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] Wrong dialog selected in on_reply/failure route in case of spirals.
Hello Daniel, Today I installed kamailio from GIT last devel 4.3.0-dev3. *The problem is solved!* Thank you so much for your help ! Julia On Mon, Jan 26, 2015 at 8:38 PM, Julia Boudniatsky juli...@gmail.com wrote: Yes it /usr/local/sbin/kamailio, I haven't internet in the test server. I get git from another server and copy received directory kamailio to test server, then make cfg/all/install. BR, Julia On Mon, Jan 26, 2015 at 8:10 PM, Daniel-Constantin Mierla mico...@gmail.com wrote: I did the same right now and I get: version: kamailio 4.3.0-dev3 (x86_64/darwin) 1b334f Can you check if you have another instance installed on a different path that takes precendence? Do: which kamailio When installed from sources, it should be: /usr/local/sbin/kamailio Cheers, Daniel On 26/01/15 18:55, Julia Boudniatsky wrote: I used from your link * http://www.kamailio.org/wiki/#installation - Install Kamailio Devel Version From GIT http://www.kamailio.org/wiki/install/devel/git #mkdir -p /usr/local/src/kamailio-devel #cd /usr/local/src/kamailio-devel #git clone --depth 1 --no-single-branch git://git.kamailio.org/kamailio kamailio #cd kamailio BR, Julia On Mon, Jan 26, 2015 at 7:04 PM, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, that is not the last devel version, it should be with -dev3. How did you get the sources? I get: version: kamailio 4.3.0-dev3 (x86_64/darwin) 1b334f And yes, if you still get the error, send the logs with the description. Cheers, Daniel On 26/01/15 17:16, Julia Boudniatsky wrote: Hello Daniel, In devel installed, I received the same problem. kamailio -V version: kamailio 4.3.0-dev2 (x86_64/linux) ecd5c5 flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: ecd5c5 compiled on 18:02:05 Jan 26 2015 with gcc 4.4.6 Do you want a log files? Thank you, Julia On Mon, Jan 26, 2015 at 12:47 PM, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, none of existing releases are good because the patches are only in git branches, added after the last relesea (I added it iver this weekend). You must install from git, as pointed by one of the tutorials at: * http://www.kamailio.org/wiki/#installation You must use devel (master), 4.2 (v4.2.x) or 4.1 (v4.1.x) branches. We will have new releases in the near future, like 1-2 weeks from now, but by then I hope to get this bug sorted out. Cheers, Daniel On 26/01/15 11:42, Julia Boudniatsky wrote: Hello Daniel, I installed last 4.2 version: kamailio 4.2.1 (x86_64/linux) d80dfc flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: d80dfc compiled on 10:06:01 Jan 26 2015 with gcc 4.4.6 *Problem in call with Call-ID : 5-5028@10.25.153.150 5-5028@10.25.153.150 * *Short log* *dialog [3748:5628]* *100 trying* Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: core [parser/msg_parser.c:633]: parse_msg(): SIP Reply (status): Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: core [parser/msg_parser.c:635]: parse_msg(): version: SIP/2.0 Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: core [parser/msg_parser.c:637]: parse_msg(): status: 100 Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: core [parser/msg_parser.c:639]: parse_msg(): reason: Trying Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: core [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=9313591363960470767 Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: core [parser/parse_addr_spec.c:898]: parse_addr_spec(): end of header reached, state=29 Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: core [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: To [71]; uri=[sip:039951004@10.25.153.149:5060;user=phone] Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: core [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [ sip:039951004@10.25.153.149:5060;user=phone] Jan 26 10:20:49
[SR-Users] Dispatcher weight dont work
Hello I use dipatcher algorithm 8 that works with weight. I added 2 Asterisks and try to call its with my kam.We use 4.3 version. Tthis config select needed dst from database with my scenario. if(!ds_select_dst($var(setid), 8)) $var(setid)- is variable for setting setid that i get from database with my own scenario. IT does not matter. When running asterisk with weight 90 - all calls goes through it. When I starting asterisk with weight 10 -calls going through asterisk 90. When I shut down asterisk with weight 90 -calls goes through asterisk 10? but when i start asterisk weight 90 all calls goes through sterisk 10 until I shut down it. root@Kamailio:~# kamailio -v version: kamailio 4.3.0-dev3 (x86_64/linux) 8cdbe7 flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 8cdbe7 compiled on 01:17:56 Jan 21 2015 with gcc 4.8.2 id setid destination flags priority attrs 1 2 sip:34.25.123.45:506000 0 weight=10 2 2 sip:10.0.1.6:506000 0weight=90 modparam(dispatcher, db_url,DBURL) modparam(dispatcher, table_name, dispatcher) modparam(dispatcher, setid_col, setid) modparam(dispatcher, destination_col, destination) modparam(dispatcher, force_dst, 1) modparam(dispatcher, flags, 3) modparam(dispatcher, dst_avp, $avp(i:271)) modparam(dispatcher, grp_avp, $avp(i:272)) modparam(dispatcher, cnt_avp, $avp(i:273)) modparam(dispatcher, ds_ping_from, sip:proxy@10.0.1.1) modparam(dispatcher, ds_ping_interval,15) modparam(dispatcher, ds_probing_mode, 1) modparam(dispatcher, ds_ping_reply_codes, class=2;code=403;code=404;code=484;class=3) modparam(tm, reparse_on_dns_failover, 0) Thanks ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] GeoIP Module and IPv6 seems not to match
Hello,
the geoip module is using the old database format (API) that is not
supporting ipv6, there is a new module in the work that uses the new
geoip database format/API.
I already got some code and I asked the developer to make a github pull
request. It should show up soon.
Cheers,
Daniel
On 27/01/15 10:18, Jöran Vinzens wrote:
Hi All,
i'm trying to get GEO IP Location for an IPv6 Setup.
We downloaded the GeoLiteCityv6.dat from maxmind and referred to it in
modparams of geoip.
If i now place a call from an V6 User and do something like:
if(geoip_match($si, src)) {
xlog(L_NOTICE, Call comes from IP '$si'
($gip(src=cc)) CID=$ci F=$fU URI=$ru\n);
append_hf(X-GeoIP: $gip(src=cc)\r\n);
}
it will not match for any reason
if i xlog the src IP it look like: Src IP = 2A01:abc:321:123:0:0:0:2
so it should match to the GEO IP Database.
In CSV version of geo IP database there is an entry for my IP Address.
Since the Module is not that verbose, i cannot tell whether the entry
exists or not. Just the result in my if is False.
Has anybody experience with GEOIP and IPv6?
thanks
Best regards,
Jöran
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Re: [SR-Users] Multiple OR into a if
Hello,
can you paste here the SIP message (or at least the To header and
request URI) for such case? I would like to reproduce.
Also, you can try removing components of the expression in the second IF
one by one to see where it breaks.
Cheers,
Daniel
On 27/01/15 09:45, Igor Potjevlesch wrote:
Hello,
I'm very disappointed because of the following behaviour:
if ($tu=~^sip:0[1-9]{9}) {
[…]
In that case, Kamailio returns TRUE because the
instructions in the block are executed.
}
if ($rU=~^33 || $rU=~^0033 || $rU=~^0[1-9]{9} ||
$rU=~^\+33 || $tu=~^sip:33 || $tu=~^sip:0033 ||
$tu=~^sip:0[1-9]{9} || $tu=~^sip:\+33 ) {
[…]
In that case, Kamailio should return FALSE because the instructions in
the block are not executed.
}
Am I missed something regarding 'OR' ?
Regards,
Igor.
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[SR-Users] Multiple OR into a if
Hello,
I'm very disappointed because of the following behaviour:
if ($tu=~^sip:0[1-9]{9}) {
[.]
In that case, Kamailio returns TRUE because the instructions in
the block are executed.
}
if ($rU=~^33 || $rU=~^0033 || $rU=~^0[1-9]{9} || $rU=~^\+33 ||
$tu=~^sip:33 || $tu=~^sip:0033 || $tu=~^sip:0[1-9]{9} ||
$tu=~^sip:\+33 ) {
[.]
In that case, Kamailio should return FALSE because the instructions in the
block are not executed.
}
Am I missed something regarding 'OR' ?
Regards,
Igor.
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[SR-Users] GeoIP Module and IPv6 seems not to match
Hi All,
i'm trying to get GEO IP Location for an IPv6 Setup.
We downloaded the GeoLiteCityv6.dat from maxmind and referred to it in
modparams of geoip.
If i now place a call from an V6 User and do something like:
if(geoip_match($si, src)) {
xlog(L_NOTICE, Call comes from IP '$si' ($gip(src=cc))
CID=$ci F=$fU URI=$ru\n);
append_hf(X-GeoIP: $gip(src=cc)\r\n);
}
it will not match for any reason
if i xlog the src IP it look like: Src IP = 2A01:abc:321:123:0:0:0:2 so
it should match to the GEO IP Database.
In CSV version of geo IP database there is an entry for my IP Address.
Since the Module is not that verbose, i cannot tell whether the entry
exists or not. Just the result in my if is False.
Has anybody experience with GEOIP and IPv6?
thanks
Best regards,
Jöran
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Re: [SR-Users] Fosdem 2015
yep, Emil is waiting for us :-) Regards, Alexandr 2015-01-26 14:53 GMT+01:00 Daniel-Constantin Mierla mico...@gmail.com: Very likely I will be around at Fosdem as well this year and going for the event in the evening, if it takes place. Have you discussed with Jitsi guys? Cheers, Daniel On 17/01/15 16:47, Alexandr Dubovikov wrote: so, same event like last year at Beer Mania. Enjoy! 2015-01-16 21:12 GMT+01:00 Alexandr Dubovikov aduv...@googlemail.com: ok, Torsten Schweizer, Heino Klier and I will be at Fosdem. We can make same beer event like last year, together with Emil and Jitsi Co. 2015-01-15 12:47 GMT+01:00 DanB danb.li...@gmail.com: I will be there this year also, if someone prepares an event. DanB. ___ 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 listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierlahttp://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] Need help on WebRTC with Kamailio as proxy
Hello Richard, Now, after upgrading to - OpenSSL 1.0.1j 15 Oct 2014 Errors like - Failed to set up SRTP after DTLS negotiation: no SRTP protection profile negotiated are not seen but still the call is not getting through. Please let me know how to proceed.. Thanks in advance On Tue, Jan 27, 2015 at 2:14 AM, Rahul MathuR rahul.ultim...@gmail.com wrote: Hi Richard, Thanks for spending some cycles on it. It is OpenSSL 1.0.1e-fips 11 Feb 2013 On Tue, Jan 27, 2015 at 2:04 AM, Richard Fuchs rfu...@sipwise.com wrote: On 26/01/15 02:21 PM, Rahul MathuR wrote: Hello, I am totally struck at a point while implementing Kamailio as proxy for WebRTC enabled UAC (Jssip). I am using Google's TURN server (rfc5766-turn-server for ICE/STUN). I am able to get to the point where the SIP server sends 183 session in progress to kamailio but after that I can only see - STUN: using this candidate Successful STUN binding request from .. SRTP output wanted, but no crypto suite was negotiated This is fairly strange: Jan 27 00:35:46 localhost rtpengine[5262]: [tsb1jrsqsadn33jjsi4f port 30794] Failed to set up SRTP after DTLS negotiation: no SRTP protection profile negotiated Jan 27 00:35:46 localhost rtpengine[5262]: [tsb1jrsqsadn33jjsi4f port 30794] Failed to set up SRTP after DTLS negotiation: no SRTP protection profile negotiated Are you running a very old OpenSSL version by any chance? cheers ___ 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 -- Warm Regds. MathuRahul -- Warm Regds. MathuRahul Jan 27 02:14:17 localhost kamailio[2246]: INFO: script: Starting of request_route - #012 port is [10080],#011 proto is [ws],#011 user is [919650926333]#012 Message [INVITE sip:919650926333@125.99.186.124 SIP/2.0#015#012Route: sip:125.99.186.126:10080;transport=ws;lr#015#012Via: SIP/2.0/WS vf6huklcsg7i.invalid;branch=z9hG4bK8132976#015#012Max-Forwards: 69#015#012To: sip:919650926333@125.99.186.124#015#012From: 114488 sip:114488@125.99.186.124;tag=iiq2flnorq#015#012Call-ID: 2mi1qf5cb3mg9vf0bp22#015#012CSeq: 2623 INVITE#015#012Contact: sip:emf13f83@vf6huklcsg7i.invalid;transport=ws;ob#015#012Content-Type: application/sdp#015#012Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS#015#012Supported: ice,outbound#015#012User-Agent: JsSIP 0.6.4#015#012Content-Length: 3042#015#012#015#012v=0#015#012o=- 3479457386797450545 2 IN IP4 127.0.0.1#015#012s=-#015#012t=0 0#015#012a=group:BUNDLE audio video#015#012a=msid-semantic: WMS 5DKLhoDH63ksNQLPuJfODFZ0We62dvZHeypu#015#012m=audio 21455 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126#015#012c=IN IP4 59.178.53.133#015#012a=rtcp:21455 IN IP4 59.178.53.133#015#012a=candidate:704553097 1 udp 2122260223 192.168.1.3 61780 typ host generation 0#015#012a=candidate:704553097 2 udp 2122260223 192.168.1.3 61780 typ host generation 0#015#012a=candidate:1736268921 1 tcp 1518280447 192.168.1.3 0 typ host tcptype active generation 0#015#012a=candidate:1736268921 2 tcp 1518280447 192.168.1.3 0 typ host tcptype active generation 0#015#012a=candidate:2158047068 1 udp 1686052607 59.178.53.133 21455 typ srflx raddr 192.168.1.3 rport 61780 generation 0#015#012a=candidate:2158047068 2 udp 1686052607 59.178.53.133 21455 typ srflx raddr 192.168.1.3 rport 61780 generation 0#015#012a=ice-ufrag:u4MhDJ9rtq6tLJi+#015#012a=ice-pwd:hbArhf2sDAU/BfClFII2LHm8#015#012a=ice-options:google-ice#015#012a=fingerprint:sha-256 AA:79:7A:FA:6E:B8:38:A4:1B:5E:60:4A:27:67:96:76:2F:09:C8:E7:2F:5B:D6:0B:0B:DF:10:31:4A:B8:27:AA#015#012a=setup:actpass#015#012a=mid:audio#015#012a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-a Jan 27 02:14:17 localhost kamailio[2246]: INFO: script: Calling REQINIT, per request initial checks#012 Jan 27 02:14:17 localhost kamailio[2246]: WARNING: script: Inside REQINIT - checking the hops Jan 27 02:14:17 localhost kamailio[2246]: WARNING: script: Inside REQINIT - Sanity check Jan 27 02:14:17 localhost kamailio[2246]: ERROR: script: Calling NATDETECT#012 Jan 27 02:14:17 localhost kamailio[2246]: ERROR: script: Inside NATDETECT Jan 27 02:14:17 localhost kamailio[2246]: ERROR: script: Inside NATDETECT - Non-REGISTER Method hence - callling add_contact_alias Jan 27 02:14:17 localhost kamailio[2246]: ERROR: script: Exiting from NATDETECT Jan 27 02:14:17 localhost kamailio[2246]: ERROR: script: CAlling WITHINDLG - handle requests within SIP dialogs Jan 27 02:14:17 localhost kamailio[2246]: ERROR: script: Inside WITHINDLG Jan 27 02:14:17 localhost kamailio[2246]: ERROR: script: Removing Route header Jan 27 02:14:17 localhost kamailio[2246]: ERROR: script: Removed Route header and adding new one.. Jan 27 02:14:17 localhost kamailio[2246]: ERROR: script: New message becomes - [INVITE sip:919650926333@125.99.186.124 SIP/2.0#015#012Route:
[SR-Users] Rpc question
Hi, I need to request to kamailio the list of all the active dialog (with some dialog variable). Now , i'm doing it with a program that call the proxy with the xmlrpc module. The problem is that when it arrive to 350-400 calls, the reply fail because it has no memory free. Now ,i have setted the pkg memory to 12MB but at peak it reply with error. Now i shoul increase it. Is there another method to optimize the request of these data, to avoid every time to increase the memory parameter? Thanks to all ___ 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] GeoIP Module and IPv6 seems not to match
The pull request with the patch submitted.
On Tuesday 27 January 2015 07:59:04 Sergey Okhapkin wrote:
Yes, geoip2 module will handle v4 and v6 addresses at the same time.
I'm learning now how to use github and will issue pull request today.
On Tuesday 27 January 2015 12:54:59 Jöran Vinzens wrote:
Hi, thanks for that info.
Will there be support to handle V4 and V6 at the same time? So far there
is
the possibility just for one database in the module to configure.
BR
Jöran
On Tue, Jan 27, 2015 at 10:26 AM, Daniel-Constantin Mierla
mico...@gmail.com wrote:
Hello,
the geoip module is using the old database format (API) that is not
supporting ipv6, there is a new module in the work that uses the new
geoip
database format/API.
I already got some code and I asked the developer to make a github pull
request. It should show up soon.
Cheers,
Daniel
On 27/01/15 10:18, Jöran Vinzens wrote:
Hi All,
i'm trying to get GEO IP Location for an IPv6 Setup.
We downloaded the GeoLiteCityv6.dat from maxmind and referred to it in
modparams of geoip.
If i now place a call from an V6 User and do something like:
if(geoip_match($si, src)) {
xlog(L_NOTICE, Call comes from IP '$si'
($gip(src=cc))
CID=$ci F=$fU URI=$ru\n);
append_hf(X-GeoIP: $gip(src=cc)\r\n);
}
it will not match for any reason
if i xlog the src IP it look like: Src IP = 2A01:abc:321:123:0:0:0:2
so it should match to the GEO IP Database.
In CSV version of geo IP database there is an entry for my IP Address.
Since the Module is not that verbose, i cannot tell whether the entry
exists or not. Just the result in my if is False.
Has anybody experience with GEOIP and IPv6?
thanks
Best regards,
Jöran
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Re: [SR-Users] Multiple OR into a if
The To header looks like this (in compact form):
t: sip:0123456...@sip.domain.tld;tag=f15e211394273201512715012\r\n
Regards,
Igor.
De : Igor Potjevlesch [mailto:igor.potjevle...@gmail.com]
Envoyé : mardi 27 janvier 2015 15:55
À : mico...@gmail.com; 'Kamailio (SER) - Users Mailing List'
Objet : RE: [SR-Users] Multiple OR into a if
Hello Daniel,
Just to let you, it's in MANAGE_REPLY. Is that make any difference?
Regards,
Igor.
De : sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de
Daniel-Constantin Mierla
Envoyé : mardi 27 janvier 2015 10:32
À : Kamailio (SER) - Users Mailing List
Objet : Re: [SR-Users] Multiple OR into a if
Hello,
can you paste here the SIP message (or at least the To header and request
URI) for such case? I would like to reproduce.
Also, you can try removing components of the expression in the second IF one
by one to see where it breaks.
Cheers,
Daniel
On 27/01/15 09:45, Igor Potjevlesch wrote:
Hello,
I'm very disappointed because of the following behaviour:
if ($tu=~^sip:0[1-9]{9}) {
[
]
In that case, Kamailio returns TRUE because the instructions in
the block are executed.
}
if ($rU=~^33 || $rU=~^0033 || $rU=~^0[1-9]{9} || $rU=~^\+33 ||
$tu=~^sip:33 || $tu=~^sip:0033 || $tu=~^sip:0[1-9]{9} ||
$tu=~^sip:\+33 ) {
[
]
In that case, Kamailio should return FALSE because the instructions in the
block are not executed.
}
Am I missed something regarding 'OR' ?
Regards,
Igor.
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Re: [SR-Users] Multiple OR into a if
Hello Daniel,
Just to let you, it's in MANAGE_REPLY. Is that make any difference?
Regards,
Igor.
De : sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de
Daniel-Constantin Mierla
Envoyé : mardi 27 janvier 2015 10:32
À : Kamailio (SER) - Users Mailing List
Objet : Re: [SR-Users] Multiple OR into a if
Hello,
can you paste here the SIP message (or at least the To header and request
URI) for such case? I would like to reproduce.
Also, you can try removing components of the expression in the second IF one
by one to see where it breaks.
Cheers,
Daniel
On 27/01/15 09:45, Igor Potjevlesch wrote:
Hello,
I'm very disappointed because of the following behaviour:
if ($tu=~^sip:0[1-9]{9}) {
[
]
In that case, Kamailio returns TRUE because the instructions in
the block are executed.
}
if ($rU=~^33 || $rU=~^0033 || $rU=~^0[1-9]{9} || $rU=~^\+33 ||
$tu=~^sip:33 || $tu=~^sip:0033 || $tu=~^sip:0[1-9]{9} ||
$tu=~^sip:\+33 ) {
[
]
In that case, Kamailio should return FALSE because the instructions in the
block are not executed.
}
Am I missed something regarding 'OR' ?
Regards,
Igor.
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Re: [SR-Users] Need help on WebRTC with Kamailio as proxy
Hi Rahul, Don't take me wrong, but you still have some homework to do. Apache is not a requirement for webrtc ( apart hosting the website ). The only difference between using http and https is that by default on http, most browsers will always ask for the user to confirm usage of the mic/cam. WS and WSS works with Kamailio, it's only a question of configuration ( for which there are many examples, most are broken but easily fixed, for example for https://github.com/caruizdiaz/kamailio-ws , it's only fixing the record routes to get sip2ws signaling working ). Regarding rtp, you have to use rtpengine ( master from repo, not a release, dtls broken in latest 3.7.1, fixed in 3.8 ) or something else to be able to terminate ICE/DTLS when remote endpoints don't support them ( most of SIP ua's today unfortunately ), again, read, experiment, you'll eventually get it and the most important, know how your platform works ! Start with basic browser to browser calls, without a rtp proxy, it should work almost out of the box, then you can add some functionnality to the basic scenario, and I'll be glad to point you to the right direction ! Good luck ! Le 27/01/2015 03:21, Rahul MathuR a écrit : Any thoughts on this gents ? On Tue, Jan 27, 2015 at 8:09 AM, rahul.ultimate rahul.ultim...@gmail.com mailto:rahul.ultim...@gmail.com wrote: Kamailio is just acting as a proxy and protocol modifier so to say. It is workin with rtpengine from sipwise to handle media as evident from he logs. This architectue uses a TURN server and the browser is chrome with latest updates. The only thing whih I haven't done is enable TLS in kamailio and create certs. (which I'm not completely sure how to do).. Also, does it necessitates to have Apache ruuning https on 443 ? Thanks in advance Sent from Samsung Mobile Original message From: Gonzalo Gasca Meza Date:27/01/2015 4:07 AM (GMT+05:30) To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Need help on WebRTC with Kamailio as proxy Are you terminating media in Kamailio or just handling WS communication? If yes which version of Kamailio and rtp-proxy ? Have you tried passing media directly between Browser and Kamailio with any TURN server? Are you using latest Chrome version or FF ? A working sample config using the following architecture: https://github.com/spicyramen/llamato/tree/LlamatoReg signalling: sipml5 -- ws/wss -- Ec2 Kamailio --sip udp-- FS --sip udp-- * media: sipml5 * On Mon, Jan 26, 2015 at 12:44 PM, Rahul MathuR rahul.ultim...@gmail.com mailto:rahul.ultim...@gmail.com wrote: Hi Richard, Thanks for spending some cycles on it. It is OpenSSL 1.0.1e-fips 11 Feb 2013 On Tue, Jan 27, 2015 at 2:04 AM, Richard Fuchs rfu...@sipwise.com mailto:rfu...@sipwise.com wrote: On 26/01/15 02:21 PM, Rahul MathuR wrote: Hello, I am totally struck at a point while implementing Kamailio as proxy for WebRTC enabled UAC (Jssip). I am using Google's TURN server (rfc5766-turn-server for ICE/STUN). I am able to get to the point where the SIP server sends 183 session in progress to kamailio but after that I can only see - STUN: using this candidate Successful STUN binding request from .. SRTP output wanted, but no crypto suite was negotiated This is fairly strange: Jan 27 00:35:46 localhost rtpengine[5262]: [tsb1jrsqsadn33jjsi4f port 30794] Failed to set up SRTP after DTLS negotiation: no SRTP protection profile negotiated Jan 27 00:35:46 localhost rtpengine[5262]: [tsb1jrsqsadn33jjsi4f port 30794] Failed to set up SRTP after DTLS negotiation: no SRTP protection profile negotiated Are you running a very old OpenSSL version by any chance? cheers ___ 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 -- Warm Regds. MathuRahul ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org
Re: [SR-Users] Need help on WebRTC with Kamailio as proxy
Any thoughts on this gents ? On Tue, Jan 27, 2015 at 8:09 AM, rahul.ultimate rahul.ultim...@gmail.com wrote: Kamailio is just acting as a proxy and protocol modifier so to say. It is workin with rtpengine from sipwise to handle media as evident from he logs. This architectue uses a TURN server and the browser is chrome with latest updates. The only thing whih I haven't done is enable TLS in kamailio and create certs. (which I'm not completely sure how to do).. Also, does it necessitates to have Apache ruuning https on 443 ? Thanks in advance Sent from Samsung Mobile Original message From: Gonzalo Gasca Meza Date:27/01/2015 4:07 AM (GMT+05:30) To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Need help on WebRTC with Kamailio as proxy Are you terminating media in Kamailio or just handling WS communication? If yes which version of Kamailio and rtp-proxy ? Have you tried passing media directly between Browser and Kamailio with any TURN server? Are you using latest Chrome version or FF ? A working sample config using the following architecture: https://github.com/spicyramen/llamato/tree/LlamatoReg signalling: sipml5 -- ws/wss -- Ec2 Kamailio --sip udp-- FS --sip udp-- * media: sipml5 * On Mon, Jan 26, 2015 at 12:44 PM, Rahul MathuR rahul.ultim...@gmail.com wrote: Hi Richard, Thanks for spending some cycles on it. It is OpenSSL 1.0.1e-fips 11 Feb 2013 On Tue, Jan 27, 2015 at 2:04 AM, Richard Fuchs rfu...@sipwise.com wrote: On 26/01/15 02:21 PM, Rahul MathuR wrote: Hello, I am totally struck at a point while implementing Kamailio as proxy for WebRTC enabled UAC (Jssip). I am using Google's TURN server (rfc5766-turn-server for ICE/STUN). I am able to get to the point where the SIP server sends 183 session in progress to kamailio but after that I can only see - STUN: using this candidate Successful STUN binding request from .. SRTP output wanted, but no crypto suite was negotiated This is fairly strange: Jan 27 00:35:46 localhost rtpengine[5262]: [tsb1jrsqsadn33jjsi4f port 30794] Failed to set up SRTP after DTLS negotiation: no SRTP protection profile negotiated Jan 27 00:35:46 localhost rtpengine[5262]: [tsb1jrsqsadn33jjsi4f port 30794] Failed to set up SRTP after DTLS negotiation: no SRTP protection profile negotiated Are you running a very old OpenSSL version by any chance? cheers ___ 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 -- Warm Regds. MathuRahul ___ 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 -- Warm Regds. MathuRahul ___ 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 can change via header's ip to public ip when Kamailio locate behind a NAT network?
Thanks, the public ip is not a real device ip, it is a NAT external public ip, so if I listen on that, server will not work. Anyway, I found my problem is caused by firewall, so even the via header is priviate ip, I still can get response, so please ignore this question, thanks for your help. At 2015-01-26 17:14:46, Olle E. Johansson o...@edvina.net wrote: On 25 Jan 2015, at 14:48, dongwf dongw...@163.com wrote: Hi Kamailio: I use Kamailio with Amazon EC2 virtual machine, it has a private ip such as 172.31.7.164, kamailio listened on this, I assigned a float public ip 54.X.X.X on it, and all the client send traffics to the public ip and it work well(REGISTER), but now I would relay kamailio's traffic to another real public server, the server found the INVITE's via header's ip address is 172.31.7.164, so it failed to response 100 trying and 200 OK to my kamalio, because it send to a private 172.31.7.164, so my question is can I change Kamailio's via header's ip value with my specified public ip? How can I do? Thanks a lot! The core cookbok documents the advertise parameter to the listen config setting for this case. http://www.kamailio.org/wiki/cookbooks/4.2.x/core#listen Go ahead and try that one. /O___ 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] GeoIP Module and IPv6 seems not to match
Hi, thanks for that info.
Will there be support to handle V4 and V6 at the same time? So far there is
the possibility just for one database in the module to configure.
BR
Jöran
On Tue, Jan 27, 2015 at 10:26 AM, Daniel-Constantin Mierla
mico...@gmail.com wrote:
Hello,
the geoip module is using the old database format (API) that is not
supporting ipv6, there is a new module in the work that uses the new geoip
database format/API.
I already got some code and I asked the developer to make a github pull
request. It should show up soon.
Cheers,
Daniel
On 27/01/15 10:18, Jöran Vinzens wrote:
Hi All,
i'm trying to get GEO IP Location for an IPv6 Setup.
We downloaded the GeoLiteCityv6.dat from maxmind and referred to it in
modparams of geoip.
If i now place a call from an V6 User and do something like:
if(geoip_match($si, src)) {
xlog(L_NOTICE, Call comes from IP '$si' ($gip(src=cc))
CID=$ci F=$fU URI=$ru\n);
append_hf(X-GeoIP: $gip(src=cc)\r\n);
}
it will not match for any reason
if i xlog the src IP it look like: Src IP = 2A01:abc:321:123:0:0:0:2
so it should match to the GEO IP Database.
In CSV version of geo IP database there is an entry for my IP Address.
Since the Module is not that verbose, i cannot tell whether the entry
exists or not. Just the result in my if is False.
Has anybody experience with GEOIP and IPv6?
thanks
Best regards,
Jöran
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Re: [SR-Users] topoh ACK Call-ID mismatch
Am I to naive to think that these ACKs to negatives always (callid masking
and whether topoh is active or not) need to use the Call-ID from the
negative (in this case 488) response? Call-ID has only to be rewritten in
forwarding the negative response towards the endpoint that triggered it
(which is done correctly in my call trace)
Attached is a diff to topoh_mod.c that implements above (with my limited
knowledge). Locally generated ACKs don't get masked Call-IDs to downstream.
Works for my 488, will test some more to see if nothing else is broken. But
maybe somebody more in touch with kamailio source code could take a look to
see it this patch looks alright?
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17097024
--- topoh_mod.c.orig 2015-01-27 11:12:48.680160911 +0100
+++ topoh_mod.c 2015-01-27 10:51:23.162037065 +0100
@@ -397,11 +397,13 @@
LM_DBG(the COOKIE is [%.*s]\n, th_cookie_value.len, th_cookie_value.s);
if(th_cookie_value.s[0]!='x')
th_del_cookie(msg);
+
+ direction = (th_cookie_value.s[0]=='u')?1:0; /* upstream/downstram */
+ dialog = (get_to(msg)-tag_value.len0)?1:0;
+ local = (th_cookie_value.s[0]!='d'th_cookie_value.s[0]!='u')?1:0;
+
if(msg.first_line.type==SIP_REQUEST)
{
- direction = (th_cookie_value.s[0]=='u')?1:0; /* upstream/downstram */
- dialog = (get_to(msg)-tag_value.len0)?1:0;
- local = (th_cookie_value.s[0]!='d'th_cookie_value.s[0]!='u')?1:0;
/* local generated requests */
if(local)
{
@@ -409,10 +411,10 @@
if(get_cseq(msg)-method_id==METHOD_ACK
|| get_cseq(msg)-method_id==METHOD_CANCEL)
{
-th_mask_callid(msg);
goto ready;
} else {
/* should be for upstream */
+th_unmask_callid(msg);
goto done;
}
}
@@ -434,24 +436,30 @@
}
} else {
/* reply */
- if(th_cookie_value.s[th_cookie_value.len-1]=='x')
- {
- /* ?!?! - we should have a cookie in any reply case */
- goto done;
- }
- if(th_cookie_value.s[th_cookie_value.len-1]=='v')
+
+ if(local direction (get_cseq(msg)-method_id==METHOD_ACK))
{
- /* reply generated locally - direction was set by request */
- if(th_cookie_value.s[0]=='u')
+ th_unmask_callid(msg);
+ } else {
+ if(th_cookie_value.s[th_cookie_value.len-1]=='x')
{
-th_mask_callid(msg);
+/* ?!?! - we should have a cookie in any reply case */
+goto done;
}
- } else {
- th_flip_record_route(msg, 1);
- th_mask_contact(msg);
- if(th_cookie_value.s[0]=='d')
+ if(th_cookie_value.s[th_cookie_value.len-1]=='v')
{
-th_mask_callid(msg);
+/* reply generated locally - direction was set by request */
+if(th_cookie_value.s[0]=='u')
+{
+ th_mask_callid(msg);
+}
+ } else {
+th_flip_record_route(msg, 1);
+th_mask_contact(msg);
+if(th_cookie_value.s[0]=='d')
+{
+ th_mask_callid(msg);
+}
}
}
}
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Re: [SR-Users] Async module taking down our server
Hello
I wanted to give an update on this.
My business partner that found the issue and has been monitoring the
problem has tracked down the issue. It turns out that the features we
implemented using the async module were leading to more calls going on con
currently (as they were intended to) and this was causing and issue with
voip monitor. So the issue was not with the Async module.
All the best.
Will Ferrer
Switchsoft
On Mon, Jan 19, 2015 at 8:43 PM, Will Ferrer will.fer...@switchsoft.com
wrote:
Hi All
We are trying to use the async module to to delay sending a bye on from
one end of the call to the other.
We are using async_route(routename, seconds) to delay the WITHINDLG route.
The idea is that in the future we want to be able to have our billing min
duration enforced (though currently we are having issues with the dialog
module that we are discussing in another thread).
After running this on our deploy servers, the delays before sending on the
byes get longer and longer, and then kamailio goes down. Then the receive
udp buffer fills up.
We tried it with both 4 and 400 async workers, and it made no difference.
I am including a screen capture of the servers stats when this happens
taken from voip monitor.
Here are the relevant parts of the config:
...
loadmodule async.so
...
modparam(async, workers, ASYNC_THREADS)
...
request_route {
...
route(DELAYED_BYE_STATIC);
...
route[DELAYED_BYE_STATIC] {
xlog(L_DEBUG,route DELAYED_BYE_STATIC);
#!ifdef WITH_DELAYED_BYE_STATIC
if (is_method(BYE)) {
xlog(L_DEBUG,route DELAYED_BYE_STATIC, from self \n);
#if (from_uri == myself) {
if ((allow_trusted() || allow_source_address()) from_uri == myself) {
xlog(L_DEBUG,route DELAYED_BYE_STATIC, Bye detected, from self \n);
send_reply(200, OK);
xlog(L_DEBUG,route DELAYED_BYE_STATIC, sent 200 about to sleep \n);
setflag(FLT_ACC); # do accounting ...
setflag(FLT_ACCFAILED); # ... even if the transaction fails
if (has_totag()) {
xlog(L_DEBUG,route DELAYED_BYE_STATIC, sleeping to WITHINDLG_DELAYED
\n);
async_route(WITHINDLG_DELAYED, MIN_DURATION);
} else {
xlog(L_DEBUG,route DELAYED_BYE_STATIC, sleeping to WITHINDLG \n);
async_route(WITHINDLG, MIN_DURATION);
}
xlog(L_DEBUG,route DELAYED_BYE_STATIC, slept\n);
exit;
}
}
#!endif
return;
}
...
route[WITHINDLG_DELAYED] {
xlog(L_DEBUG, route WITHINDLG_DELAYED: triggered \n);
$avp(was_delayed) = 1;
route(WITHINDLG);
}
...
route[WITHINDLG] {
xlog(L_DEBUG, route WITHINDLG: will -- DLG triggered, request method:
$rm \n);
#!ifdef WITH_DISPATCHER
if(is_method(BYE|CANCEL)) {
xlog(L_DEBUG,route WITHINDLG: cancel or bye detected, request method:
$rm \n);
#!ifdef WITH_DISPATCHER_LOAD_AWARE
xlog(L_DEBUG,route WITHINDLG: running ds_load_update, request method:
$rm \n);
ds_load_update();
#dlg_get ($ci,$ft,$tt);
#dlg_bye (all);
#!endif
}
#!endif
if (has_totag() || $avp(was_delayed) == 1) {
xlog(L_DEBUG, route WITHINDLG: will -- DLG has totag or was_delayed:
$avp(was_delayed) \n);
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
xlog(L_DEBUG, route WITHINDLG: will -- DLG has loose route \n);
route(DLGURI);
if (is_method(BYE)) {
xlog(L_DEBUG,route WITHINDLG: BYE detected);
setflag(FLT_ACC); # do accounting ...
setflag(FLT_ACCFAILED); # ... even if the transaction fails
xlog(L_DEBUG,route WITHINDLG: ACC flag set);
}
else if ( is_method(ACK) ) {
# ACK is forwarded statelessy
route(NATMANAGE);
}
else if ( is_method(NOTIFY) ) {
# Add Record-Route for in-dialog NOTIFY as per RFC 6665.
record_route();
}
xlog(L_DEBUG, route WITHINDLG: will -- DLG RELAY 1\n);
route(RELAY);
} else {
xlog(L_DEBUG, route WITHINDLG: will -- DLG else \n);
if (is_method(SUBSCRIBE) uri == myself) {
# in-dialog subscribe requests
route(PRESENCE);
exit;
}
if ( is_method(ACK) ) {
xlog(L_DEBUG, route WITHINDLG: will -- DLG is ack \n);
if ( t_check_trans() ) {
xlog(L_DEBUG, route WITHINDLG: will -- DLG t_check_trans \n);
# no loose-route, but stateful ACK;
# must be an ACK after a 487
# or e.g. 404 from upstream server
xlog(L_DEBUG, route WITHINDLG: will -- DLG RELAY 2\n);
route(RELAY);
exit;
} else {
# ACK without matching transaction ... ignore and discard
exit;
}
}
sl_send_reply(404,Not here);
}
exit;
}
}
...
Does any one know if this is a bug or a leak with in the async module, or
perhaps something I am doing in my config?
Thanks in advance for an assistance you can offer me.
All the best.
Will Ferrer
Switchsoft
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Re: [SR-Users] Trouble getting start time / call duration from dialog module
Hi.
I wanted to update every one on this issue.
I found that while the dlg start time: $dlg(start_ts), DLG_lifetime:
$DLG_lifetime aren't populating I can still use the following to record the
start time of the call as a dialog variable:
$dlg_var(TVs) = $TV(s);
A strange problem with this however is that when I compare that start time
to the time when the bye occurs I see 10 extra seconds that never actually
happened.
In other words this:
$avp(elapsed) = ( $TV(s) - $dlg_var(TVs));
sets my $avp(elapsed) to be 10 seconds longer than the call actually was.
I fixed it simply enough like so:
$var(dialog_duration_fix) = DIALOG_DURATION_FIX;
$avp(elapsed) = ( ($TV(s) - $var(dialog_duration_fix)) - $dlg_var(TVs));
I have worked around these issues for my own purposes but still seems like
there are some odd bugs here.
I hope this message finds every well.
All the best.
Will Ferrer
Switchsoft
On Thu, Jan 22, 2015 at 3:56 PM, Will Ferrer will.fer...@switchsoft.com
wrote:
Hi Abdul
I just wanted to check int. Did you have any ideas of things we could try
or more information we could get you to assistance?
Thanks again for your help, the input we have gotten from you and Daniel
has always been invaluable.
All the best.
Will Ferrer
Switchsoft
On Fri, Jan 16, 2015 at 7:42 PM, Will Ferrer will.fer...@switchsoft.com
wrote:
Hi Abdul
That's great. I look forward to the advice.
Thanks again for the assistance and I hope you have a great weekend.
All the best.
Will
On Fri, Jan 16, 2015 at 6:44 PM, Abdul Gafar abdul.gafar@gmail.com
wrote:
Hi Wiil
I will be try help you
//Gafar
On Sat, Jan 17, 2015 at 4:43 AM, Will Ferrer will.fer...@switchsoft.com
wrote:
Hi Adbul
Thanks, I am glad the information is useful.
Do you have any thoughts on what I could I try next to get that
$dlg(start_ts)
value populated in the dialog as it is still not working about adding the
loose_route() function?
I also tried looking at the logs for any reference to dialog -- tail -f
/var/log/syslog | grep dialog
The only thing that came up with rtpengine talking about dialogs so no
clues there.
Thanks to you guys for the help.
All the best.
Will
On Fri, Jan 16, 2015 at 1:25 AM, Abdul Gafar abdul.gafar@gmail.com
wrote:
*Hi Will,Thanks for sharing very useful information.*
*//Gafar*
On Fri, Jan 16, 2015 at 12:48 PM, Will Ferrer
will.fer...@switchsoft.com wrote:
Hi Daniel
Thanks so much for the response and help as always.
I tried changing my config to use loose route.
It looks like this now:
loadmodule dialog.so
...
modparam(dialog, db_url, DBURL)
modparam(dialog, db_mode, 1)
modparam(dialog, dlg_flag, 4)
modparam(dialog, dlg_match_mode, 1)
...
request_route {
if (is_method(INVITE) (! has_totag() ) ) {
dlg_manage();
xlog (L_INFO, request_route DIALOG TEST: Dialog initiated);
}
if (is_method(BYE)) {
#dlg_manage();
loose_route();
$var(elapsed) = ( $TV(s) - $dlg(start_ts) );
xlog (L_INFO, request_route DIALOG TEST: Completed $dlg(from_uri)
to $dlg(to_uri), elapsed: $var(elapsed), now seconds: $TV(s), dlg start
time: $dlg(start_ts), DLG_lifetime: $DLG_lifetime);
}
I now get:
INFO: script: request_route DIALOG TEST: Dialog initiate
INFO: script: request_route DIALOG TEST: Completed
sip:willf1976t...@develop-sbc.switchsoft.com to
sip:+18054515...@develop-sbc.switchsoft.com, elapsed: 1421386898,
now seconds: 1421386898, dlg start time: 0, DLG_lifetime: 1421386898
So the $DLG_lifetime is being populated, but it has all the seconds
since epoch time. You can also see that the $dlg(start_ts) is 0.
I also tried using the dlg_manage() instead of the loose route in my
test and got the same result.
Any idea what might be missing?
Thanks again for your help.
All the best.
Will
On Thu, Jan 15, 2015 at 5:53 AM, Daniel-Constantin Mierla
mico...@gmail.com wrote:
Hello,
try to do dlg_manage() or lose_route() before accessing the dialog
variables.
Cheers,
Daniel
On 15/01/15 08:25, Will Ferrer wrote:
An update on this.
I tried setting my dialog module to the use the db. No db entry is
ever made.
My config looks like this now:
loadmodule dialog.so
...
modparam(dialog, db_url, DBURL)
modparam(dialog, db_mode, 1)
modparam(dialog, dlg_flag, 4)
modparam(dialog, dlg_match_mode, 1)
...
request_route {
if (is_method(INVITE) (! has_totag() ) ) {
dlg_manage();
}
if (is_method(BYE)) {
$var(elapsed) = ( $TV(s) - $dlg(start_ts) );
xlog (L_INFO, request_route DIALOG TEST: Completed
$dlg(from_uri) to $dlg(to_uri), elapsed: $var(elapsed), now seconds:
$TV(s), dlg start time: $dlg(start_ts), DLG_lifetime: $DLG_lifetime);
}
I hope this message finds every one well.
All the best.
Will
On Thu, Jan 15, 2015 at 12:03 AM, Will Ferrer
will.fer...@switchsoft.com wrote:
Hi All
I am
Re: [SR-Users] GeoIP Module and IPv6 seems not to match
Yes, geoip2 module will handle v4 and v6 addresses at the same time.
I'm learning now how to use github and will issue pull request today.
On Tuesday 27 January 2015 12:54:59 Jöran Vinzens wrote:
Hi, thanks for that info.
Will there be support to handle V4 and V6 at the same time? So far there is
the possibility just for one database in the module to configure.
BR
Jöran
On Tue, Jan 27, 2015 at 10:26 AM, Daniel-Constantin Mierla
mico...@gmail.com wrote:
Hello,
the geoip module is using the old database format (API) that is not
supporting ipv6, there is a new module in the work that uses the new geoip
database format/API.
I already got some code and I asked the developer to make a github pull
request. It should show up soon.
Cheers,
Daniel
On 27/01/15 10:18, Jöran Vinzens wrote:
Hi All,
i'm trying to get GEO IP Location for an IPv6 Setup.
We downloaded the GeoLiteCityv6.dat from maxmind and referred to it in
modparams of geoip.
If i now place a call from an V6 User and do something like:
if(geoip_match($si, src)) {
xlog(L_NOTICE, Call comes from IP '$si' ($gip(src=cc))
CID=$ci F=$fU URI=$ru\n);
append_hf(X-GeoIP: $gip(src=cc)\r\n);
}
it will not match for any reason
if i xlog the src IP it look like: Src IP = 2A01:abc:321:123:0:0:0:2
so it should match to the GEO IP Database.
In CSV version of geo IP database there is an entry for my IP Address.
Since the Module is not that verbose, i cannot tell whether the entry
exists or not. Just the result in my if is False.
Has anybody experience with GEOIP and IPv6?
thanks
Best regards,
Jöran
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Daniel-Constantin Mierlahttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
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Re: [SR-Users] topoh ACK Call-ID mismatch
On Tuesday 27 January 2015 11:24:59 Daniel Tryba wrote: Works for my 488, will test some more to see if nothing else is broken. I was a bit optimistic, patch breaks if the called endpoint responds with a 4xx :( -- Telefoon: 088 0100 700 Sales: sa...@pocos.nl | Service: serviced...@pocos.nl http://www.pocos.nl/ | Croy 9c, 5653 LC Eindhoven | Kamer van Koophandel 17097024 ___ 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] CDRs for failed calls
Hi Mickael Have you tried looking at these examples in the ACC module to see if what you are looking for can be accomplished at least as far creating ACC records: http://kamailio.org/docs/modules/4.0.x/modules/acc.html#idp1626120 http://kamailio.org/docs/modules/4.0.x/modules/acc.html#idp1628288 http://kamailio.org/docs/modules/4.0.x/modules/acc.html#idp1632512 http://kamailio.org/docs/modules/4.0.x/modules/acc.html#idp1644232 We also use a mysql routine to move ACC into CDRs in the database, so if we wan't CDRs for failed calls I would modify our routine to make them. While we do not use it our selves, there is this info here on having the ACC module make CDRs: http://kamailio.org/docs/modules/4.0.x/modules/acc.html#idp91080 I hope that helps. All the best. Will Ferrer Switchsoft On Mon, Jan 26, 2015 at 5:23 AM, Mickael Marrache mickaelmarra...@gmail.com wrote: Hi, Is there a way to write CDRs for failed calls? I tried calling the acc_db_request command from my script, but it inserts a transaction log not a CDR. Thanks, Mickael ___ 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
