.com/u/i Hope is going to be useful for many of you! Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Trainings Nov 22-25, 2010, Berlin, Germany Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

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iphone
by oscar ying 18 Nov '10

18 Nov '10

Will my iptel SIP work with iPhone 3G and iPhone app Media5fone? Thanks, Oscar Ying 650 776 9821 oscar.ying(a)yahoo.com

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Doubts with Registration
by Danny Dias 17 Nov '10

17 Nov '10

Hello, I'm having problems with my kamailio, my phones are unable to register into the Proxy. I have been checking with ngrep-sip and all i'm receiving is many REGISTER from the phones but never getting some response from the Proxy The Proxy has a public IP and the phones are behind NAT. The weir thing is that my logs are not showing any output...and as you can see on the script, i have a section to receive REGISTER requests.... if (is_method("REGISTER")) { if (!save("location")) { sl_reply_error(); } exit; } Thanks in advance for your help

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Kamailio 1.5.4 - IP addresses WhiteList
by Miguel Baptista 17 Nov '10

17 Nov '10

Hi all, I want to enable pike module based on IP addresses. I mean ... I don't want to check if requests from some specific IP addresses trigged the Pike's threshold. but for all other IP addresses I want to check it. I was thinking about using some kind of IP addresses' whitelist. The userblacklist module has some whitelist but it's based on username/domain not IP addresses. Can someone please give me some advice how to implement it properly. Is there any module that will allow me to do it properly? Best Regards, Miguel Baptista

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Dispatcher is using gateways in Probing mode
by alex pappas 17 Nov '10

17 Nov '10

Dear all, Does anyone can explain how Dispatcher module is sending calls to the gateways in the different states? I mean if I have 3 gateways and the one of them is in Probing mode why Dispatcher is continuing sending there calls? Why this gateway is not changing from Probing to Inactive since is not replaying to OPTIONS messages send by Kamailio? Configuration: modparam("dispatcher", "flags", 2) modparam("dispatcher", "ds_ping_interval", 5) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshhold", 2) modparam("dispatcher", "dst_avp", "$avp(s:dst_avp)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(i:202)") modparam("dispatcher", "ds_append_branch", 0) using db for gw: setid 123,destination sip:xxx.xxxx.xxx, flag 0 ... in ROUTE: if(ds_select_domain("$avp(s:disp_dstgrp)", "4")) in Failure Route if(!ds_next_domain()) I want Dispatcher when it sees a node gateway that is dead , to Not send any other calls. Thank you in advance Alex

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Creating multiple Accounts on kamailio with script.
by Siddhardha Garige 16 Nov '10

16 Nov '10

Hello all, I am conducting some load testing and need to create 50,000 accounts on kamailio. I want to write a shell script to add 50,000 users. Kamctl add [user] [password] prompts for a password and I checked kamctl -help and there is no option for providing password along with the command. Is is possible to disable password and run my script? Thanks Sid "May the light be with you." ______________________________________________ Siddhardha Garige www.luminepixels.com

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Regarding SER Listening on Custom Port
by Saurabh Srivastava 16 Nov '10

16 Nov '10

Hello SER Team, We have made SER listen on two ports via adding this command (listen=eth0:5060 eth0:5062) in ser.cfg file. Now we are able to get the server listening on two ports as 5060 & 5062 and it is sending 200 OK for the user agents which try to communicate on both the ports. SER is able to register two user agents, one on 5060 and one on 5062. But when we are trying to make a communication between the two registered user agents (i.e. one on 5060 and other on 5062) then no communication is possible (i.e. we cannot make calls and chats). Though when the two user agents are registered on same port (i.e. both UAs on 5060 or both UAs on 5062) then communication is possible between the two clients (i.e. we can make calls and chats). Can you people please guide me through that how can I make the 2 UAs registered to SER on different ports (i.e. one on 5060 and other on 5062) communicate with each other (i.e. they can make calls and chats). I have attached a log as well which shows the same problem. This is a wireshark captured output on the machine which has SER installed. Thanks and Best Regards Saurabh Srivastava

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Handle '486 busy here' from upstream carrier
by JR Richardson 15 Nov '10

15 Nov '10

Hi All, Asterisk><kamailio 3.0.2><sip carrier I see normal tansactions until the carrier send back a '486 Busy Here' then kamailio sends an 'INVITE SDP', then I get a '100 trying' from the carrier then a '482 Loop Detected" from the carrier. Shouldn't kamailio relay the 486 to the original invite server and not send SDP? I'm guessing I'm not handling 486 propperly in my config: http://pastebin.com/crfMe81D Here is a pastebin of the call graph: http://pastebin.com/rnQZDyFU I was thinking about including this in my failure route: if (t_check_status("486")) { append_branch(); t_relay(); } Would that do any good? Thanks. JR -- JR Richardson Engineering for the Masses

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sr-users November 2010