Re: [SR-Users] How to disable retransmits via TCP connection? etc.
Hi, Don't want to sound like a captain obvious, but what I can say after a quick look at your traces is that both client and server are behind NAT and either some entity drops big packets silently or somebody drops ICMP (fragmentation needed) which doesn't allow TCP to adjust max MSS. The other strange thing is that you said that you set server's MTU to 800 bytes, but server has sent a huge TCP segment with a total length of 1548 bytes and Don't fragment bit set. On 24/06/15 17:30, Andrey Utkin wrote: Thanks for all the suggestions. Have taken pcap dumps with commands tcpdump -i any -w /tmp/sip_tcp.pcap host counterpart host http://whdd.org/sip_tcp_server.pcap http://whdd.org/sip_tcp_client.pcap The only obviously bad thing I see is [TCP Previous segment not captured] mark on the packet client side dump which is the ending of INVITE request body. Googling on this message, I've found some advice at StackOverflow to tune TCP socket TCP_MAXSEG option. http://whdd.org/sip_tcp_mtu800_server.pcap http://whdd.org/sip_tcp_mtu800_client.pcap For this test I have set MTU to 800 on the server's only actual net interface, and restarted Kamailio. The situation stays the same, despite the packets show differently in wireshark. On client side, there's still [TCP Previous segment not captured], and on server side there's a mark [TCP segment of a reassembled PDU]. Any review is greatly appreciated! ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] How to disable retransmits via TCP connection? etc.
Hi, But the fact that packet sent from a server to a client looks differently captured from both points says that somebody tries to help you (read SIP ALG). So I would join to Daniel's suggestion to try with TLS. On 24/06/15 17:30, Andrey Utkin wrote: Thanks for all the suggestions. Have taken pcap dumps with commands tcpdump -i any -w /tmp/sip_tcp.pcap host counterpart host http://whdd.org/sip_tcp_server.pcap http://whdd.org/sip_tcp_client.pcap The only obviously bad thing I see is [TCP Previous segment not captured] mark on the packet client side dump which is the ending of INVITE request body. Googling on this message, I've found some advice at StackOverflow to tune TCP socket TCP_MAXSEG option. http://whdd.org/sip_tcp_mtu800_server.pcap http://whdd.org/sip_tcp_mtu800_client.pcap For this test I have set MTU to 800 on the server's only actual net interface, and restarted Kamailio. The situation stays the same, despite the packets show differently in wireshark. On client side, there's still [TCP Previous segment not captured], and on server side there's a mark [TCP segment of a reassembled PDU]. Any review is greatly appreciated! ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] How to disable retransmits via TCP connection? etc.
Hi, I would also add that if you see partial packets you can try to remove any transport protocol (udp/tcp) and port filters. It will help if you are dealing with IP fragmentation. Otherwise sniffer won't catch IP fragments since they don't have transport level headers. Best Regards, Vitaliy ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Recommended configuration for TCP support
What would happen in this case? If I understand correctly, the proxy will always forward the request but the last hop (before the UAS) will forward the request only if a connection is active? correct. If the last hop doesn't forward the request because no TCP connection exists, what happens? How the proxy will detect this situation? It won't find any matching connection and won't try to establish a new one since you asked it to not. So it will just send a reply back to the registrar with an error. After receiving an error registrar in its turn can perform some smart actions like forwarding to a voicemail system. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Recommended configuration for TCP support
*set_forward_no_connect()* called after lookup() prevents kamailio from making a new connection to a phone if there is no active connection. This is useful when you are trying to call to a UA which kamailio still keeps registration record for, but TCP/TLS connect to this phone is already gone (due to timeout, network problem, etc...). On 05/05/15 14:54, Mickael Marrache wrote: In general, it happens in our network (as more and more Via headers are added). Sorry, I didn't understand your last sentence. What do you mean by not connect if there is no active connection after location lookup? *From:*sr-users [mailto:sr-users-boun...@lists.sip-router.org] *On Behalf Of *Daniel-Constantin Mierla *Sent:* Monday, May 04, 2015 11:07 PM *To:* Kamailio (SER) - Users Mailing List *Subject:* Re: [SR-Users] Recommended configuration for TCP support Hello, where are typically those issues with truncated SDP? On the phones? If you use tcp and have phones behind the nat, it is recommended to not connect if there is no active connection after location lookup -- see core functions for tcp. Cheers, Daniel On 04/05/15 13:54, Mickael Marrache wrote: Hi, We sometimes get issues caused by truncated SDP bodies. Therefore, we are looking at using TCP instead of UDP. What is the recommended configuration when TCP is used? Currently, I only have the following: children=16 listen=udp:ip:5060 listen=tcp:ip:5060 disable_tcp=no tcp_connection_lifetime=605 (registration expiry is 600) Also, is there a difference with UDP regarding NAT handling? Thanks, Mickael ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda http://twitter.com/#%21/miconda -http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany -http://www.kamailioworld.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] LCR with gateway capabilities
On 29/04/15 12:34, Grant Bagdasarian wrote: That sounds possible! Thanks. That does this description mean: “Execution time of load_gws() function is O(N) * O(M), where N is number of different prefix lengths and M is number of collisions for matching prefix(es) in lcr rules hash table of the LCR instance.”? Does this mean it loads the data from its internal memory or does it load the data from the database again? load_gws() prepares an ordered list of gateways for current destination. It loads data only from internal in-memory gateways db (hash table). How fast does it select a gateway after load_gws using next_gw()? Is it O(N) again? O(1) since load_gws already prepared ordered (by price) list and next_gw() only takes next gw in a list. What about multiple combinations of capabilities? For example caller id spoofing and g729 codec, or caller id spoofing and g711ulaw codec? Etc. Wouldn’t this cause the data to grow exponentially? It would. Another idea: You can also have only one lcr instance and keep gateways capabilities in the tag field (serialized somehow). Then you can just iterate through a list generated by load_gws() skipping gateways which don't match your criteria and in the end a call will be routed to the cheapest gateway with proper capabilities. *From:*sr-users [mailto:sr-users-boun...@lists.sip-router.org] *On Behalf Of *Vitaliy Aleksandrov *Sent:* Wednesday, April 29, 2015 10:36 AM *To:* sr-users@lists.sip-router.org *Subject:* Re: [SR-Users] LCR with gateway capabilities What about configuring two LCR instances with different lcr_id. The first one can use only gateways with requested capabilities and the second one all gateways. Then you can make a decision about which instance to use during call routing process providing this lcr_id to load_gws() function. Hello, From what I understand about the LCR module is that the rules have to be prioritized by the admin, be it manually or automatically by an application. Let’s say the LCR database has 10 gateways, each with their own rules etc. 4 of these gateways support caller id spoofing and the others don’t. But the other 6 are cheaper and Kamailio has to route a call using a gateway which support caller id spoofing and is the cheapest of the 4. Would it be possible to tell the LCR module to select a gateway based on certain capabilities and is the cheapest of the ones which support a certain capability? ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@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@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] LCR with gateway capabilities
What about configuring two LCR instances with different lcr_id. The first one can use only gateways with requested capabilities and the second one all gateways. Then you can make a decision about which instance to use during call routing process providing this lcr_id to load_gws() function. Hello, From what I understand about the LCR module is that the rules have to be prioritized by the admin, be it manually or automatically by an application. Let’s say the LCR database has 10 gateways, each with their own rules etc. 4 of these gateways support caller id spoofing and the others don’t. But the other 6 are cheaper and Kamailio has to route a call using a gateway which support caller id spoofing and is the cheapest of the 4. Would it be possible to tell the LCR module to select a gateway based on certain capabilities and is the cheapest of the ones which support a certain capability? ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] LCR with gateway capabilities
The Tag column size is 64. If I make this larger in the database, will it be truncated once it is loaded into memory? According to modules source code Tag's max size is hardcoded and will be truncated. But this is not a bit problem. You can keep capabilities list in htable and only put a key for this htable info to the tag. Through which list do I need to iterate? You mentioned the data is stored in a hash table, what is the name of this hash table? load_gws() creates a list of gateways and next_gw() fetches them one by one. After calling load_gws() you can just call next_gw() which puts gateways tag to the tag_avp. If current gateway doesn't have requested capabilities just call next_gw() again. Basically for each call I need to call the load_gws() function, which is O(N) * O(M). What if I take a different approach and call a stored procedure (for each call) which does the selection of a gateway based on the supplied criteria. Of course the tables would be optimized with indices for the best possible performance. When comparing load_gws() (O(N) * O(M)) with the stored procedure approach, would there be a huge performance loss when taking the stored procedure approach? Who knows.. You can test it and share results with community :) I know this approach brings a whole new set of problem with availability and the loss of features, but for now I’m only interested in the performance aspect of both methods. Actually going this way you don't need LCR module at all. You can call stored procedure, lua/python/perl/... script, or request an external system using radius/xmlrpc. But be careful making blocking operations from a script or have enough number of kamailio processes. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Expect the kamailio's sip account is unregistered state when the client app is shutdown
Hi, You can take a look at this option: http://kamailio.org/docs/modules/4.2.x/modules/usrloc.html#usrloc.p.handle_lost_tcp It's a bit limited and doesn't work for: - usrloc db only mode - if you have separated edge proxy and registrar Thank you for information.. What can Kamailio do about clients, that exit abnormally (e.g. crash, turn off, ..) which do unregister properly? Can Kamailio send this expire=0 on behalf of these clients? Can Kamailio determine that a client got offline based on e.g. TLS connection or other heartbeats? On 20.04.2015 10:16, Daniel Grotti wrote: Hi, here you are: https://tools.ietf.org/html/rfc3665#section-2.4 -- Daniel Grotti VoIP Engineer Sipwise GmbH Europaring F15 | 2345 Brunn am Gebirge, Austria | www.sipwise.com On 04/20/2015 10:10 AM, Mickael Marrache wrote: Hi, You need to configure your client to un register when the app is closed. For that, your client must send a REGISTER with Expires: 0. Mickael *From:*sr-users [mailto:sr-users-boun...@lists.sip-router.org] *On Behalf Of *xuefeng zhang *Sent:* Monday, April 20, 2015 10:58 AM *To:* Daniel-Constantin Mierla; Kamailio (SER) - Users Mailing List *Subject:* [SR-Users] Expect the kamailio's sip account is unregistered state when the client app is shutdown Hi Daniel, When the client's app is closed,the sip account may be registered state,because the expired time may have some rest time.I want to implement when the client's app is closed,the account also unregister. How can I implement this function?Would you give me some advices? Thanks! Xuefeng Zhang ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@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@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@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Routing PSTN calls from Asterisk through Kamailio to UAC loose_route
According to your description BYE was sent using the information from R-URI which had no 5080 port. Asterisk should have added port 5080 to the outgoing Invite contact so that it could be used for in-dialog routing. Can you show a full trace with sip traffic between kamailio and asterisk. To catch sip traffic on all interfaces use -i any option for tcpdump or -d any for ngrep. I've been working on integration of Asterisk and Kamailio, currently on the same host with different ports, and have come across a problem with calls that originate from the Asterisk side (PSTN/DAHDI) and route through Kamailio to a SIP UAC. In short, when the SIP UAC (10.1.1.9) sends the BYE, loose_route() is returning -1 and the BYE is routed back to Kamailio (10.1.1.1:5060) instead of Asterisk (10.1.1.1:5080). I am using the stock WITHINDLG route configuration. RR module settings are as follows: modparam(rr, enable_full_lr, 1) modparam(rr, append_fromtag, 1) The BYE from the SIP UAC contains the following Route header which only contains the contents of Kamailio's Record-Route header. I have attached the full sip trace for review as well. Route: sip:10.1.1.1;lr=on;ftag=2400d939-de0b-4456-9e01-f9a3302f3e25;nat=yes. What would be the best method to resolve this issue in either Asterisk or Kamailio? Should I manually add a Record-Route header for the Asterisk host:port to Kamailio config? Is there something to be done in Asterisk? Thanks. -A ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] missing BYE when 2 redundant kamailio servers share the same database
We use TLS for SIP. The Wireshark pcap would be encrypted. I’ll try to get a pcap anyway. Wonder if there is a way to dump pcap from inside kamailio. Wireshark can decrypt SIP signalling sent over TLS connections if you provide server's private key to it. All the requests within dialog are routed through 2 kamailio instances. We want to make sure each phone only sends requests through its registrar. I have included pjsua logs in subsequent emails in this thread. Those logs have SIP messages, but only provide client perspective. Thanks for the help, Ding On Mar 20, 2015, at 3:00 AM, Daniel-Constantin Mierla mico...@gmail.com mailto:mico...@gmail.com wrote: Hello, On 19/03/15 02:54, Ding Ma wrote: [...] My first question is why k1 loose_route sends the BYE to itself instead of the client. Is this a bug? can you get the pcap of such call? We have to see the routing headers to say what is next hop address. Are all the requests within dialog routed via same instance of kamailio? My next question is whether the above location routing for BYE from peer kamailio a good/safe approach. The SIP traces will be sent later to avoid exceeding email size limit. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda -http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany -http://www.kamailioworld.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] tcp_max_connection to 4096
You shouldn't feel any performance issues after increasing tcp_max_conn to 4096. Connections hash table size is pretty high (1024) so it's not a problem at all. Of course if you want to handle twice bigger number of simultaneous clients you need to check if you current hardware can handle it (RAM, CPU). Hi Team, We are seeing some errors in our kamilio for TCP max conn (ERROR) : 2048 (the default). We are thinking to double the TCP connection for our kamailio registrar server. tcp_max_connections=4096 Is there any performance issue if we double the tcp_max_connections ?. Currently we are setting these parameters for TCP. tcp_connection_lifetime=3605 tcp_accept_no_cl=yes tcp_rd_buf_size=16384 Do we need to tune any other variable if we are setting max tcp connections to 4096 for better performance ? Thanks for looking into this. Regards Varghese Paul ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] async workers
Hi, What is the practical limit to the number of async worker processes? With SIP child processes, it seems to be about the number of available CPUs in /proc/cpuinfo. After that--at least, per my testing--one begins to hit the point of diminishing returns, presumably due to SHM IPC and synchronisation issues. Is the restriction similar in the async execution context? no specific restriction. Also, I haven't seen any degradation when using more sip worker processes that cpus, which I do have always (at least 2 per CPU), because a worker can do quite a lot of I/O. I have seen such degradation, at least in my VM testing environment. I find that I get the highest CPS (e.g. with sipp) with the smallest number of workers. Actually, I get as good throughput with 4 workers (in an 8 CPU scenario on a quad-core processor) as I do with 8! Increasing beyond 8 leads to diminishing returns. With real traffic some of workers can be busy on blocking operations such as database queries or host name resolution. So I prefer to add more workers to not get in situation when all my workers are blocked by something. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] How to uniquely identify SIP WS / WSS endpoint
On 22.08.14 03:26, Muhammad Shahzad wrote: Sorry for putting this question on both dev and user mailing lists, as it is a rather theoretical question and i hope some SIP guru on either mail list will answer. For non-WS endpoints which use TCP or UDP for SIP transport, each upstream request has top most VIA header pointing to the previous hop which forwarded the request to current hop while each downstream request has top most VIA header pointing to next hop to which it will be forwarded from current hop. But for WS endpoints, the top most VIA has dummy static value, so there is no way to identify who sent this request or to whom the reply is going to. Please note that i am not specifically interested in network address of remote endpoint (though VIA header is suppose to provide it), i only need to match requests and responses from / to a specific device using SIP v2.0 standard. Any help is highly appreciated. Thank you. Can you provide an example of scenario you want to create ? Do you want to understand how transaction and dialog matching works in SIP ? ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] How to uniquely identify SIP WS / WSS endpoint
When kamailio processes a request script writer can check if there any Route header or valid R-URI or R-URI alias parameter to determine the destination. You can check it manually maybe reusing kamailio sip parser. As you've already said to find a destination where kamailio is going to send a reply you can parse via header or its received and rport parameters. Even when via doesn't have valid destination (ws/wss transport) it has correct received and rport parameters which kamailio adds during a request processing. Via header in INVITE received from WSS client and forwarded to a destination looks like this: Via: SIP/2.0/WSS df7jal23ls0d.invalid;received=1.2.3.4;branch=z9hG4bKTp9lzCApgHsdbRUrFcZ4XTCI49EZbbDf;rport=37213 Not really, the main context of this question is in reference to this thread, https://www.mail-archive.com/sr-users@lists.sip-router.org/msg19962.html A patched to allow network IO intercept in kamailio corex module was add to trunk as discussed in this thread, https://www.mail-archive.com/sr-users@lists.sip-router.org/msg20183.html Currently i am able to compress / decompress entire sip message coming from or going to remote endpoint in kamailio server. It works fine. Now i want to try ITV encryption algorithm for this on-wire data. https://github.com/mshary/itv For this i need to keep track of remote endpoint. At this low level, i only have raw data received from or being transmitted to remote UA, without even the remote socket address, so i have no choice but to look at this raw data to determine the identity of remote endpoint. For non-WS transport, i can easily look at topmost VIA and extract network address to use as unique identification of endpoint who sent the data or would receive the data. However, for WS transport this topmost VIA is useless static constant string. So VIA checking is pointless (all remote endpoints will or may have same top most VIA). So i was thinking if there is another way to do it? I thought of using GRUU, but it is not always present, especially in SIP replies. Thank you. On Mon, Aug 25, 2014 at 3:24 PM, Vitaliy Aleksandrov vitalik.v...@gmail.com mailto:vitalik.v...@gmail.com wrote: On 22.08.14 03:26, Muhammad Shahzad wrote: Sorry for putting this question on both dev and user mailing lists, as it is a rather theoretical question and i hope some SIP guru on either mail list will answer. For non-WS endpoints which use TCP or UDP for SIP transport, each upstream request has top most VIA header pointing to the previous hop which forwarded the request to current hop while each downstream request has top most VIA header pointing to next hop to which it will be forwarded from current hop. But for WS endpoints, the top most VIA has dummy static value, so there is no way to identify who sent this request or to whom the reply is going to. Please note that i am not specifically interested in network address of remote endpoint (though VIA header is suppose to provide it), i only need to match requests and responses from / to a specific device using SIP v2.0 standard. Any help is highly appreciated. Thank you. Can you provide an example of scenario you want to create ? Do you want to understand how transaction and dialog matching works in SIP ? ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@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@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] pkg.stats
Hello,
If there are no changes in configuration script logic and in behavior of
main modules I can try it.
Hello,
there were some issues with pkgstats that I worked on in master
branch. Can you give it a try and see if works ok? If all fine, I will
look at backporting, but there were other changes I did in master for
a bit more optimizations for memory manager.
Cheers,
Daniel
On 18/07/14 18:22, Vitaliy Aleksandrov wrote:
Hi list,
I have a problem with pkg.stats for tcp.main process on kamailio
4.1.2 73ea61-dirty.
kamailio has 64M of private memory for each process (-M 64).
pkg.stats for tcp.main process shows that is uses 75M and still has
62M free:
{
entry:36
pid:10032
rank: -4
used: 73109720
free: 65891192
real_used: 79146048
}
root@webRTC:~# ps -p 10032 -o
pid,tid,state,pcpu,pmem,rss,size,sz,vsz,args
PID TID S %CPU RSSSIZESZ VSZ COMMAND
10032 10032 S 0 26420 67508 76821307284
/.../kamailio -f /.../kamailio.cfg -P /.../kamailio.pid -m 128 -M 64
-u kamailio -g kamailio
This server handles 1000+ tcp/tls connections and works just fine for
the last two months. That's why i think this is not a memory leak and
just a pkg.stats problem.
Does anybody else saw this or maybe somebody is already working to
fix it ?
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[SR-Users] pkg.stats
Hi list,
I have a problem with pkg.stats for tcp.main process on kamailio 4.1.2
73ea61-dirty.
kamailio has 64M of private memory for each process (-M 64).
pkg.stats for tcp.main process shows that is uses 75M and still has 62M
free:
{
entry:36
pid:10032
rank: -4
used: 73109720
free: 65891192
real_used: 79146048
}
root@webRTC:~# ps -p 10032 -o pid,tid,state,pcpu,pmem,rss,size,sz,vsz,args
PID TID S %CPU RSSSIZESZ VSZ COMMAND
10032 10032 S 0 26420 67508 76821307284
/.../kamailio -f /.../kamailio.cfg -P /.../kamailio.pid -m 128 -M 64 -u
kamailio -g kamailio
This server handles 1000+ tcp/tls connections and works just fine for
the last two months. That's why i think this is not a memory leak and
just a pkg.stats problem.
Does anybody else saw this or maybe somebody is already working to fix it ?
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[SR-Users] How to remove attribute from SDP ?
Hello list, Can anybody suggest the way to remove a line/attribute from SDP that matches some pattern (regexp), some kind of grep -v for SDP ? I'm trying to remove unnecessary ICE candidates from the SDP body and only thing I know is the IP address of the candidate I need to remove. I was trying to use avp_subst() + set_body() or subst_body() but didn't get valid SDP. kamailio-3.2.4. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] How to remove attribute from SDP ?
I absolutely forgot about the possibility of embedding scripting languages to configuration. Lua did the trick. Thanks for the right direction and example. I can share my solution, using app_python. I'm using this in production environment during 2 years. loadmodule app_python.so modparam(app_python, script_name, /opt/kamailio/python/sdp.py) modparam(app_python, mod_init_function, mod_init) modparam(app_python, child_init_method, child_init) and put in route(): python_exec(ParseSDP, $mb); Here I shared sdp.py: http://paste.debian.net/99473/ Note, that it is just for reference only, you need to know what to remove/replace. That script is not exactly what are you're looking for! ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Does tls/wss actually work or What is required for tls/wss
Hi, What are the requirements for connecting with tls/wss. I have not come across any information or example for this. My config is working when the client uses ws. However if I change this to use wss, (this is it only paramter I change) it does not work. I understand Kamailio does not support DTLS, I set the jssip client DtlsSrtpKeyAgreement to false to disable this, I also set the tls option to not require or verify certicficates and it still does not work. What if other considaerstion do I need to check? Hi, Kamailio has nothing to do with DTLS since it handles only sip signalling. To enable wss support you have to: - enable tls by enable_tls=yes - add listening socket listen=tls:x.x.x.x:443 and use this port for wss connections - configure tls module. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio without tls
Just add listen=tcp:_kamailio_server_ip:5060 to the kamailio.cfg if don't have one and change sip settings of your SIP application to disable tls. You can have both clear tcp and tls at the same time and application will choose which type of connection to use. Hi Support, I have recently kamailio 4.0.x version with tls on port 5061. Now our development team needs to connect to it using unencrypted on port 5060 to test an app for features and then work towards connecting through tls connection. So my question is how do I temporarily disable tls to test all the features on the kamailio server and then once feature testing is done revert back to tls connection? Best Regards, Neville D'Souza ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] pkg.stats problem, possible memory leak in websockets.
On 10/01/2013 12:54 PM, Peter Dunkley wrote: Done. The patch is in master and the 4.0 branch. Thanks, Peter Thanks for applying the patch. It looks like websockets in master are stable now. No memory leaks are detected for the last two days. pks.stats still shows some strange numbers, but this is another issue. Hope to get some free time in the nearest future to dig more about memory stats. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] pkg.stats problem, possible memory leak in websockets.
Hello,
Thank you for the explanation.
Could somebody review the patch in the attachment ? I tried to fix the
problem with a growing tcpconn-refcnt for websocket connections.
On 30 September 2013 17:14, Vitaliy Aleksandrov
vitalik.v...@gmail.com mailto:vitalik.v...@gmail.com wrote:
Could you please share why nathelper aggregates both WS and WSS
transports to ws and then msg_translator have to detect the type
of a connection to a destination to build correct via ?
modules/nathelper/nathelper.c create_rcv_uri() function :
case PROTO_WS:
case PROTO_WSS:
proto.s = WS;
proto.len = 2;
break;
Because when the transport is WS (WebSockets over TCP) the URI has a
transport parameter like this ;transport=ws and when the transport
is WSS (Secure WebSockets over TLS over TCP) the URI has a transport
parameter like this ;transport=ws. In other words, the transport
parameter is the same for both and you need to make the determination
within Kamailio core by checking how the specified socket is actually
used.
--
Peter Dunkley
Technical Director
Crocodile RCS Ltd
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diff --git a/msg_translator.c b/msg_translator.c
index b7d4894..b4581e7 100644
--- a/msg_translator.c
+++ b/msg_translator.c
@@ -2509,9 +2509,11 @@ char* via_builder( unsigned int *len,
} else if (con-rcv.proto==PROTO_WSS) {
memcpy(line_buf+MY_VIA_LEN-4, WSS , 4);
} else {
+ tcpconn_put(con);
LOG(L_CRIT, BUG: via_builder: unknown proto %d\n, con-rcv.proto);
return 0;
}
+ tcpconn_put(con);
}else if (send_info-proto==PROTO_WSS){
memcpy(line_buf+MY_VIA_LEN-4, WSS , 4);
}else{
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Re: [SR-Users] pkg.stats problem
I found one place where tcpconn_put() never called after tcpconn_get():
--- a/msg_translator.c
+++ b/msg_translator.c
@@ -2509,9 +2509,11 @@ char* via_builder( unsigned int *len,
} else if (con-rcv.proto==PROTO_WSS) {
memcpy(line_buf+MY_VIA_LEN-4, WSS , 4);
} else {
+ tcpconn_put(con);
LOG(L_CRIT, BUG: via_builder: unknown proto
%d\n, con-rcv.proto);
return 0;
}
+ tcpconn_put(con);
}else if (send_info-proto==PROTO_WSS){
memcpy(line_buf+MY_VIA_LEN-4, WSS , 4);
}else{
I've tried this patch and it fixed my problem with constantly growing
ref_count for WSS connections.
Could you please share why nathelper aggregates both WS and WSS
transports to ws and then msg_translator have to detect the type of a
connection to a destination to build correct via ?
modules/nathelper/nathelper.c create_rcv_uri() function :
case PROTO_WS:
case PROTO_WSS:
proto.s = WS;
proto.len = 2;
break;
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Re: [SR-Users] pkg.stats problem
At first thanks for trying to help. It's my fault that I messed up top to this story, just wanted to show that while my system is working just fine: 1. used and real_used fields of a process (tcp receiver) is bigger that I set in -M 2. free hasn't changed from the last restart. root@proxy:~# kamcmd pkg.stats | grep '2480' -A 4 pid: 2480 rank: 17 used: 34156336 free: 32611216 real_used: 35228672 root@proxy:~# ps aux|grep 2480 kamailio 2480 0.0 0.3 348160 15836 ?SSep23 0:10 /usr/local/kamailio-master/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 128 -M 32 -u kamailio -g kamailio root@proxy:~# kamctl ps | grep 2480 Process:: ID=22 PID=2480 Type=tcp receiver (generic) child=6 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] pkg.stats problem
Yes, I found you commit. That's why now I'm using latest master. tcl_list shows 200+ tcp connections and only a few of them have ref_count bigger that 1. netstat shows the same number of established connections. If lost tcp_conn structures are not shown in tcp_list how can I check if it is my problem and there are lost chunks of memory somewhere ? When I was searching for the way how correctly decrement ref_count in tcp_conn I decided to call tcpconn_put(), as you did for webcoskets. I found that there are several function to decrement it: - tcpconn_put() - tcpconn_chld_put() - atomic_dec_and_test(). When I was looking for examples in other modules I saw that tcp_read.c uses tcpconn_chld_put(), but forward.h and msg_translator.c uses only tcpconn_get() without decrementing ref counter at all. I hope someone who knows tcp implementation in kamailio can shed light on this question. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] pkg.stats problem
I switched to the latest master branch and it seems it works better, but unfortunately I can't understand how much PKG memory kamailio really uses to know it still has problems with PKG. For instance kamcmd pkg.stats always shows that tcp_main process has free: 32627984 (started with -M 32), while real_used: 22560048 and RES in top output for the same process is 8904. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] unregister user when kamailio looses TCP connection.
On 09/23/2013 11:23 PM, Andreas Granig wrote: Hi, On 09/13/2013 11:27 AM, Daniel-Constantin Mierla wrote: thanks, patch was commited and pushed to remote repository. The patch only handles the case where a tcp connection is directly made to the registrar, as no event route is fired, right? You are right. Current version works only when registrar accepts tcp connections. Anyway it's a good idea to call event_route[] when kamailio looses a tcp connection to give user a chance to process it somehow. On the one hand it's great when experienced user can achieve what he wants by different ways depending on the situation, but on the other it makes a mess for new kamailio users. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] pkg.stats problem
Additional info: - kamailio-4.0.2 started with -m 128 -M 16 - Process:: ID=36 PID=2599 Type=tcp main process - this server works as a websocket(ws,wss) to udp/tcp gateway. PKG: Also top output showed that RES column of tcp workers was constantly growing. SHM: After some time kamailio stoped to accept wss connections and printed ssl bug #1491 workaround: not enough memory for safe operation. According to tls_server.c it's a sign of shm leak. Master branch has a lot of fixes to websocket module which solve problems with pkg and shm memory: Is it a good a good idea to use websocket module from the master branch with a 4.0 verion ? Hello, can you get the type of the process with 'kamctl ps'? Cheers, Daniel On 9/20/13 6:51 PM, Vitaliy Aleksandrov wrote: Didn't check master branch before writing previous email. There were some commits about memory leaks in websocket module. Will try master. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] pkg.stats problem
Didn't check master branch before writing previous email. There were some commits about memory leaks in websocket module. Will try master. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] unregister user when kamailio looses TCP connection.
Maybe when I opened the previous patch by a text editor to check if it has all I wanted to add tabs were accidentally replaced by spaces. Anyway fixed version is in the attachment. How did you generate the patch? Apparently the tabs in existing code were replaced by whitespaces and the patch fails to apply because it doesn't match any longer the context. Can you resend in it raw format with 'git diff ../filename.patch'? Cheers, Daniel On Fri, Sep 6, 2013 at 12:46 AM, Vitaliy Aleksandrov vitalik.v...@gmail.com mailto:vitalik.v...@gmail.com wrote: Make the patch for master branch, 4.0 doesn't take any new feature and I applied the patches to master branch anyhow. Also, do not forget the patch for xml documentation. Cheers, Daniel Patch for master branch is in the attachment. Changelog: - documentation is updated - tcpconn_id initialized to -1 for contacts loaded from a database during startup (Write-Back Write-Through database modes). - tcpconn_id is being updated when UA reconnects and refreshes its registration. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda http://twitter.com/#%21/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Oct 21-24; Miami, Nov 11-13, 2013 - more details about Kamailio trainings at http://www.asipto.com - ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users diff --git a/modules/usrloc/doc/usrloc_admin.xml b/modules/usrloc/doc/usrloc_admin.xml index ce7ddfb..4f7e3e5 100644 --- a/modules/usrloc/doc/usrloc_admin.xml +++ b/modules/usrloc/doc/usrloc_admin.xml @@ -875,6 +875,26 @@ modparam(usrloc, db_ops_ruid, 1) /example /section + section id=usrloc.p.handle_lost_tcp + titlevarnamehandle_lost_tcp/varname (int)/title + para + If set to 1, Kamailio will remove location records made via + TCP/TLS/WS/WSS transports when it looses corresponding tcp connections. + /para + para + emphasis + Default value is quote0/quote. + /emphasis + example + titleSet varnamehandle_lost_tcp/varname parameter/title + programlist format=linespecific +... +modparam(usrloc, handle_lost_tcp, 1) +... +programlisting + /example + /section + /section section diff --git a/modules/usrloc/ucontact.c b/modules/usrloc/ucontact.c index e7f8938..9cc2e06 100644 --- a/modules/usrloc/ucontact.c +++ b/modules/usrloc/ucontact.c @@ -307,6 +307,7 @@ int mem_update_ucontact(ucontact_t* _c, ucontact_info_t* _ci) _c-last_keepalive = _ci-last_modified; _c-flags = _ci-flags; _c-cflags = _ci-cflags; + _c-tcpconn_id = _ci-tcpconn_id; return 0; } diff --git a/modules/usrloc/udomain.c b/modules/usrloc/udomain.c index f587c1b..244f8e7 100644 --- a/modules/usrloc/udomain.c +++ b/modules/usrloc/udomain.c @@ -351,6 +351,9 @@ static inline ucontact_info_t* dbrow2info( db_val_t *vals, str *contact) ci.reg_id = VAL_UINT(vals+15); } + /* tcp connection id */ + ci.tcpconn_id = -1; + return ci; } diff --git a/modules/usrloc/ul_mod.c b/modules/usrloc/ul_mod.c index 7d88f67..b1be56b 100644 --- a/modules/usrloc/ul_mod.c +++ b/modules/usrloc/ul_mod.c @@ -161,7 +161,7 @@ int timer_interval = 60;/*! Timer interval in seconds */ int db_mode = 0;/*! Database sync scheme: 0-no db, 1-write through, 2-write back, 3-only db */ int use_domain = 0;/*! Whether usrloc should use domain part of aor */ int desc_time_order = 0;/*! By default do not enable timestamp ordering */ -int handle_lost_tcp = 0; +int handle_lost_tcp = 0;/*! By default do not remove contacts before expiration time */ int ul_fetch_rows = 2000;/*! number of rows to fetch from result */ int ul_hash_size = 9; ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] unregister user when kamailio looses TCP connection.
This patch has a bug which leads to false contact removal. This happens when UA reconnects to a proxy from the same src port and sends new REGISTER with the same Contact. In that case kamailio updates an existing contact and I forgot to update tcp connection id in the ucontact_t object. I've fixed this bug in my production system (kamailio-3.2) and it works well. The patch for 4.0.3 will be ready tonight. Hello, I applied your patches with some changes. There is no need to store tcp connection id in other structure in the core, it is stored in msg-rcv.proto_reserved1. You have to add the documentation for the new usrloc parameter, in xml files from modules/usrloc/doc/ - send over the patch and I will commit it as well. Cheers, Daniel On 8/29/13 12:47 PM, Vitaliy Aleksandrov wrote: Thanks community for all replies. I did the second try. Result is in the attachment. This time a did it in the way proposed by Daniel-Constantin Mierla. All job is done in timer process when it iterates through all registrations looking for expired ones. It works for all database modes except DB_ONLY. I'm not sure about tcpconn_get() / tcpconn_put() functions. tcpconn_get() increments ref counter, but there are several function to decrement it: tcpconn_put(), tcpconn_chld_put() and atomic_dec_and_test(). When I was looking for examples in other modules I saw that tcp_read.c uses tcpconn_chld_put(), but forward.h and msg_translator.c uses only tcpconn_get() without decrementing ref counter at all. It would be great if somebody more experienced shed some light on this. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -http://www.asipto.com http://twitter.com/#!/miconda -http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Oct 21-24; Miami, Nov 11-13, 2013 - more details about Kamailio trainings athttp://www.asipto.com - ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] unregister user when kamailio looses TCP connection.
Make the patch for master branch, 4.0 doesn't take any new feature and I applied the patches to master branch anyhow. Also, do not forget the patch for xml documentation. Cheers, Daniel Patch for master branch is in the attachment. Changelog: - documentation is updated - tcpconn_id initialized to -1 for contacts loaded from a database during startup (Write-Back Write-Through database modes). - tcpconn_id is being updated when UA reconnects and refreshes its registration. diff --git a/modules/usrloc/doc/usrloc_admin.xml b/modules/usrloc/doc/usrloc_admin.xml index ce7ddfb..4f7e3e5 100644 --- a/modules/usrloc/doc/usrloc_admin.xml +++ b/modules/usrloc/doc/usrloc_admin.xml @@ -875,6 +875,26 @@ modparam(usrloc, db_ops_ruid, 1) /example /section + section id=usrloc.p.handle_lost_tcp + titlevarnamehandle_lost_tcp/varname (int)/title + para + If set to 1, Kamailio will remove location records made via + TCP/TLS/WS/WSS transports when it looses corresponding tcp connections. + /para + para + emphasis + Default value is quote0/quote. + /emphasis + example + titleSet varnamehandle_lost_tcp/varname parameter/title + programlist format=linespecific +... +modparam(usrloc, handle_lost_tcp, 1) +... +programlisting + /example + /section + /section section diff --git a/modules/usrloc/ucontact.c b/modules/usrloc/ucontact.c index e7f8938..9cc2e06 100644 --- a/modules/usrloc/ucontact.c +++ b/modules/usrloc/ucontact.c @@ -307,6 +307,7 @@ int mem_update_ucontact(ucontact_t* _c, ucontact_info_t* _ci) _c-last_keepalive = _ci-last_modified; _c-flags = _ci-flags; _c-cflags = _ci-cflags; + _c-tcpconn_id = _ci-tcpconn_id; return 0; } diff --git a/modules/usrloc/udomain.c b/modules/usrloc/udomain.c index f587c1b..244f8e7 100644 --- a/modules/usrloc/udomain.c +++ b/modules/usrloc/udomain.c @@ -351,6 +351,9 @@ static inline ucontact_info_t* dbrow2info( db_val_t *vals, str *contact) ci.reg_id = VAL_UINT(vals+15); } + /* tcp connection id */ + ci.tcpconn_id = -1; + return ci; } diff --git a/modules/usrloc/ul_mod.c b/modules/usrloc/ul_mod.c index 7d88f67..b1be56b 100644 --- a/modules/usrloc/ul_mod.c +++ b/modules/usrloc/ul_mod.c @@ -161,7 +161,7 @@ int timer_interval = 60; /*! Timer interval in seconds */ int db_mode = 0; /*! Database sync scheme: 0-no db, 1-write through, 2-write back, 3-only db */ int use_domain = 0; /*! Whether usrloc should use domain part of aor */ int desc_time_order = 0; /*! By default do not enable timestamp ordering */ -int handle_lost_tcp = 0; +int handle_lost_tcp = 0; /*! By default do not remove contacts before expiration time */ int ul_fetch_rows = 2000; /*! number of rows to fetch from result */ int ul_hash_size = 9; ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] kamailio load balancing
As written many time in this mailing list it depends on what services you want to provide using such voice servers. If they are transparent media relays then rtpproxy or mediaproxy(-ng) can help. Also you can integrate it with your billing to authorize calls and limit their duration. Though if you want to provide voicemail, call recording, transcoding, etc.. you have to use software like asterisk or freeswitch. Hello, I have a question about the load balancing module of kamailio. As the site http://kb.asipto.com/ say, Kamailio is as a SIP proxy router to scale Asterisk. Can I run a kamailio instance as load balancer, and other several instances as voice server replace of Asterisk? If I can do that, could you give me a tutorial? We are using kamailio as our server. Thank you very much. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] unregister user when kamailio looses TCP connection.
Thanks community for all replies.
I did the second try. Result is in the attachment.
This time a did it in the way proposed by Daniel-Constantin Mierla.
All job is done in timer process when it iterates through all
registrations looking for expired ones.
It works for all database modes except DB_ONLY.
I'm not sure about tcpconn_get() / tcpconn_put() functions.
tcpconn_get() increments ref counter, but there are several function to
decrement it: tcpconn_put(), tcpconn_chld_put() and
atomic_dec_and_test(). When I was looking for examples in other modules
I saw that tcp_read.c uses tcpconn_chld_put(), but forward.h and
msg_translator.c uses only tcpconn_get() without decrementing ref
counter at all.
It would be great if somebody more experienced shed some light on this.
diff -uNpr kamailio-4.0.3/ip_addr.h sshfs_kamailio-4.0.3/ip_addr.h
--- kamailio-4.0.3/ip_addr.h 2013-08-15 16:42:39.0 +0300
+++ sshfs_kamailio-4.0.3/ip_addr.h 2013-08-28 18:28:08.0 +0300
@@ -142,6 +142,7 @@ struct receive_info{
struct socket_info* bind_address; /* sock_info structure on which
the msg was received*/
char proto;
+ int tcpconn_id; /* tcp_connnection-id */
#ifdef USE_COMP
short comp; /* compression */
#endif
diff -uNpr kamailio-4.0.3/modules/registrar/save.c sshfs_kamailio-4.0.3/modules/registrar/save.c
--- kamailio-4.0.3/modules/registrar/save.c 2013-08-15 16:42:39.0 +0300
+++ sshfs_kamailio-4.0.3/modules/registrar/save.c 2013-08-29 01:44:11.0 +0300
@@ -259,6 +259,14 @@ static inline ucontact_info_t* pack_ci(
ci.sock = _m-rcv.bind_address;
}
+ /* set tcp connection id */
+ if (_m-rcv.proto==PROTO_TCP || _m-rcv.proto==PROTO_TLS
+ || _m-rcv.proto==PROTO_WS || _m-rcv.proto==PROTO_WSS) {
+ ci.tcpconn_id = _m-rcv.tcpconn_id;
+ } else {
+ ci.tcpconn_id = -1;
+ }
+
/* additional info from message */
if (parse_headers(_m, HDR_USERAGENT_F, 0) != -1 _m-user_agent
_m-user_agent-body.len0 _m-user_agent-body.lenUA_MAX_SIZE) {
@@ -488,7 +496,7 @@ static inline int insert_contacts(struct
/* hack to work with buggy clients having many contacts with same
* address in one REGISTER - increase CSeq to detect if there was
- * one alredy added, then update */
+ * one already added, then update */
ci-cseq++;
if ( r-contacts==0
|| ul.get_ucontact_by_instance(r, _c-uri, ci, c) != 0) {
diff -uNpr kamailio-4.0.3/modules/usrloc/ucontact.c sshfs_kamailio-4.0.3/modules/usrloc/ucontact.c
--- kamailio-4.0.3/modules/usrloc/ucontact.c 2013-08-15 16:42:39.0 +0300
+++ sshfs_kamailio-4.0.3/modules/usrloc/ucontact.c 2013-08-29 01:49:56.0 +0300
@@ -135,6 +135,7 @@ ucontact_t* new_ucontact(str* _dom, str*
c-reg_id = _ci-reg_id;
c-last_modified = _ci-last_modified;
c-last_keepalive = _ci-last_modified;
+ c-tcpconn_id = _ci-tcpconn_id;
#ifdef WITH_XAVP
ucontact_xavp_store(c);
#endif
diff -uNpr kamailio-4.0.3/modules/usrloc/ul_mod.c sshfs_kamailio-4.0.3/modules/usrloc/ul_mod.c
--- kamailio-4.0.3/modules/usrloc/ul_mod.c 2013-08-15 16:42:39.0 +0300
+++ sshfs_kamailio-4.0.3/modules/usrloc/ul_mod.c 2013-08-29 13:04:13.0 +0300
@@ -161,6 +161,7 @@ int timer_interval = 60;/*! Timer
int db_mode = 0;/*! Database sync scheme: 0-no db, 1-write through, 2-write back, 3-only db */
int use_domain = 0;/*! Whether usrloc should use domain part of aor */
int desc_time_order = 0;/*! By default do not enable timestamp ordering */
+int handle_lost_tcp = 0;
int ul_fetch_rows = 2000;/*! number of rows to fetch from result */
int ul_hash_size = 9;
@@ -214,6 +215,7 @@ static param_export_t params[] = {
{fetch_rows, INT_PARAM, ul_fetch_rows },
{hash_size, INT_PARAM, ul_hash_size},
{nat_bflag, INT_PARAM, nat_bflag },
+ {handle_lost_tcp, INT_PARAM, handle_lost_tcp },
{preload, STR_PARAM|USE_FUNC_PARAM, (void*)ul_preload_param},
{db_update_as_insert, INT_PARAM, ul_db_update_as_insert},
{timer_procs, INT_PARAM, ul_timer_procs},
@@ -387,6 +389,10 @@ static int mod_init(void)
return -1;
}
}
+
+ if (handle_lost_tcp db_mode == DB_ONLY)
+ LM_WARN(handle_lost_tcp option makes nothing in DB_ONLY mode\n);
+
init_flag = 1;
return 0;
diff -uNpr kamailio-4.0.3/modules/usrloc/ul_mod.h sshfs_kamailio-4.0.3/modules/usrloc/ul_mod.h
--- kamailio-4.0.3/modules/usrloc/ul_mod.h 2013-08-15 16:42:38.0 +0300
+++ sshfs_kamailio-4.0.3/modules/usrloc/ul_mod.h 2013-08-29 02:06:38.0 +0300
@@ -84,6 +84,7 @@ extern int ul_hash_size;
extern int ul_db_update_as_insert;
extern int ul_db_check_update;
extern int ul_keepalive_timeout;
+extern int handle_lost_tcp;
/*! nat branch flag */
extern unsigned int nat_bflag;
diff -uNpr kamailio-4.0.3/modules/usrloc/urecord.c sshfs_kamailio-4.0.3/modules/usrloc/urecord.c
--- kamailio-4.0.3/modules/usrloc/urecord.c 2013-08-15 16:42:39.0 +0300
+++
Re: [SR-Users] unregister user when kamailio looses TCP connection.
On 08/28/2013 06:45 PM, Juha Heinanen wrote: Vitaliy Aleksandrov writes: If anybody else except me need this It would be great to fix known problems and add it to kamailio. i don't know if this come already up, but why not use this in branch failure route: unregister(location, , $T_reply_rid); Didn't know about $T_reply_rid variable and that unregister can remove only single contact. In my case the problem with unregister is that stale contact will be removed only if somebody tries to call to a disconnected phone. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] unregister user when kamailio looses TCP connection.
On 08/27/2013 10:22 AM, Olle E. Johansson wrote: Let's work on a description on the logic needed and see if Vitally's code is close: 1. Connection dies (tcp, tls, sctp) 2. Event-route activates 3. Check if there's a outbound flow with reg-id associated with the flow 4. If not, is there a registration associated with the flow 5. If 3 or 4 is true, do something Do we have functions for 3 and 4? Can this be done in scripting? Haven't seen functions which can do 3 and 4 at the last stable usrloc module. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] unregister user when kamailio looses TCP connection.
On 08/27/2013 11:57 AM, Daniel-Constantin Mierla wrote: Hello, thanks for the patch. Some remarks. As you highlighted that the solution is not very efficient (due to no direct mapping between records and tcp connection), it should have a way to disable it (maybe via mod param for usrloc). At this moment is hardcoded for 'location' table, but there can be other names used, since this value is given as parameter to save()/lookup(). It will have to iterate through all usrloc tables (called domains interanlly in the module). Ok, I'll remove hardcoded location domain. It will search in all domains from dlist_t. Module parameter will be added too. I understand that search through the all contacts is not for production use at all. I really need this feature and I'm sure everyone who uses tcp/tls/ws too so I wanted to start this discussion from a little peace of code to get feedback from community. Also, if I didn't get it wrong, it works only with in-memory records, if the usrloc is configured in db-only mode, it does not work with current patch, right? To remove old contacts i've used delete_ucontact from usrloc api. This function checks db_mode and removes contact from db in case of DB_ONLY mode and just updates expiration in case of WRITE_BACK. I had in mind to implement such features, but using timer, to iterate through tcp connections and see if there is still an open connection for it. This will do delayed discovery, but I am not sure it will bring more benefits -- initially I thought is faster lookup done by a dedicated timer, presumly there are less tcp connections that usrloc records. Anyhow, perhaps we should do some optimization here, we can store the connection id in usrloc structure and then match received address only if there is same conenction id. Or add received built from tcp connection close callback in a list that is used by usrloc expires timers. Searching through a single list from a separate process sounds much better than my version. I'll try to implement this when I have free time. The way Olle proposed looks more preferable as it can work in environment with separate registrar and outbound servers. But I have to read some theory about outbound before moving this way. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] unregister user when kamailio looses TCP connection.
Also, if I didn't get it wrong, it works only with in-memory records, if the usrloc is configured in db-only mode, it does not work with current patch, right? To remove old contacts i've used delete_ucontact from usrloc api. This function checks db_mode and removes contact from db in case of DB_ONLY mode and just updates expiration in case of WRITE_BACK. But the issue would be that the hash tables will not be in memory, so you will not get in this way to call the delete_ucontact() function. This function is called via a different code path for db only mode. Tnx, good remark. It shows that I missed how DB_ONLY mode works. Will spent some time to examine it. Just remembered that kamailio can close connection from its side due to tcp_connection_lifetime parameter. Haven't checked where how it works. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] unregister user when kamailio looses TCP connection.
Hello,
I've made a patch to kamailio-4.0.3 which removes stale registration
when kamailio looses an incoming tcp connection.
Of course this patch needs more work.
Since the are no direct references between user location contacts and
tcp connections callback function uses linear search through the whole
location table using received field as a key.
Can anybody more experienced in kamailio internals check if I chose the
right place to get information about lost tcp connections ?
Another thing I wanted to ask is maybe somebody can suggest a better way
to tie a tcp connection to the user location information without
complicating usrloc module by any heavy data structures.
If anybody else except me need this It would be great to fix known
problems and add it to kamailio.
diff -uNr kamailio-4.0.3/events.c sshfs-4.0.3/events.c
--- kamailio-4.0.3/events.c 2013-08-15 16:42:39.0 +0300
+++ sshfs-4.0.3/events.c 2013-08-26 17:02:15.0 +0300
@@ -106,6 +106,11 @@
_sr_events_list.tcp_ws_frame_out = f;
else return -1;
break;
+ case SREV_TCP_CONN_LOST:
+if(_sr_events_list.tcp_conn_lost==0)
+ _sr_events_list.tcp_conn_lost = f;
+else return -1;
+ break;
default:
return -1;
}
@@ -209,6 +214,12 @@
ret = _sr_events_list.tcp_ws_frame_out(data);
return ret;
} else return 1;
+ case SREV_TCP_CONN_LOST:
+if(unlikely(_sr_events_list.tcp_conn_lost!=0))
+{
+ ret = _sr_events_list.tcp_conn_lost(data);
+ return ret;
+} else return 1;
default:
return -1;
}
@@ -242,6 +253,8 @@
return (_sr_events_list.tcp_ws_frame_in!=0)?1:0;
case SREV_TCP_WS_FRAME_OUT:
return (_sr_events_list.tcp_ws_frame_out!=0)?1:0;
+ case SREV_TCP_CONN_LOST:
+return (_sr_events_list.tcp_conn_lost!=0)?1:0;
}
return 0;
}
diff -uNr kamailio-4.0.3/events.h sshfs-4.0.3/events.h
--- kamailio-4.0.3/events.h 2013-08-15 16:42:39.0 +0300
+++ sshfs-4.0.3/events.h 2013-08-26 17:01:51.0 +0300
@@ -30,10 +30,11 @@
#define SREV_PKG_SET_USED 5
#define SREV_PKG_SET_REAL_USED 6
#define SREV_NET_DGRAM_IN 7
-#define SREV_TCP_HTTP_100C 8
-#define SREV_TCP_MSRP_FRAME 9
-#define SREV_TCP_WS_FRAME_IN 10
-#define SREV_TCP_WS_FRAME_OUT 11
+#define SREV_TCP_HTTP_100C 8
+#define SREV_TCP_MSRP_FRAME 9
+#define SREV_TCP_WS_FRAME_IN10
+#define SREV_TCP_WS_FRAME_OUT 11
+#define SREV_TCP_CONN_LOST 12
typedef int (*sr_event_cb_f)(void *data);
@@ -50,6 +51,7 @@
sr_event_cb_f tcp_msrp_frame;
sr_event_cb_f tcp_ws_frame_in;
sr_event_cb_f tcp_ws_frame_out;
+ sr_event_cb_f tcp_conn_lost;
} sr_event_cb_t;
void sr_event_cb_init(void);
diff -uNr kamailio-4.0.3/modules/usrloc/tcp_conn_cb.c sshfs-4.0.3/modules/usrloc/tcp_conn_cb.c
--- kamailio-4.0.3/modules/usrloc/tcp_conn_cb.c 1970-01-01 03:00:00.0 +0300
+++ sshfs-4.0.3/modules/usrloc/tcp_conn_cb.c 2013-08-26 18:36:16.0 +0300
@@ -0,0 +1,179 @@
+#include ../../dprint.h
+#include ../../tcp_conn.h
+#include ../../ip_addr.h
+#include tcp_conn_cb.h
+#include usrloc.h
+#include dlist.h
+#include udomain.h
+#include urecord.h
+#include ucontact.h
+
+/* result points to a static buffer.
+ * Ones should copy it before the next call to not loose data */
+static int
+create_rcv_uri(str* uri, struct tcp_connection *c)
+{
+ static char buf[MAX_URI_SIZE];
+ char *p;
+ str ip, port;
+ int len;
+ str proto;
+
+ if (!uri || !c) {
+ LM_ERR(invalid parameter value\n);
+ return -1;
+ }
+
+ ip.s = ip_addr2a(c-rcv.src_ip);
+ ip.len = strlen(ip.s);
+
+ port.s = int2str(c-rcv.src_port, port.len);
+
+ switch(c-rcv.proto) {
+ case PROTO_NONE:
+ case PROTO_UDP:
+ proto.s = 0; /* Do not add transport parameter, UDP is default */
+ proto.len = 0;
+ break;
+
+ case PROTO_TCP:
+ proto.s = TCP;
+ proto.len = 3;
+ break;
+
+ case PROTO_TLS:
+ proto.s = TLS;
+ proto.len = 3;
+ break;
+
+ case PROTO_SCTP:
+ proto.s = SCTP;
+ proto.len = 4;
+ break;
+
+ case PROTO_WS:
+ case PROTO_WSS:
+ proto.s = WS;
+ proto.len = 2;
+ break;
+
+ default:
+ LM_ERR(unknown transport protocol\n);
+ return -1;
+ }
+
+ /* sip: + ip + [only for ipv6] + : + port */
+ len = 4 + ip.len + 2*(c-rcv.src_ip.af==AF_INET6) + 1 + port.len;
+ if (proto.s) {
+ len += TRANSPORT_PARAM_LEN;
+ len += proto.len;
+ }
+
+ if (len MAX_URI_SIZE) {
+ LM_ERR(buffer too small\n);
+ return -1;
+ }
+
+ p = buf;
+ memcpy(p, sip:, 4);
+ p += 4;
+
+ if (c-rcv.src_ip.af==AF_INET6)
+ *p++ = '[';
+ memcpy(p, ip.s, ip.len);
+ p += ip.len;
+ if (c-rcv.src_ip.af==AF_INET6)
+ *p++ = ']';
+
+ *p++ = ':';
+
+ memcpy(p, port.s, port.len);
+ p += port.len;
+
+ if (proto.s) {
+ memcpy(p, TRANSPORT_PARAM, TRANSPORT_PARAM_LEN);
+ p += TRANSPORT_PARAM_LEN;
+
+ memcpy(p, proto.s, proto.len);
+ p += proto.len;
+ }
+
+ uri-s = buf;
+ uri-len = len;
+
+ return 0;
+}
+
+static int remove_by_rcv(str *rcv)
+{
+ udomain_t *d = NULL;
+ urecord_t *r = NULL;
+ ucontact_t *c = NULL;
+ str d_name = {location, strlen(location)};
+ int
Re: [SR-Users] Finding Destination IP:Port for 200 OK
AFAIK there is no any single pv that provides a destination for both sip
message types (request/reply).
As you can see from my example $sel(next_hop.host) contains dst only for
sip requests. To know where kamailio is going to send
a reply you can check the second VIA header parameters like received or
host if received is not present.
BTW after calling such a route $var(next_hop) or any other you want to
use has what you need in a single pv.
Thanks Vitaliy,
That seems working, I am testing with it..it's great.
Can I ask if there is any psuedo variable for this thing !
Best Regards,
Sammy
On Sat, Aug 24, 2013 at 3:10 PM, Vitaliy Aleksandrov
vitalik.v...@gmail.com mailto:vitalik.v...@gmail.com wrote:
How about next config snippet:
/route[GET_NEXT_HOP]//
//{//
//$var(next_hop) = $null;//
//if (is_request()) {//
//$var(next_hop) = $sel(next_hop.host);//
//} else if (is_reply()) {//
// if ($sel(via[2].received) != $null)//
//$var(next_hop) = $sel(via[2].received);//
//else//
//$var(next_hop) = $sel(via[2].host);//
//}//
//}/
If you need to know what next hop kamailio will use to forward
current sip message (request/reply) you can just call
route(GET_NEXT_HOP) which puts what you need into $var(next_hop).
Dear list,
Its just a little thing to ask but I somehow can't seem to figure
out which psuedo variable to use to find out where a particular
200 OK is destined to go.
the scenario is I've multiple media servers in dispatcher and
calls are distributed. I need to capture a 200OK that is destined
to go to a particular ip!
I've tried using the dst_ip, $dd, $du but none of them are giving
any results for the 200 OK. I just need to know where do kamailio
is going to send this 200 since it is already a part of an
on-going INVITE/call.
Thanks,
Sammy
___
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___
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list
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http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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Re: [SR-Users] Kamailio v4.0.2, iPhone, TCP connection and PJLIB
On 08/14/2013 07:32 PM, Roberto Fichera wrote: On 08/14/2013 04:36 PM, Vitaliy Aleksandrov wrote: If you won't be able to disable SIP ALG on your router you can fill $avp(received) manually before calling save(): $avp(received) = sip: + $si + : + $sp + ;transport= + $proto; In this case all user location records will have the received attribut even if a UA isn't behind NAT, but I don't see any problems with that. This one looks working, but the callee doesn't answer correctly because the TCP isn't correct: Contact:: sip:528@94.94.X.X:1380;transport=TCP;ob;q=;expires=294;flags=0x0;cflags=0x0;state=0;socket=tcp:178.79.X.X:5060;methods=0x1FDF;received=sip:94.94.X.X:37030;transport=tcp;user_agent=PJSUA v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17;reg-id=0 The contact uses a port which isn't translated inside by the router, the received field shows the right one. Should I change the Contact header instead? In case, how can I do that? I didn't understand what callee doesn't answer correctly means. Callee doesn't know what's in the received field of its registration. The only problem I see is that your router changes ip in the contact field of REGISTER requests and then kamailio puts this value (new_ip:old_port) to INVITEs destined to UAs behind your NAT router. It's not likely, but maybe pjsip doesn't like INVITEs with RURI which differs from what it put to the contact during registration. IIRC pjsua prints lots of debugging information. So you can check if pjsua shows anything when INVITE comes in its log and also attach a trace of such a call. P.S. IMHO the best way to pass sip through such routers is sip over TLS. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Two servers with shared registration database
On 08/01/2013 03:30 PM, Eugene Prokopiev wrote: Hi, Is it possible to configure two Kamailio servers with shared registration database? Any SIP UA can be registered on any server but it must transparently call another SIP UA which can be registered on the same or on another Kamailio server. Is where any example of this configuration? Hello, Yes it is possible. usrloc module has db_mode options. If it is set to 3 (db only) mode it will be possible to call to a registered UA via either of sip servers. Haven't used this scheme in practice. Community, correct me if I'm wrong. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] websocket double record_route()
Hello. I'm trying to configure kamailio as a gateway between Websocket and TCP/TLS transports. When I call record_route() for an initial INVITE that comes via WS and will be forwarded via TCP to a registered UA kamailio inserts only a one record-route header with its IP and transport=ws instead of two record-route headers with both incoming/outgoing transports. This behaviour breaks in-dialog requests routing. rr module parameters are: modparam(rr, enable_full_lr, 1) modparam(rr, append_fromtag, 0) modparam(rr, enable_double_rr, 1) I use: - kamailio-4.0.2 - sipml5 as sip client Can anybody point me in the right direction to understand why it happens ? ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] websocket double record_route()
Hello, Thanks for a quick reply. No, outbound module is not loaded. As I understood from RR docs double_rr won't work if outbound module is present. I've added mhomed=1 to my config, but record_route() still inserts only one rr with transport=ws. Are you using outbound? If you are I suggest trying git master rather than the 4.0 branch. Regards, Peter On 31 July 2013 10:53, Vitaliy Aleksandrov vitalik.v...@gmail.com mailto:vitalik.v...@gmail.com wrote: Hello. I'm trying to configure kamailio as a gateway between Websocket and TCP/TLS transports. When I call record_route() for an initial INVITE that comes via WS and will be forwarded via TCP to a registered UA kamailio inserts only a one record-route header with its IP and transport=ws instead of two record-route headers with both incoming/outgoing transports. This behaviour breaks in-dialog requests routing. rr module parameters are: modparam(rr, enable_full_lr, 1) modparam(rr, append_fromtag, 0) modparam(rr, enable_double_rr, 1) I use: - kamailio-4.0.2 - sipml5 as sip client Can anybody point me in the right direction to understand why it happens ? ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] websocket double record_route()
Hello, does it add two record route headers when calling to an UDP destination? IIRC, the condition for double route is that incoming socket is different than outgoing socket and I am not sure if Peter updated it to detect a different sub-protocol type (as ws/wss are on top of tcp/tls). Cheers, Daniel Hello, That's it ! I've connected the phone using udp as transport and kamailio added two record route headers. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] websocket double record_route()
It works. Thanks a lot. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] websocket double record_route()
On 07/31/2013 04:18 PM, Vitaliy Aleksandrov wrote: It works. Thanks a lot. I'm going to test this with the latest master branch as Peter Dunkley proposed. Should I add this issue to Jira if this behaviour is not fixed in the master ? ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio as a sip/rtp proxy
For example you can add dns records for
customer{one,two,...}.whatever.com pointing to your proxy.
Then proxy can forward requests from outside world to asterisks based on
uri domain part (customerone - 192.168.0.5, two- .10).
You can keep table of uri-ip conversions either in the kamailio.cfg or
in any database you like.
Maybe it's not the best way but you can start from it.
IIRC carrierroute module can be used for such routing, so you can check
it too.
Hi there,
I am trying to implement a sip/rtp proxy.
I have one public ip address with a domain name. example (whatever.com
http://whatever.com)
On the lan side i have multiple asterisk instances all listening on
5060 for sip requests. These instances will be assigned to different
customers
I am hoping to run kamailio in front of these instances and route sip
and rtp traffic to the correct asterisk instance.
So in this use case asterisk instances will be used as application
servers and kamailio will be used as a sip/rtpproxy with nathelper
So my question is what would be the best way and using which modules
can i route the requests for customerone.whatever.com
http://customerone.whatever.com to the first asterisk instance and
customertwo.whatever.com http://customertwo.whatever.com to the second.
is this achieved with the dispatcher module? and if so, where can it
be defined that requests for cusomertone.whatever.com
http://cusomertone.whatever.com is routed to the first asterisk
instance.
Kind regards,
Cormac O'Loughlin
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Re: [SR-Users] How to tie Registration with Client TCP connection
Hello, Is this option still not available ? I'm asking because there was a message from, if I recall right, Olle Johansson that removing stale registrations is on the todo list of someone from the dev team. So maybe someone already has this option in its devel branch ? On Fri, Apr 5, 2013 at 10:48 AM, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, On 4/4/13 10:43 PM, Krishna Kurapati wrote: Hi, With TCP, there is no way Server can establish a TCP connection through client's Firewall/NAT. So, when Client connection is disconnected from the server due to Network issues, the Registration should be removed. I tried to handle this with a Event Route in tcp_read.c. However, There is not an easy way to find the Registration with TCP socket info (IP and Port of the Client). Could anyone suggest a solution? you can use set_forward_no_connect http://www.kamailio.org/wiki/**cookbooks/4.0.x/core#set_** forward_no_connecthttp://www.kamailio.org/wiki/cookbooks/4.0.x/core#set_forward_no_connect Local branches will be created, but not forwarded if no connection matches the destination. If you follow the code for this functionality, then you should see how the connection is associated with a destination. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/**micondahttp://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com - __**_ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**usershttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Asterisk cluster with Kamailio for load balancing.
Indeed, this is a very good suggestion. I can propose a little plan for the first step: 1. Learn how sip registration works 2. Configure kamailio as a sip registrar server. 3. Try to route calls between registered phones. 4. Read about in-dialog requests routing (record_route(), loose_route()) and configure calls between connected endpoint in a right way. And if you want to learn mode: 5. Try to handle NAT issues at kamailio (symmetric sip / rtp relay) Asterisk supports symmetric rtp pretty well, so if you want to route all calls through asterisk boxes you can skip rtp relay configuration step. We use Kamailio for registration of users and loadbalancing. I think you can start with next steps: 1. Single installation of kamailio with registration of users, fixing NAT problems etc. 2. After you get some experience with Kamailio - you can add Asterisk servers. In asterisk servers - I accept all calls which comes from Kamailio - registration of users is purely Kamailio job. In Kamailio - again I accept all calls which come from any of Asterisk servers. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Why does Kamailio return many Contacts in Contact header in REGISTER response ?
Kamailio can do this but it doesn't by default. http://kamailio.org/docs/modules/4.0.x/modules/registrar.html#idp110136 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] How to make BLF work
I think you have missed the fact that you don't have PUA component. PUA client is the thing that should send PUBLISH requests with dialoginfo body to kamailio. Kamailio in its turn will use mentioned PUBLISHes to send NOTIFY to your phones. Hi again, I doubt SCA module will work for me but let me get this a shot tonight and get back with a sipgrep trace. Meanwhile I was just thinking if the 'skype like service' blog at Asipto http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour might work straight away. I really don;t see any difference in my PRESENCE route but maybe I've missed anything. Let me get back with more working on this. Thanks for the reply. Sammy Go. On Wed, May 29, 2013 at 1:35 PM, Daniel-Constantin Mierla mico...@gmail.com mailto:mico...@gmail.com wrote: Hello, just to say that SCA module might be an alternative, not sure it works for blf, could be only for bla/sla, but worth checking the email. Also, if you post ngrep output (sip traffic as sip), could be easier to review the traces from a device that is not yet capable of installing tcpdump/ngrep/etc ... tools, like tablets, smartphones ... increasing your chance to get more people to an answer. Cheers, Daniel On 5/28/13 11:41 PM, SamyGo wrote: Yeah, I'm getting your point. So now, here are my loaded modules. #!ifdef WITH_PRESENCE loadmodule presence.so loadmodule presence_xml.so loadmodule presence_dialoginfo.so loadmodule pua.so loadmodule pua_dialoginfo.so loadmodule pua_usrloc.so #!endif And of course: loadmodule dialog.so Attached are the relevant sip traces. On Wed, May 29, 2013 at 2:32 AM, SamyGo govoi...@gmail.com mailto:govoi...@gmail.com wrote: Yeah, I'm getting your point. So now, here are my loaded modules. #!ifdef WITH_PRESENCE loadmodule presence.so loadmodule presence_xml.so loadmodule presence_dialoginfo.so loadmodule pua.so loadmodule pua_dialoginfo.so loadmodule pua_usrloc.so #!endif And of course: loadmodule dialog.so Attached are the relevant sip traces. On Wed, May 29, 2013 at 1:00 AM, Vitaliy Aleksandrov vitalik.v...@gmail.com mailto:vitalik.v...@gmail.com wrote: You didn't receive any NOTIFY requests because you didn't have a PUA client. rfc3903 describes what a PUA client is. To make BLF work you can try to use the next modules: 1. dialog - to enable dialog awareness 2. pua + pua_dialoginfo - to enable a PUA client based on active dialogs information provided by the dialog module 3. presence + presence_dialoginfo - to handle subscriptions and send notifications to phones Hi list, I've been trying to make my Yealink phone to give BLF indications but I haven't been able to achieve this successfully yet so I need some expert advise here. My Yealink phone, as soon as it registers to Asterisk, gives me BLF lights. The same phone registering to Kamailio sends SUBSCRIBE and nothing happens. The Presence Handling is enabled in configurations and yet no dialog-info+xml NOTIFY is received. Please help. Best Regards, Sammy Go. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@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@lists.sip-router.org mailto:sr-users@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@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -http://www.asipto.com http://twitter.com/#!/miconda http://twitter.com/#%21/miconda -http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 *http://asipto.com/u/katu * ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP
Re: [SR-Users] How to make BLF work
Have you checked presence_dialoginfo documentation ? http://kamailio.org/docs/modules/devel/modules/presence_dialoginfo.html The link above has a description of the way BLF can be implemented on kamailio and I think it has all the details you need to make it work. I haven't used this scheme and know about it only from the mentioned paper, so I can't explain it in more detail. Yes, As OEJ mentioned there were no Dialogs created so had to put the dlg_manage(); in the cfg and now in kamcmd the dlg.list command displays call data. How can I add presence stuff now. Vitaliy, please explain further how can I add the PUA component. Thanks, Sammy On Thu, May 30, 2013 at 6:17 PM, Olle E. Johansson o...@edvina.net mailto:o...@edvina.net wrote: 30 maj 2013 kl. 13:07 skrev SamyGo govoi...@gmail.com mailto:govoi...@gmail.com: Hi again, I doubt SCA module will work for me but let me get this a shot tonight and get back with a sipgrep trace. Meanwhile I was just thinking if the 'skype like service' blog at Asipto http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour might work straight away. I really don;t see any difference in my PRESENCE route but maybe I've missed anything. Let me get back with more working on this. Start with making sure that the Dialog module works as expected for you. Check with kamcmd that you have active dialogs when you have active calls. When that works, start with adding presence. Cheers, /O Thanks for the reply. Sammy Go. On Wed, May 29, 2013 at 1:35 PM, Daniel-Constantin Mierla mico...@gmail.com mailto:mico...@gmail.com wrote: Hello, just to say that SCA module might be an alternative, not sure it works for blf, could be only for bla/sla, but worth checking the email. Also, if you post ngrep output (sip traffic as sip), could be easier to review the traces from a device that is not yet capable of installing tcpdump/ngrep/etc ... tools, like tablets, smartphones ... increasing your chance to get more people to an answer. Cheers, Daniel On 5/28/13 11:41 PM, SamyGo wrote: Yeah, I'm getting your point. So now, here are my loaded modules. #!ifdef WITH_PRESENCE loadmodule presence.so loadmodule presence_xml.so loadmodule presence_dialoginfo.so loadmodule pua.so loadmodule pua_dialoginfo.so loadmodule pua_usrloc.so #!endif And of course: loadmodule dialog.so Attached are the relevant sip traces. On Wed, May 29, 2013 at 2:32 AM, SamyGo govoi...@gmail.com mailto:govoi...@gmail.com wrote: Yeah, I'm getting your point. So now, here are my loaded modules. #!ifdef WITH_PRESENCE loadmodule presence.so loadmodule presence_xml.so loadmodule presence_dialoginfo.so loadmodule pua.so loadmodule pua_dialoginfo.so loadmodule pua_usrloc.so #!endif And of course: loadmodule dialog.so Attached are the relevant sip traces. On Wed, May 29, 2013 at 1:00 AM, Vitaliy Aleksandrov vitalik.v...@gmail.com mailto:vitalik.v...@gmail.com wrote: You didn't receive any NOTIFY requests because you didn't have a PUA client. rfc3903 describes what a PUA client is. To make BLF work you can try to use the next modules: 1. dialog - to enable dialog awareness 2. pua + pua_dialoginfo - to enable a PUA client based on active dialogs information provided by the dialog module 3. presence + presence_dialoginfo - to handle subscriptions and send notifications to phones Hi list, I've been trying to make my Yealink phone to give BLF indications but I haven't been able to achieve this successfully yet so I need some expert advise here. My Yealink phone, as soon as it registers to Asterisk, gives me BLF lights. The same phone registering to Kamailio sends SUBSCRIBE and nothing happens. The Presence Handling is enabled in configurations and yet no dialog-info+xml NOTIFY is received. Please help. Best Regards, Sammy Go. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin
Re: [SR-Users] How to make BLF work
You didn't receive any NOTIFY requests because you didn't have a PUA client. rfc3903 describes what a PUA client is. To make BLF work you can try to use the next modules: 1. dialog - to enable dialog awareness 2. pua + pua_dialoginfo - to enable a PUA client based on active dialogs information provided by the dialog module 3. presence + presence_dialoginfo - to handle subscriptions and send notifications to phones Hi list, I've been trying to make my Yealink phone to give BLF indications but I haven't been able to achieve this successfully yet so I need some expert advise here. My Yealink phone, as soon as it registers to Asterisk, gives me BLF lights. The same phone registering to Kamailio sends SUBSCRIBE and nothing happens. The Presence Handling is enabled in configurations and yet no dialog-info+xml NOTIFY is received. Please help. Best Regards, Sammy Go. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Serial forking
I missed that $branch(..) variable is r/w. I like this way more than playing with AVPs. According to documentation $branch(..) variable gives access to additional branches. Is it possible to change Q value for the first branch ? ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Serial forking
Hello. As I understand t_load_contacts() / t_next_contacts() can help only in case when branches have different q values. The only way I see is to get all contacts via reg_fetch_contacts(), save the list to an AVP and then iterate through the list in the way described in tm module docs. Does anybody know a better way of how serial forking can be implemented if all contacts have equal q values ? ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] var vs avp...
On 04/29/2013 03:52 PM, Victor V. Kustov wrote:
В Mon, 29 Apr 2013 14:46:34 +0200
Olle E. Johansson o...@edvina.net пишет:
All functions of this module load AVPs from SIP-AVP reply items
received from RADIUS upon a successful request. Value of the SIP-AVP
reply item must be a string of form:
• value = SIP_AVP_NAME SIP_AVP_VALUE
• SIP_AVP_NAME = STRING_NAME | '#'ID_NUMBER
• SIP_AVP_VALUE = ':'STRING_VALUE | '#'NUMBER_VALUE
sorry its not clear for me.
$avp('next-hop-ip') is ok in my case? Or i need _special_ radius answer?
Not so special .. You need to add SIP-AVP attributes to an
access-accept reply from radius. As I remember this attribute must be
declared somewhere in dictionary.openser.
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Re: [SR-Users] [SR-users] Radius AAA
radius_load_caller_avps() from misc_radius module can help you. Sorry can't find an example, but as I remember it's quite simple to use this function. Radius can insert next-hop to its Access-Accept reply and then you can use it to send an INVITE to right destination: $rd = $avp(s:next-hop); # or avp's name is equal to attribute name you received in access-accept. t_relay(); Hi again! Need tips to some tricks with radius. 1. when receive sip message with REGISTER method - initiate auth with radius. (Authentification) - radius_www_authorize in auth_radius.so 2. when receive INVITE message - ask radius 'bout permissions and parse answer: get permission and next hop IP. (Authorization) misc_radius 3. when call a) before call send Acct-Status-Type = Start with attributes. b) after call send Acct-Status-Type = Stop with attributes. (Accounting) - acc_radius.so 1,3 - clear, I understand how it works. 2 - ? Kamailio must send request to radius server, get answer and route call to server from next-hop-IP field. Can I make it? ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] What IP in REGISTER message does Kamalio use to store location ?
Save() function of the registrar module always saves Contact and can save real source to the received field of a location record. lookup() function uses Contact to form R-URI and puts ip:port from the received field to the $du. If $du is present t_relay() uses it to forward a request to a correct destination. @Jason I'm not using any NAT protocol (STUN, ICE, ...) at client. I'm using the default Kamalio config file 3.3.5. If it is the NAT helping code in Kamalio, it means that Kamalio is using 3. IP from IP packet ? On Tue, Apr 2, 2013 at 1:19 PM, Jason Penton jason.pen...@smilecoms.com mailto:jason.pen...@smilecoms.com wrote: There is probably some NAT helping code in your Kamailio config file. Either that or you are using one the NAT protocols, (STUN,ICE,etc). cheers Jason On Tue, Apr 2, 2013 at 2:39 AM, Khoa Pham onmyway...@gmail.com mailto:onmyway...@gmail.com wrote: @Jason Thanks, but it strange. I only send 1 REGISTER to the server, with the Contact field contains my local IP address. How can I still receive incoming call ? On Mon, Apr 1, 2013 at 8:10 PM, Jason Penton jason.pen...@smilecoms.com mailto:jason.pen...@smilecoms.com wrote: 2 -- This email has been sent using my phone. It may be brief, to the point or contain typos. -- On 1 Apr 2013 14:57, Khoa Pham onmyway...@gmail.com mailto:onmyway...@gmail.com wrote: Hi, when client send REGISTER message to Kamalio. Kamailio will save that client's IP to database. But what is this IP ? Is is 1,2 or 3 ? 1. The IP from Via field 2. The IP from Contact field 3. The IP from the IP packet Many thanks -- Khoa Pham HCMC University of Science Faculty of Information Technology ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users This email is subject to the disclaimer of Smile Communications (PTY) Ltd. athttp://www.smilecoms.com/disclaimer ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Khoa Pham HCMC University of Science Faculty of Information Technology ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Jason Penton Senior Manager: Applications and Services Smile Communications Johannesburg, South Africa Phone: +27 83 283 7000 This email is subject to the disclaimer of Smile Communications (PTY) Ltd. athttp://www.smilecoms.com/disclaimer ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Khoa Pham HCMC University of Science Faculty of Information Technology ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] KAMAILIO Config Hot/Live reload
On 03/25/2013 02:36 PM, Daniel-Constantin Mierla wrote: Hello, On 3/24/13 4:20 PM, Vitaliy Aleksandrov wrote: On 03/21/2013 12:59 AM, Daniel-Constantin Mierla wrote: Hello, On 3/18/13 7:09 PM, Omar wrote: Kamailio Team. Was this feature added already or we still need to restart kamailio after we modify the Kamailio.cfg. it is still required to restart after changing the config file. But some core and module parameters can be now changed at runtime, via rpc commands (e.g., lowering tcp connecting timeout). Cheers, Daniel As I understand I have to edit kamailio.cfg to preserve values changed by rpc between reboots. Is it possible to save current state of config variables somewhere (db, tmp file) ? theoretically there is a module that allows to load/reload the values from database. Unfortunately is not much documented, being inherited from SER past development. Look over it, at the end should not be impossible to get it work, we have the sources. See its readme at: - http://kamailio.org/docs/modules/stable/modules/cfg_db.html Thanks for giving me direction. If you can contribute back some documentation, will be greatly appreciated. I'll check what I can do. Cheers, Daniel ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] KAMAILIO Config Hot/Live reload
On 03/21/2013 12:59 AM, Daniel-Constantin Mierla wrote: Hello, On 3/18/13 7:09 PM, Omar wrote: Kamailio Team. Was this feature added already or we still need to restart kamailio after we modify the Kamailio.cfg. it is still required to restart after changing the config file. But some core and module parameters can be now changed at runtime, via rpc commands (e.g., lowering tcp connecting timeout). Cheers, Daniel As I understand I have to edit kamailio.cfg to preserve values changed by rpc between reboots. Is it possible to save current state of config variables somewhere (db, tmp file) ? ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] SIP debug.
On 03/18/2013 01:14 PM, Victor V. Kustov wrote: В Mon, 18 Mar 2013 11:40:24 +0100 davy van de moere davy.van.de.mo...@gmail.com write: funny, but not usable :) debug=9 loaded debugger.so but in syslog no sip debugging info What exactly you want to debug ? Wireshark produces nice sip flows from captures made by tcpdump. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] DNS error
Hi, all Have a question about failure_routes. As I remember if kamailio statefuly sent a request and didn't receive a reply script writer can handle this situation at a failure_route[] block. But if kamailio can't resolve a destination from R-URI it sends back 478 reply and finishes current transaction without reaching a failure_route[]. Is it possible to process such a request in a failure_route[] ? Additional details: version: kamailio-3.2.4 t_on_failure() is called before t_relay() I have checked with REGISTER and INVITE transactions. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Syslog
On 02/04/2013 06:07 PM, Klaus Darilion wrote: On 04.02.2013 11:26, Vitaliy Aleksandrov wrote: The problem was caused by memlog and memdbg options. So, how the problem is triggered and how did you solved it? regards Klaus Both memlog and memdbg were removed from a kamailio config as I don't need them for now. Unfortunately I didn't have a chance to spent more time on resolving this issue yet. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Syslog
The problem was caused by memlog and memdbg options. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] How to rewrite tu in CANCEL message ?
I had the same problem with CANCEL requests for transactions created by an INVITE that was rewritten in a branch_route. I sent a small patch to dev list in October and didn't receive any comment. You can try to find it in sr-dev archive for October 2012. Yes, we know it's wrong. But if we really need it, is there a way to do it? -Original Message- From: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Alex Balashov Sent: Wednesday, January 30, 2013 4:36 PM To: sr-users@lists.sip-router.org Subject: Re: [SR-Users] How to rewrite tu in CANCEL message ? On 01/30/2013 09:31 AM, Julia wrote: We need some number manipulation for outgoing calls to PSTN GW. The same manipulations must be in ru and tu, because our PSTN GW adds redirection when tu ≠ ru. When we used INVITE re-parsing for CANCEL, modified tu is not included. I would say your PSTN GW is doing it wrong. No routing should ever be done on To. It is a purely cosmetic commentary on the intended logical destination of the call. Only the RURI should be used for that. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Syslog
Hi list. I have a strange problem with syslog. Kamailio doesn't send any messages to syslog from UDP workers that are bound to sock=127.0.0.1:5060. Neither LM_DBG() from a source code nor xlog(any level, ) works. I use kamailio-3.2.4 with this opions: log_stderror=no debug=3. With log_stderror=yes kamailio writes all debug messages to stderr without any problems. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Syslog
On 02/01/2013 11:38 AM, Vitaliy Aleksandrov wrote: Hi list. I have a strange problem with syslog. Kamailio doesn't send any messages to syslog from UDP workers that are bound to sock=127.0.0.1:5060. Neither LM_DBG() from a source code nor xlog(any level, ) works. I use kamailio-3.2.4 with this opions: log_stderror=no debug=3. With log_stderror=yes kamailio writes all debug messages to stderr without any problems. I was wrong in my conclusions It looks like only the first udp listener has the described problem. children parameter was set to 1 and after it was changed to 5 UDP syslog showed debug messages that were sent by newly added workers. So the real problem in logging to a syslog from the first UDP worker regardless of interface it bound to (eth0, lo). Does anybody have any idea what can cause this problem ? ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] TLS questions
On 01/11/2013 05:03 PM, Fabian Borot wrote: Hi, we need to implement TLS in our current setup and have some questions about it: 1- We have a farm of Asterisk servers behind our kamailio proxy. The call flow is Customers/Providers -- Proxy -- Asterisks. Is it possible to have TLS only between the Customers/Providers and the Proxy and then just regular SIP over UDP between the Proxy and the Asterisks? Yes you can. 2- Can we have TLS over both TCP and UDP at the same time? Thank you a lot TLS is working only via TCP. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Regarding Kamailio Registrar
You are right after reboot registrar module takes registrations from a db. If someone send UnRegister during outage, kamailio will keep active registration until occurence of expiration time that was calculated during registration. To speed up user location cleaning you can decrease the expire value in registrations. Dear all, We are using Kamailio SIP Server in our application. We want to know 1 thing: We are using Kamailio SIP server as registrar and proxy server. For registration database, we are using dbtext module. We maintain record of SIP Extensions who have registered in Our proprietary file too. Now when we boot the Kamailio SIP server, At that time our file should be empty i.e. previously registered Extension entries should be deleted or it must be kept as it is. But if during power off, any extension had sent REGISTER request to Unregister then it was not executed as kamailio SIP server was not ON. But at boot time, as its entry was present, Kamailio SIP server will Consider it a registered extension. Please guide us for this situation. Thank you, Amit Shah. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio 3.X.X + BLF configuration trouble
On 11/27/2012 06:30 AM, Hemanshu Patel wrote: Hi vitaliy, thank you very much for your reply. Here I am not talking about kamailio as Presence server, I am asking about BLF handling For SUBSCRIBE+NOTIFY+PUBLISH in case of presence, everything is working fine In case of BFL, once the phone/user subcribes, he doesnt send any PUBLISH message, only Server sends him NOTIFY based upon the status of the user he subscribed for, like IDLE, TALKING, RINGING,etc,etc. As I understand BLF is a subscription to a dialog event. So you need a source of PUBLISH requests with dialog event them. What phone you are talking about ? Does it have an option to send its line status to a proxy via PUBLISH with a dialog event? If your phones can't send PUBLISH requests with a dialog event you can try dialog + pua_dialoginfo modules. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio 3.X.X + BLF configuration trouble
The NOTIFY without an XML body you received just after a subscription is correct. rfc3265 says: This SUBSCRIBE request will be confirmed with a final response. 200-class responses indicate that the subscription has been accepted, and that a NOTIFY will be sent immediately. You can disable such notifies via send_fast_notify presence module parameter. What about further notifications. Kamailio handles subscriptions and sends notifications when it receives PUBLISH requests. Using terms of the rfc3856 kamailio is a presence server and you need something to act as Presence User Agent. You can take a look at pua_dialoginfo module. I haven't used it, but its documentation says it can send PUBLISH requests based on active dialogs from the dialog module. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] check if the sent reply was local
||I thought t_any_replied() can be used for that purpose.
For example in failure route I used:
if (t_any_replied() t_check_status(408) ) {
xlog(L_DBG, 408 reply received\n);
}
or
if ( !(t_any_replied()) t_check_status(408)) {
xlog(L_DBG, Local timeout\n);
}
But I carefully read tm docs and found If called from a failure or
onreply route, the __current__ reply is not taken into account.
According to t_any_replied() description my examples should not work,
but they do (at production system for about one year).
On 11/11/12 1:54 PM, Uri Shacked wrote:
Hi,
what is the command to check if the reply was recieved or localy
generated?
I assume you need that in failure_route, test the $T(reply_type), it
should be 1 for local generated replies:
http://www.kamailio.org/wiki/cookbooks/3.3.x/pseudovariables#t_name
Cheers,
Daniel
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Re: [SR-Users] db_sqlite bugs
Hello,
Have you found the way to run SELECT COUNT(*) ?
I use kamailio 3.2.4 with sqlite 3.7.5-1 and it crashes on such queries too.
As a workaround I use $dbr(res=rows) with SELECT some_field FROM
table query, but the bug is really exists.
Hello Timo,
We've done a basic test and it seems to solve the memory leak problem.
We'll do exhaustive tests in a few days.
We'll keep you informed.
Thanks a lot,
Pedro
De: Timo Teras timo.te...@iki.fi
Para: SIP Router - Kamailio \(OpenSER\) and SIP Express Router
\(SER\) - Users Mailing List sr-users@lists.sip-router.org
cc: pvsol...@amper.es
Fecha: 15/10/2012 16:13
Asunto: Re: [SR-Users] db_sqlite bugs
Enviado por: Timo Teräs timo.te...@gmail.com
On Thu, 11 Oct 2012 12:37:23 +0200 Pedro Antonio Vico Solano
pvsol...@amper.es wrote:
Hello everyone,
We are developing a solution based on the SQLite module for Kamailio
and we've found some bugs/errors regarding it. We use kamailio 3.2.4
and SQLite 3.7.14.1. The errors are about this:
- Memory leaks on querys via sqlops. Kamailio main process leaks 4kB
each 3 queries (aprox.) and that memory is never freed. [Attached
basic example]
- Segmentation fault when doing a SELECT COUNT(*) via sqlops.
[Attached basic example 2]
Can you help us?
Does this help for the memory leak?
diff --git a/modules_k/db_sqlite/dbase.c b/modules_k/db_sqlite/dbase.c
index 0b32e07..c1d3b71 100644
--- a/modules_k/db_sqlite/dbase.c
+++ b/modules_k/db_sqlite/dbase.c
@@ -543,7 +543,12 @@ int db_sqlite_update(const db1_con_t* _h, const
db_key_t* _k, const db_op_t* _o,
int db_sqlite_raw_query(const db1_con_t* _h, const str* _s,
db1_res_t** _r)
{
- return db_do_raw_query(_h, _s, _r,
+ int rc;
+
+ rc = db_do_raw_query(_h, _s, _r,
db_sqlite_submit_query,
db_sqlite_store_result);
+ db_sqlite_cleanup_query(_h);
+
+ return rc;
}
Seems that all other database driver postpone the resource release to
free_results - but there's no reason why we couldn't do this right away
in sqlite. Since in sqlite free_results we just free cached result set.
Will take a look at the other bug later when I get the time to
reproduce it.
-Timo
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Re: [SR-Users] Message Routing To/From Gateway
If you want to receive any suggestions it would be better to show you
config file with some comments.
It will reduce a time to understand you configuration and will increase
a chance to get any help here.
After further testing it was found that setting the alias like this
somehow broke outgoing messages. I'm not certain exactly how Kamailio
handles aliases, but it seems like it is not as simple as I expected.
I am working on figuring out a new solution now if anyone has any
thoughts.
Thanks,
Ed
On Wed, Oct 24, 2012 at 12:27 AM, Ed Brady ebr...@talkforgood.com
mailto:ebr...@talkforgood.com wrote:
I was able to resolve the issue by setting the return domain as a
local domain alias. With the previous setting I had made to
forward all non-local accounts to our gateway domain, this would
allow the improper return domain to be seamlessly switched to the
correct domain before being forwarded out of our server. Now users
can have normal two way conversations despite a poor design by our
vendor.
Thanks for a giant push in the right direction!
Ed
On Mon, Oct 22, 2012 at 4:02 PM, Ed Brady ebr...@talkforgood.com
mailto:ebr...@talkforgood.com wrote:
I have this all working now. I had previously configured this
as a gateway and once issues were fixed on our vendor's end,
it worked. Is there a disadvantage to using this method rather
than the permissions module?
On a related note (though maybe not related enough), when we
get responses from their system it comes back with a different
domain in the URI than what they require us to send to. When
questioned about this they said we would need to handle it on
our end. Is there a common method for checking the to URI for
a particular string (in this case the domain), changing it,
and then forwarding it?
Thanks,
Ed
On Sat, Oct 20, 2012 at 7:23 PM, Vitaliy Aleksandrov
vitalik.v...@gmail.com mailto:vitalik.v...@gmail.com wrote:
Permissions module should help you. If allowed ip
addresses are kept in a database you can reload them by MI
command without kamailio restart.
As I remember permissions module allows to create groups
of ip addresses, so you can create such a group for hosts
which you want to receive MESSAGE requests from.
I tested this example and it worked for me. I was
misunderstanding some of the values and thought they
applied to users being registered or not.
I am still uncertain what setting applies to allowing all
traffic inbound from a particular address. Any tips on
this would be appreciated.
Thanks,
Ed
On Fri, Oct 19, 2012 at 1:41 PM, Ed Brady
ebr...@talkforgood.com mailto:ebr...@talkforgood.com
wrote:
I was looking at this logic some more and wanted to
clarify my configuration. I need to base the R-URI
rewrite and MESSAGE forward based on whether the
account exists as a local subscriber or not rather
than whether the user is registered or not. This is
connecting to an SMS provider using SIP.
I excitedly looked into the SMS module but this
appears to only be if you are dealing with a GSM
modem and not sending/receiving SIP MESSAGEs with an
SMS gateway.
Any resources you can point me to would be extremely
helpful, thanks!
Ed
On Fri, Oct 19, 2012 at 10:42 AM, Ed Brady
ebr...@talkforgood.com
mailto:ebr...@talkforgood.com wrote:
Will this also permit incoming messages from this
domain? Do I need to add the gateway in any other
settings besides the one you outlined?
Thanks,
Ed
On Fri, Oct 19, 2012 at 6:17 AM, Vitaliy
Aleksandrov vitalik.v...@gmail.com
mailto:vitalik.v...@gmail.com wrote:
If I understood you right you just need to
rewrite R-URI domain and forward MESSAGE if a
user in not registered.
if (!lookup(location)) {
switch($retcode) {
case -1:
$rd = gatewaydomain.com
http://gatewaydomain.com;
t_relay();
exit;
default:
sl_send_reply(404, Not Found
Re: [SR-Users] Message Routing To/From Gateway
Permissions module should help you. If allowed ip addresses are kept
in a database you can reload them by MI command without kamailio restart.
As I remember permissions module allows to create groups of ip
addresses, so you can create such a group for hosts which you want to
receive MESSAGE requests from.
I tested this example and it worked for me. I was misunderstanding
some of the values and thought they applied to users being registered
or not.
I am still uncertain what setting applies to allowing all traffic
inbound from a particular address. Any tips on this would be appreciated.
Thanks,
Ed
On Fri, Oct 19, 2012 at 1:41 PM, Ed Brady ebr...@talkforgood.com
mailto:ebr...@talkforgood.com wrote:
I was looking at this logic some more and wanted to clarify my
configuration. I need to base the R-URI rewrite and MESSAGE
forward based on whether the account exists as a local subscriber
or not rather than whether the user is registered or not. This is
connecting to an SMS provider using SIP.
I excitedly looked into the SMS module but this appears to only be
if you are dealing with a GSM modem and not sending/receiving SIP
MESSAGEs with an SMS gateway.
Any resources you can point me to would be extremely helpful, thanks!
Ed
On Fri, Oct 19, 2012 at 10:42 AM, Ed Brady ebr...@talkforgood.com
mailto:ebr...@talkforgood.com wrote:
Will this also permit incoming messages from this domain? Do I
need to add the gateway in any other settings besides the one
you outlined?
Thanks,
Ed
On Fri, Oct 19, 2012 at 6:17 AM, Vitaliy Aleksandrov
vitalik.v...@gmail.com mailto:vitalik.v...@gmail.com wrote:
If I understood you right you just need to rewrite R-URI
domain and forward MESSAGE if a user in not registered.
if (!lookup(location)) {
switch($retcode) {
case -1:
$rd = gatewaydomain.com
http://gatewaydomain.com;
t_relay();
exit;
default:
sl_send_reply(404, Not Found);
exit;
}
}
I am trying to route SIP messages to a gateway. I am
using SIP MESSAGE over UDP as well as the MSILO module
for storing messages for users that aren't currently
logged in. I have read through the Dynamic Routing module
info but am not able to determine the best way to
accomplish the routing I require. Our scenario is
outlined below.
User A sends a message addressed to
sip:12345678...@mydomain.com
mailto:sip%3a12345678...@mydomain.com
Kamailio at mydomain.com http://mydomain.com checks if
12345678900 tel:12345678900 is a registered user in the
system.
If not, the message header is translated to
sip:12345678...@gatewaydomain..com
mailto:sip:12345678...@gatewaydomain..com and forwarded.
Incoming messages need to be accepted from their IP as well.
It would also be useful if the address came as
sip:2345678...@mydomain.com
mailto:sip%3a2345678...@mydomain.com could be
translated to sip:_1_2345678...@gatewaydomain.com
mailto:2345678...@gatewaydomain.com.
Is there any built in functionality I can configure for
this, or will this need to be scripted?
Thanks,
Ed
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Re: [SR-Users] Kamailio 3.2.4 presence
Readme at the link stated below said that db_mode is a valid parameter.
http://kamailio.org/docs/modules/3.2.x/modules_k/presence.html#id2546358
This patch fixes subscription restoration after kamailio-3.2.4 restart
in case when db_mode of presence module is 0 (MEMORY_ONLY).
vitalik@work:~/src$ diff modules_k/presence/hash.c modules_k/presence/hash.c
264c264
if(dbmode == DB_FALLBACK || dbmode == DB_MEMORY_ONLY) {
---
if(dbmode == DB_FALLBACK) {
Now it saves subscriptions in memory and periodically(db_update_period
parameter) dumps them to the active_watchers table.
After restart kamailio fills subscription hash table from the
active_watchers db table.
Upgrade is a good suggestion as presence module of 3.3.х branch differs
much from the one in the previous release and I hope it doesn't have
described problems..
I have to port a few custom modules to switch to the new branch.
Vitaliy Aleksandrov writes:
db_mode is 0 (DB_MEMORY_ONLY) and according to comments in the
presence.h kamailio holds subscription in memory and periodically
updates to db, but retrieves from db only at startup.
And it's true, I see that kamailio tries to update the active_watchers
table, but it never inserts new subscriptions there.
i suggest you upgrade to 3.3. according to its readme, subs_db_mode=0
means that db is disabled. perhaps it was the same with 3.2, which
according to its readme does not have db_mode param.
-- juha
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[SR-Users] Kamailio 3.2.4 presence
Hi all, I'm facing problems with active subscription restoration after sipproxy restart. db_mode is 0 (DB_MEMORY_ONLY) and according to comments in the presence.h kamailio holds subscription in memory and periodically updates to db, but retrieves from db only at startup. And it's true, I see that kamailio tries to update the active_watchers table, but it never inserts new subscriptions there. There are no problems with access to the database since when db_mode is 2 (DB_ONLY) kamailio inserts new records into the active_watchers without any problems. Here are presence module settings: modparam(presence, db_url, sqlite:var/db/presence.db) modparam(presence, db_mode, 0) modparam(presence, timeout_rm_subs, 1) modparam(presence, pres_htable_size, 8) modparam(presence, subs_htable_size, 8) modparam(presence, max_expires, 3600) modparam(presence, db_update_period, 60) modparam(presence, clean_period, 60) Does anybody have success with db_mode = DB_MEMORY_ONLY and active_watchers db table ? ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Locate UA in usrloc table
Hi all. Is there any easy method to find if somebody with $si:$sp is registered (usrloc-received.ip == $si, usrloc-received.port == $sp) ? reg_fetch_contacts uses uri as search key, but in my case I don't have it. The only way I know is to write a module that will use get_all_ucontacts function exported by usrloc module in its API. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] [ ... ] operator
Hello, On 9/6/12 6:33 PM, Vitaliy Aleksandrov wrote: Migrating kamailio v1.5.x to v.3.0.0 says that test operator no longer exists. But it is mentioned in Core Cookbook 3.2 (and 3.3) and can confuse someone. Can somebody remove it from the documentation ? Indeed the operator was removed, same functionality can be obtained by simply removing the [ ]. You can create an account and update it yourself, captcha-based anonymous editing was disable due to spam bots going easy through the system. https://www.kamailio.org/wiki/start?do=register Cheers, Daniel Yes it works without [ ]. I didn't know that anyone can update documentation without any pre moderation. Wiki is updated. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] [ ... ] operator
Migrating kamailio v1.5.x to v.3.0.0 says that test operator no longer exists. But it is mentioned in Core Cookbook 3.2 (and 3.3) and can confuse someone. Can somebody remove it from the documentation ? ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Why append_branch should be called in a failure route ?
Below is the example from http://kamailio.org/docs/modules/3.2.x/modules/tm.htm
:
failure_route[0]{
if (t_check_status(5[0-9][0-9])){
# I do not like the 5xx responses,
# so I give another chance to foobar.com,
# and I drop all the replies to make sure that
# they are not forwarded to the caller.
t_drop_replies();
rewritehostport(foobar.com);
append_branch();
t_relay();
}
}
Why do append_branch() is present before t_relay().
I'm asking because I didn't used it in failure routes and call rerouting on
various conditions worked great.
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[SR-Users] AVP string or integer ID
Hi all. I'm working with kamailio for about 1 year and still don't understand difference between integer and string IDs of AVPs. Is it just a question of usability ? ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Request for advice, running SIP-Router in OpenVZ Container
On 08/17/2012 03:42 PM, JR Richardson wrote: Hi All, I'm considering running Kamailio as a virtual machine, with such low utilization, it doesn't seem to make sense to keep running it on a physical host server. I've been virtualizing Asterisk PBX's for years and run a host of other virtualized servers with OpenVZ, VMware and MS V-Server without issues. I'd like any opinions or thoughts about pitfalls or caveats I should be aware of specific to virtualized SIP-Router/Kamailio. Thanks in advanced. JR ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users It works great in VMware and XEN VMs without any issues. In my experience i had only two problems with virtual environment: - I/O (network, disc). - timing precision. Disc I/O can be a problem if you are going to write lots of logs and the VM running kamailio are connected to overloaded disc array or external storage. Use only async syslog mode otherwise kamailio process will be blocked in log operation and will be not available to process incoming sip traffic. So use fast drives and try to avoid time consuming operations (syslog, db queries, radius, etc...) during message processing as they block current process and kamailio has fixed number of running processes. From my point of view timing precision is not a problem for a sip-proxy as it's not a big deal to send some re-transmit few milliseconds earlier/later. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] ACK and BYE messages uses wrong socket.
Have you specified interfaces with listen command ? I had a problem as you described and have fixed it by moving a listen directive with a floating ip to the top of the list. So you can try to specify interfaces you will use for SIP and set a virtual ip at the top of that list. Kamailio 3.2.0 I am trying to setup kamailio to do the sip trunking. It receive the sip traffic from customer and then send it to carrier. I have two NIC interface's assigned with three IP's: Interface 1: ( Public IP's) x.x.130.34 x.x.130.36 floating IP interface 2: (private IP's) 10.10.1.31 .36 is a floating IP assigned by Linux-HA (heartbeat/pacemaker). I only want to use .36 to receive and send sip traffic. I uses force_send_socket() to send INVITE with .36 IP. But the ACK message always want to use .34 IP even the Route header has .36 in it unless I force it with force_send_socket() . How can I fix this problem? See below for the SIP messages: (x.x.128.205 is customer IP, x.x.129.200 is PSTN gateway IP) U x.x.128.205:51694 - x.x.130.36:5060 INVITE sip:5033441174@x.x.130.36:5060 SIP/2.0..Via: SIP/2.0/UDP x.x.128.205:5060;branch=z9hG 4bK1D3CD1..From: sip:5024427578@x.x.128.205;tag=24513088-D59..To: sip:5033441174@x.x.130.3 6..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.20 5..Supported: 100rel,timer,replaces..Min-SE: 1800..Cisco-Guid: 411443261-3293254113-3191264919-256 0877466..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, CO MET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..CSeq: 101 INVITE..Max-Forwards: 70..Remote-P arty-ID: sip:5024427578@x.x.128.205;party=calling;screen=no;privacy=off..Timestamp: 1341322661 ..Contact: sip:5024427578@x.x.128.205:5060..Expires: 180..Allow-Events: telephone-event..Conte nt-Type: application/sdp..Content-Length: 366v=0..o=CiscoSystemsSIP-GW-UserAgent 9094 579 IN IP 4 x.x.128.205..s=SIP Call..c=IN IP4 x.x.128.205..t=0 0..m=audio 19312 RTP/AVP 125 0 18 100 10 1..c=IN IP4 x.x.128.205..a=rtpmap:125 X-CCD/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:18 G729/8000..a =fmtp:18 annexb=yes..a=rtpmap:100 X-NSE/8000..a=fmtp:100 192-194..a=rtpmap:101 telephone-event/8000 ..a=fmtp:101 0-16.. # U x.x.130.36:5060 - x.x.128.205:51694 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP x.x.128.205:5060;branch=z9 hG4bK1D3CD1;rport=51694..From: sip:5024427578@x.x.128.205;tag=24513088-D59..To: sip:502244117 4@x.x.130.36..Call-ID: 1887588d-c44b11e1-be38d697-98a3e...@x.x.128.205..cseq: 101 INVITE..Se rver: LVS Proxy 1.0..Content-Length: 0 # U x.x.130.36:5060 - x.x.129.200:5060 INVITE sip:15033441174@x.x.129.200:5060 SIP/2.0..Record-Route: sip:x.x.130.36;lr=on..Via: S IP/2.0/UDP x.x.130.36;branch=z9hG4bKc2ce.17a955a3.0..Via: SIP/2.0/UDP x.x.128.205:5060;rport =51694;branch=z9hG4bK1D3CD1..From: sip:5024427578@x.x.128.205;tag=24513088-D59..To: sip:50224 41174@x.x.130.36..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3 E39A@x.x.128.205..Supported: 100rel,timer,replaces..Min-SE: 1800..Cisco-Guid: 411443261-3293254 113-3191264919-2560877466..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANC EL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..CSeq: 101 INVITE..Max-Forw ards: 69..Remote-Party-ID: sip:5024427578@x.x.128.205;party=calling;screen=no;privacy=off..Tim estamp: 1341322661..Contact: sip:5024427578@x.x.128.205:51694..Expires: 180..Allow-Events: tel ephone-event..Content-Type: application/sdp..Content-Length: 375v=0..o=CiscoSystemsSIP-GW-UserA gent 9094 579 IN IP4 10.200.1.51..s=SIP Call..c=IN IP4 10.200.1.51..t=0 0..m=audio 20464 RTP/AVP 12 5 0 18 100 101..c=IN IP4 10.200.1.51..a=rtpmap:125 X-CCD/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:18 G7 29/8000..a=fmtp:18 annexb=yes..a=rtpmap:100 X-NSE/8000..a=fmtp:100 192-194..a=rtpmap:101 telephone- event/8000..a=fmtp:101 0-16..a=nortpproxy:yes.. U x.x.129.200:5060 - x.x.130.36:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP x.x.130.36;branch=z9hG4bKc2ce.17a955a3.0,SIP/2.0/UDP 216.4 9.128.205:5060;rport=51694;branch=z9hG4bK1D3CD1..From: sip:5024427578@x.x.128.205;tag=24513088 -D59..To: sip:5033441174@x.x.130.36;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:41 GMT..Cal l-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..Timestamp: 1341322661..Server: Cisco-SIPG ateway/IOS-12.x..CSeq: 101 INVITE..Allow-Events: telephone-event..Content-Length: 0 # U x.x.129.200:5060 - x.x.130.36:5060 SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP x.x.130.36;branch=z9hG4bKc2ce.17a955a3.0,SIP/2.0/ UDP x.x.128.205:5060;rport=51694;branch=z9hG4bK1D3CD1..From: sip:5024427578@x.x.128.205;ta g=24513088-D59..To: sip:5033441174@x.x.130.36;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:4 1
[SR-Users] cfgutils lock / unlock + $sht() variables.
Hi all, I use $sht() pseudo-variabled provided by htable module for caching various information about sip customers. I thought that htable has some locking mechanism inside it and today while i was reading core-cookbook i payed attention to one example with the lock/unlock functions and $sht(): lock(calls-to::10.10.10.10); $sht(a=calls-to::10.10.10.10) = $sht(a=calls-to::10.10.10.10) + 1; unlock(calls-to::10.10.10.10); According to that example an explicit locking mechanism should be used when i want to write something to a hash table. On the other hand there are no locks in htable documentation. So the question is when do the explicit locks should be used and for what purpose ? ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] default kamailio configuration.
I have lost in Route and Record-Route headers. But now it's clear for me. Huge thanks. On 25.06.2012 14:44, Vitaliy Aleksandrov wrote: Hi All, After reading default kamailio configuration i can't understand why does kamailio remove preloaded route headers from the incoming initial INVITE before calling record_route(). remove_hf(Route); if (is_method(INVITE|SUBSCRIBE)) record_route(); Pre-loaded route sets can be used to abuse the proxy as an open relay. This is bad. If a request hits my proxy, then I want to define how the message is routed, and I do not want to route the message like somebody else wants it. What will happen if an INVITE comes from another SIP proxy which wants to stay in the middle of the dialogue ? Only Route headers are removed, not Record-Route headers which create the route-set. For example, during Re-INVITE UA from our local site will construct in-dialog INVITE with R-URI taken from the remote contact and send it to our local proxy according to RR. Local proxy will process it in route[WITHNDLG] where loose_route() function will remove top most RR header because it points to us and since there is no more RR records forward the request to a destination from the R-URI. As i understand in that case remote proxy will not receive any in-dialog requests from us. Sure it does. As only the pre-loaded route is remove, not the Record-Route headers. Thus, the in-dialog route set is not affected, only the out-of-dialog route set (pre-loaded) is manipulated. regards Klaus ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] default kamailio configuration
Hi All, After reading default kamailio configuration i can't understand why does kamailio remove preloaded route headers from the incoming initial INVITE before calling record_route(). remove_hf(Route); if (is_method(INVITE|SUBSCRIBE)) record_route(); What will happen if an INVITE comes from another SIP proxy which wants to stay in the middle of the dialogue ? For example, during Re-INVITE UA from our local site will construct in-dialog INVITE with R-URI taken from the remote contact and send it to our local proxy according to RR. Local proxy will process it in route[WITHNDLG] where loose_route() function will remove top most RR header because it points to us and since there is no more RR records forward the request to a destination from the R-URI. As i understand in that case remote proxy will not receive any in-dialog requests from us. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] multiple t_on_branch[]
Hi all, Does kamailio allow to configure more than one branch_route for an initial INVITE ? I have tried to add: t_on_branch(branch_r1); t_on_branch(branch_r2); before t_relay(). Mentioned branch routes write debug to the kamailio.log and it looks like that only the last branch_route[branch_r2] works. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Max registered users
Hi all, I'm trying to find a way to get a number of the registered user locations and if it is bigger that some limit then kamailio will not save new registrations. reg_fetch_contacts() from registrar module doesn't help as it can fetch contacts only for one user. max_contacts parameter of the registrar module limits the number of contacts per AOR, but i need to limit max number of all contacts that proxy will keep in usrloc. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Creating RURI ($ru) from Contact ($ct)
On 05/26/2012 05:39 PM, Aft nix wrote:
Hi,
I'm trying to create a ruri from contact header like following :
In request route I've saved contact header in a hashtable using htable module.
request_route{
-
-
#!ifdef WITH_HASH
if (is_method(INVITE) !has_totag()){
xdbg (Contact [$ct] with Callid [$ci]);
$sht(a=$ci) = $ct;
}
#!endif
}
And in route[WITHINDLG]
route[WITHINDLG] {
---
---
#!ifdef WITH_HASH
xdbg (ruri [$ru]);
if ($sht(a=$ci) != $ru){
$ru = ($sht(a=$ci));
xdbg (rewriting ruri to [$ru]);
}
#!endif
But it log shows that its unable to parse the uri :
[4039]: DEBUG:core [parser/parse_uri.c:1277]: parse_uri: bad uri,
state 0 parsed:sip (4) /sip:2020@103.23.169.2:64358 (29)
: DEBUG:core [parser/parse_uri.c:1327]: ERROR: parse_sip_msg_uri:
bad urisip:2020@103.23.169.2:64358
[4039]: ERROR: pv [pv_core.c:232]: failed to parse the R-URI
[4039]: DEBUG:script: rewriting ruri to [null]
I'm not understanding what's wrong with that. May be the port part
should be discarded, but how i'm gonna do that?
It looks like $sht(a=$ci) is empty. I see that you have xdbg (Contact
[$ct] with Callid [$ci]); line, but you didn't show us that line at
the debug.
If you add $ci to the debug information at the route[WITHINDLG] and show
full debug from INVITE to BYE it will help to find where the mistake is.
And i think it would be better if you change _if ($sht(a=$ci) != $ru)_
to _if ( $sht(a=$ci) != $null $sht(a=$ci) != $ru)_ , because when
$sht() is empty it always not equal to $ru.
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Re: [SR-Users] Hop by hop CANCEL, wrong to header
Hello Daniel, Thanks for the reply. Sorry i have made a typo. If E2E_CANCEL_HOP_BY_HOP is defined then e2e_cancel() from t_fwd.c calls cancel_branch() function for each branch it found at a t_invite transaction. Then if cfg_get(tm, tm_cfg, reparse_invite) is true cancel_branch() calls build_local_reparse() which returns new CANCEL request buffer based on outgoing INVITE, but with original To header. I've searched a bit deeper and found that if E2E_CANCEL_HOP_BY_HOP is not defined e2e_cancel() calls e2e_cancel_branch(), which then calls build_local_reparse() like in previous description when E2E_CANCEL_HOP_BY_HOP was defined. So the main question is why does kamailio take To field from a transaction cell and not from outgoing buffer like it does with other headers ? I'm using kamailio-3.0.2, but i have checked 3.2.2 sources and haven't found any changes at the functions mentioned above. Cheers, Vitaliy Aleksandrov Hello, On 5/23/12 5:22 PM, Vitaliy Aleksandrov wrote: Hi all, I have a question about CANCEL message processing. My call sceraio: When an INVITE request comes I need to rewrite a domain part of the several headers(to, from, contact, SDP ip) according to the outgoing interface. I can do it before t_relay(), but when destination user has more then one locations and they are reachable through different interfaces all forked INVITEs will have the same domain. To avoid that problem i have tried to move rewriting part(subst from the textops module) to the branch_route. Unfortunately that solution didn't helped me, because To header of the outgoing CANCEL messages is wrong (unchanged. as it was at the moment when transaction was created by the t_relay() ). As i understood from the documentation and mailing list kamailio builds CANCEL based on outgoing INVITE. I did a small research and found that Kamailio really takes outgoing INVITE from each branch (invite_transaction_cell - uac[b_id].request.buffer) and builds CANCEL (build_local_reparse() from tm/t_msgbuilder.c) based on it. But kamailio does an exception for the To header which is described below: ***cancel_branch* function (from t_cancel.c) calls *build_local_reparse*() and fills one of the parameters with a pointer to unmodified To header. *build_local_reparse*() uses received To header to construct outgoing CANCEL. I have changed: cancel = build_local_reparse(t, branch, len, CANCEL, CANCEL_LEN, t-to, reason); to: cancel = build_local_reparse(t, branch, len, CANCEL, CANCEL_LEN, NULL, reason); and it works for me now. It looks like if E2E_CANCEL_HOP_BY_HOP *e2e_cancel()* (t_fwd.c) will call *e2e_cancel_branch()* which works as i want instead of *cancel_branch()*, but it is just my assumption. do you mean if E2E_CANCEL_HOP_BY_HOP is defined? I quick grep at this time showed it is defined in t_fwd.h... What is the version you are using? Cheers, Daniel Why does kamailio generate CANCEL requests in such a way ? Did i miss something from the RFC3261 or kamailio documentation ? Thanks in advance for any help ! ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -http://www.asipto.com http://twitter.com/#!/miconda -http://www.linkedin.com/in/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] kamailioi is not relaying BYE message to UAC
Thanks for the reply.
We have already came to the same conclusion by some testing in our
lab. It seems its a bug in provider which not constructing BYE message
properly.
But i'm interested in if its possible to detect the fault in this BYE
and construct a new one and then relay it to the UAC.
I mean can i do this :
contact-header = INVITE's contact-header
if (contact-header != BYE's ruri)
{
construct BYE message with contact header
t_relay()
}
Cheers
aft
Hi aft,
I think to do what you want you can save Contact field + callid pair
taken from the INVITEs that comes from your side.
Then when BYE comes from such provider you should find a correct Contact
for that call-id and if it exists and not equal to R-URI you can rewrite it.
For example to write to R-URI value from $var(correct_ruri) you can use
*$ru = $var(correct_ruri);* statement.
What about a place where contact can be saved, i think htable will be
the nice one.
Please pay attention to autoexpire parameter of htable module. You
should take care about the staled records to avoid memory usage problems.
Cheers,
Vitaliy Aleksandrov
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[SR-Users] Hop by hop CANCEL, wrong to header
Hi all, I have a question about CANCEL message processing. My call sceraio: When an INVITE request comes I need to rewrite a domain part of the several headers(to, from, contact, SDP ip) according to the outgoing interface. I can do it before t_relay(), but when destination user has more then one locations and they are reachable through different interfaces all forked INVITEs will have the same domain. To avoid that problem i have tried to move rewriting part(subst from the textops module) to the branch_route. Unfortunately that solution didn't helped me, because To header of the outgoing CANCEL messages is wrong (unchanged. as it was at the moment when transaction was created by the t_relay() ). As i understood from the documentation and mailing list kamailio builds CANCEL based on outgoing INVITE. I did a small research and found that Kamailio really takes outgoing INVITE from each branch (invite_transaction_cell - uac[b_id].request.buffer) and builds CANCEL (build_local_reparse() from tm/t_msgbuilder.c) based on it. But kamailio does an exception for the To header which is described below: ***cancel_branch* function (from t_cancel.c) calls *build_local_reparse*() and fills one of the parameters with a pointer to unmodified To header. *build_local_reparse*() uses received To header to construct outgoing CANCEL. I have changed: cancel = build_local_reparse(t, branch, len, CANCEL, CANCEL_LEN, t-to, reason); to: cancel = build_local_reparse(t, branch, len, CANCEL, CANCEL_LEN, NULL, reason); and it works for me now. It looks like if E2E_CANCEL_HOP_BY_HOP *e2e_cancel()* (t_fwd.c) will call *e2e_cancel_branch()* which works as i want instead of *cancel_branch()*, but it is just my assumption. Why does kamailio generate CANCEL requests in such a way ? Did i miss something from the RFC3261 or kamailio documentation ? Thanks in advance for any help ! ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Copy reason field from 503 to 500.
According to
http://www.asipto.com/pub/kamailio-devel-guide/#c02message_processing;
all replies comes to on_reply and if it is negative reply it will be
processed in failure_route after onreply_route.
But that is not the point i tried failure_route too.
append_hf works great for forwarded replies. But as i understand when
kamailio receives final 503 reply it sends 500 reply generated by
build_res_buf_from_sip_req (it is part of tm code, not dialplan function).
Thanks for your tries to help.
||Well 5xx error codes are processed in failure_route and not
onreply_route. Not sure though if you can append_hf there.. Give it a try.
FYI, for using a function in event_route (seems not possible for
append_hf):
event_route[tm:local-request] {
...
append_hf(...);
...
}
Reda
On 26 avr. 2012, at 18:10, Vitaliy Aleksandrov vitalik.v...@gmail.com
mailto:vitalik.v...@gmail.com wrote:
I have tried to use append_hf in onreply_route.
How can i use it in local-request ? According to documentation it can
be used from REQUEST_ROUTE, ONREPLY_ROUTE, FAILURE_ROUTE, BRANCH_ROUTE.
Sorry forgot to mention that kamailio version is 3.0.2, but i have
looked at t_repy.c (tm module from 3.2) and didn't find any changes
which can help me.
Hi,
In which route did you try append_hf ?
event_route[tm:local-request] is the route executed for locally
generated requests, I think you should be trying to modify it there.
Reda
On Thu, Apr 26, 2012 at 16:10, Vitaliy Aleksandrov
vitalik.v...@gmail.com mailto:vitalik.v...@gmail.com wrote:
Hi all,
I have Kamailio connected to PSTN gateway.
Sometimes gateway replies with 503 temporary unavailable. PSTN
gateway adds a reason header with ISUP code.
But according to RFC 3261 tel:3261 kamailio doesn't forward
503 replies transparently. When 503 reply is received Kamailio
replaces 503 with a 500 which is created from original request
(build_res_buf_from_sip_req).
I have tried append_hf from textops module, but it's not working
with a locally generated replies.
How can i copy Reason header from 503 to 500 ?
I will appreciate your help with this situation.
Best regards,
Vitaliy Aleksandrov.
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[SR-Users] Copy reason field from 503 to 500.
Hi all, I have Kamailio connected to PSTN gateway. Sometimes gateway replies with 503 temporary unavailable. PSTN gateway adds a reason header with ISUP code. But according to RFC 3261 kamailio doesn't forward 503 replies transparently. When 503 reply is received Kamailio replaces 503 with a 500 which is created from original request (build_res_buf_from_sip_req). I have tried append_hf from textops module, but it's not working with a locally generated replies. How can i copy Reason header from 503 to 500 ? I will appreciate your help with this situation. Best regards, Vitaliy Aleksandrov. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Copy reason field from 503 to 500.
I have tried to use append_hf in onreply_route. How can i use it in local-request ? According to documentation it can be used from REQUEST_ROUTE, ONREPLY_ROUTE, FAILURE_ROUTE, BRANCH_ROUTE. Sorry forgot to mention that kamailio version is 3.0.2, but i have looked at t_repy.c (tm module from 3.2) and didn't find any changes which can help me. Hi, In which route did you try append_hf ? event_route[tm:local-request] is the route executed for locally generated requests, I think you should be trying to modify it there. Reda On Thu, Apr 26, 2012 at 16:10, Vitaliy Aleksandrov vitalik.v...@gmail.com mailto:vitalik.v...@gmail.com wrote: Hi all, I have Kamailio connected to PSTN gateway. Sometimes gateway replies with 503 temporary unavailable. PSTN gateway adds a reason header with ISUP code. But according to RFC 3261 tel:3261 kamailio doesn't forward 503 replies transparently. When 503 reply is received Kamailio replaces 503 with a 500 which is created from original request (build_res_buf_from_sip_req). I have tried append_hf from textops module, but it's not working with a locally generated replies. How can i copy Reason header from 503 to 500 ? I will appreciate your help with this situation. Best regards, Vitaliy Aleksandrov. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@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@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] T1 timer
Yes, i tried to call t_set_retr from route block before t_relay. On 10/27/11 6:15 PM, Vitaliy Aleksandrov wrote: Thank you for your quick reply. t_set_retr doesn't affects transaction created by uac_req_send. The second way proposed by you works great (tm settings for uac, t_set_retr for t_relay). Just to be clear, you have tested t_set_retr only in route block, not in event_route[tm:local-request], right? Cheers, Daniel Hello, On 10/26/11 12:10 PM, Vitaliy wrote: Hello, How can i change T1 timeout for transaction created by the UAC module (uac_req_send). Do i have to call t_set_retr before uac_req_send and then t_reset_retr() before t_relay ? I use uac_req_send to send accounting to the remote server and i want to increase retransmit timeout for such signalling messages. it is not a way dedicated for uac_req_send() and I haven't played with variants -- you can try what you wrote above and if does not work, try to set the timer parameters for tm to the one you want for uac and use for forwarded transactions t_set_retr. One other way to try is to make a event_route[tm:local-request] and use there the t_set_retr. Let me know if any of it works, we can add to docs if so, otherwise I will look for implementing a solution for it. Cheers, Daniel ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] T1 timer
Thank you for your quick reply. t_set_retr doesn't affects transaction created by uac_req_send. The second way proposed by you works great (tm settings for uac, t_set_retr for t_relay). Hello, On 10/26/11 12:10 PM, Vitaliy wrote: Hello, How can i change T1 timeout for transaction created by the UAC module (uac_req_send). Do i have to call t_set_retr before uac_req_send and then t_reset_retr() before t_relay ? I use uac_req_send to send accounting to the remote server and i want to increase retransmit timeout for such signalling messages. it is not a way dedicated for uac_req_send() and I haven't played with variants -- you can try what you wrote above and if does not work, try to set the timer parameters for tm to the one you want for uac and use for forwarded transactions t_set_retr. One other way to try is to make a event_route[tm:local-request] and use there the t_set_retr. Let me know if any of it works, we can add to docs if so, otherwise I will look for implementing a solution for it. Cheers, Daniel ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
