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37542 discussions

Dispatcher algorithm 9 ignores weight=xx attribute
by Alejandro Recarey 05 Sep '14

05 Sep '14

Hi, I have a kamailio with the dispatcher module setup. I use: ds_select_dst("1", "9"); to load balance between 3 gateways. My database looks like this: id setid destination flags priority attrs 5 1 sip:1.1.1.1 0 0 weight=10 8 1 sip:1.1.1.2 0 0 weight=45 9 1 sip:1.1.1.3 0 0 weight=45 As I understand it, most of the calls should be split evenly between gateways 8 and 9, and some calls (around 10%) going to gateway 5. Well, gateway 8 gets 99% of the calls. I am using kamailio 4.1 and I really cannot understand why this is happening. Running ds_list shows me that all gateays and weights are correctly parsed into kamailio. This is the result of my ds_list: SET:: 1 URI:: sip:1.1.1.1 flags=AP priority=0 attrs=weight=10 URI:: sip:1.1.1.2 flags=AP priority=0 attrs=weight=45 URI:: sip:1.1.1.3 flags=AP priority=0 attrs=weight=45 I have tried changing the priority to 100, or to 1 in every gateway, and the result is still the same. I have tried adding ; after the weight parameter, the colon is gone in the ds_list once kamailio parses it. If, keeping the same database, I use ds_select_dst("1", "0") the calls are split evenly across all 3 gateways. This is great, but not what I want, but it at least proves that something is working. Is there anything else I can try? Thanks a lot for the help! Alex

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OT: RTCP logging
by Daniel Tryba 04 Sep '14

04 Sep '14

Slightly offtopic, but for QA reasons I'm interested in logging RTCP reports from peers that send them. It looks like rtpproxy-ng might be used to do this (I'm already using rtpproxy anyway). Is someone already doing this and willing to share some pointers?

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Looking for some help
by Sharan Harkisoon 04 Sep '14

04 Sep '14

I am in need of a Kamailio expert that has some availability for consulting services (remote is fine). Feel free to contact me for details. Thanks, Sharan Harkisoon

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Redirect server vs CANCEL
by Mihaly Zachar 04 Sep '14

04 Sep '14

Hi Gents, We have created a small redirect server using the Lua API to get the destinations. Lets say that it takes some time while the script generates the answer, during this time we got a CANCEL. How we can handle this properly ? As far as I could see the TM module always sends back the answer (302) and after that I get the "CANCEL" in my script. Here is my config: http://pastebin.com/7PyKBJyD Here is my test Lua script: http://pastebin.com/6VXn5y8m Could you please help me what is the best way to handle the CANCEL ? Is it possible to do it properly ? Thanks, Misi

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nat_traversal vs nathelper for mediaproxy
by Satish Patel 04 Sep '14

04 Sep '14

I am planing to implement mediaproxy so which NAT module will be good for media proxy and why? should i use nat_traversal ro nathelper ? I haven't see any example people using nat_traversa. most of example i found on google are based on nathelper so just want to make sure which one i should use and why?

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Shared memory
by Morten Tryfoss 04 Sep '14

04 Sep '14

Hi, I wonder about the difference between “used” and “real used” shared memory in Kamailio. Best regards, Morten Tryfoss

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How to determine if a 4xx message came from dispatcher or a client?
by Olli Heiskanen 04 Sep '14

04 Sep '14

Hello, A question on Kamailio variables and using dispatcher: When in failure_route I want to know if the request message was going to a dispatcher ip or a sip client ip (as in any other than dispatcher ip), how do I make an if statement for that? If I use ds_is_from_list(), I get wrong results because in both cases the request was sent by Kamailio. Also, I don't seem to be able to get the source ip of the 4xx message (src_ip shows the source ip of the request message). cheers, Olli

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Can log in from SIP
by Alexis Walhovd 04 Sep '14

04 Sep '14

Hi I just created an account with iptel.org. But I can not log in to my account through my SIP software from OSX. Please see below for details: I type “Alexis82” as password. All the best, Alexis

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need help - Insert_hf when Route: missing
by Satish Patel 04 Sep '14

04 Sep '14

I have post many question on topology hiding any not get any reply back from people and developers now I don't have any option except some goofy hack When I use topology hiding it removes Route: and because of that callee doesn't able to send BYE back to opensips. I want use insert_hf to inject Route: so I get clean BYE message. Do you guys have any suggestion? Sent from my iPhone

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Asterisk cluster behind kamailio natted to pubic IP, presenting internal ip addresses in From tag
by Tim Chubb 04 Sep '14

04 Sep '14

Hi All I'm scratching my head on this one, I have set everything up in the following configuration: Public ip natted to DMZ via Cisco ASA 5510 <= DMZ => Kamailio & RTP Proxy <= Voice DMZ => Asterisk My kamailio configuration is based heavily on the one found here: http://saevolgo.blogspot.co.uk/2013/08/rtpproxy-revisited-kamailio-40.html & http://saevolgo.blogspot.co.uk/2011/11/how-to-increasing-voip-services.html So far everything is working fine, I can register via the public IP address, IM & presence is working, and as does audio, however when I dial an extension the caller id comes up like this 12345@172.16.15.123:5080<mailto:12345@172.16.15.123:5080> which is the ip addresses of the asterisk server that the dispatcher has assigned to the call, I have tried setting the P-Asserted-Identity, P-Preferred-Identity & Remote-Party-ID headers to no effect. What I would like to achieve is that the public IP or domain name comes up when I call an extension. From examining the sip traffic traversing the kamailio box, it seems that the From and Contact headers sent by asterisk are the source of the internal implementation information reaching the end-user. How can I prevent implementation information like this leaking out? I consider this to be a security risk as I don't want anyone to be able to enumerate the number of servers that comprise a given cluster as easily as placing a few calls, ideally I only want data relating to the public side of the network getting out. But I would be happy if it takes digging into the sip traffic to glean any details initially. I'm getting the feeling that this could well be an asterisk configuration issue, so apologies if this is a little off topic, however I have been unable to find a similar configuration to try and achieve my desired results. Tim.

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