Re: [SR-Users] Help with sip balancer
First of all I'd suggest to use http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb guide in combination with http://saevolgo.blogspot.com/2011/11/how-to-increasing-voip-services.html But, assuming your platform is behind NAT, you need: 1st. Use rtpengine instead of rtpproxy. You can read about how to advertise your external public adress on rtpengine git page. 2nd. In Kamailio configuration when you define listen, you should use listen - advertise construction ( http://www.kamailio.org/wiki/cookbooks/4.0.x/core#listen). 3d. Be sure to leave secret column empty on asterisk database, otherwise all users registered on asterisks won't have OK status, what can cause problems with queues etc. 2015-08-12 0:19 GMT+03:00 Bruno d4rks...@gmail.com: Hello, i'm on my first try with kamailio. I need to build a SIP balancer that should keep SIP registration from VoIP provider and route the calls to the asterisk boxes where an IVR will take care to answer. Here's my network topology: +--- [asterisk1] [public_ip] |10.50.10.131 [router] ---NAT--- [kamailio] ---+ 10.50.10.110.50.10.120| +--- [asterisk2] 10.50.10.132 In my setup i planned to use UAC and DISPATCHER modules. I started from the kamailio-basic.cfg and added some extra lines to handle UAC and DISPATCHER. All is working fine when i do a test call from a softphone inside network 10.50.10.0/24. When a call is coming from the sip carrier, troubles occurs because asterisk boxes are sending their internal ip in SDP. I understand that i need to rewrite SDP in that case, but i actually don't know how/where. I've attached kamailio configuration and a sip trace taken with sngrep where the problem is visible. For security reasons, i would like to force the RTP through RTPProxy. I'm missing something, and need your help me to understand my errors. Best Regards, Bruno ___ 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 -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.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
[SR-Users] Help with sip balancer
Hello, i'm on my first try with kamailio. I need to build a SIP balancer that should keep SIP registration from VoIP provider and route the calls to the asterisk boxes where an IVR will take care to answer. Here's my network topology: +--- [asterisk1] [public_ip] |10.50.10.131 [router] ---NAT--- [kamailio] ---+ 10.50.10.110.50.10.120| +--- [asterisk2] 10.50.10.132 In my setup i planned to use UAC and DISPATCHER modules. I started from the kamailio-basic.cfg and added some extra lines to handle UAC and DISPATCHER. All is working fine when i do a test call from a softphone inside network 10.50.10.0/24. When a call is coming from the sip carrier, troubles occurs because asterisk boxes are sending their internal ip in SDP. I understand that i need to rewrite SDP in that case, but i actually don't know how/where. I've attached kamailio configuration and a sip trace taken with sngrep where the problem is visible. For security reasons, i would like to force the RTP through RTPProxy. I'm missing something, and need your help me to understand my errors. Best Regards, Bruno Call flow for 929B936-3FE5-9C28C7E1-A16E3F0E (Color by Request/Response) │SIP/2.0 200 OK 80.110.120.10:506010.50.10.120:5060 10.50.10.132:5060 │Via: SIP/2.0/UDP 10.50.10.120;branch=z9hG4bKe82d.a3414799f56e1046d9fede67b168a2ae.0;recei ──┬─ ──┬─ ──┬─ │d=10.50.10.120;rport=5060 05:37:29.196212 │ INV (80.110.120.12:57662) │ │ │Via: SIP/2.0/UDP 80.110.120.10:5060;rport=5060;branch=z9hG4bKe82d.0079d4c5.0 │ ── │ │ │Via: SIP/2.0/UDP 80.110.16.2:5060;rport=61413;received=80.110.16.2;x-route-tag=tgrp:Slot 05:37:29.204187 │ 100 trying -- your call is │ │ │;branch=z9hG4bKA4079B1AD1 │ ── │ │ │Record-Route: sip:10.50.10.120;lr=on;ftag=32CDDD90-24CE 05:37:29.205294 │ │ INV (80.110.120.12:57662) │ │Record-Route: sip:80.110.120.10;lr;ftag=32CDDD90-24CE;did=bb31.a983e793 │ │ ── │ │From: sip:8231288481@80.110.16.2;tag=32CDDD90-24CE 05:37:29.229975 │ │ 100 Trying │ │To: sip:9822147...@voip.carrier.me;tag=as355bc928 │ │ ── │ │Call-ID: 929B936-3FE5-9C28C7E1-A16E3F0E@80.110.16.2 05:37:29.233990 │ │ 200 (10.50.10.132:10832) │ │CSeq: 101 INVITE │ │ ── │ │Server: Asterisk PBX 13.1-cert2 05:37:29.235113 │ 200 (10.50.10.132:10832) │ │ │Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAG │ ── │ │ │Supported: replaces, timer 05:37:29.333226 │ │ 200 (10.50.10.132:10832) │ │Contact: sip:9822147941@10.50.10.132:5060 │ │ │ │Content-Type: application/sdp 05:37:29.335947 │ 200 (10.50.10.132:10832) │ │ │Content-Length: 347 │ │ │ │ 05:37:29.533537 │ │ 200 (10.50.10.132:10832) │ │v=0 │ │ │ │o=root 397373482 397373482 IN IP4 10.50.10.132 05:37:29.535938 │ 200 (10.50.10.132:10832) │ │ │s=Asterisk PBX 13.1-cert2 │ │ │ │c=IN IP4 10.50.10.132 05:37:29.934272 │ │ 200 (10.50.10.132:10832) │ │t=0 0 │ │ │ │m=audio 10832 RTP/AVP 3 18 8 0 101 05:37:29.935421 │ 200 (10.50.10.132:10832) │ │ │a=rtpmap:3 GSM/8000 │ │ │ │a=rtpmap:18 G729/8000 05:37:30.734051 │ │ 200
Re: [SR-Users] Help with sip balancer
Also, take a look at kamailio-advanced.cfg, there is PSTN GW route already included. Also you can use LCR for routing calls to different providers, a simple guide can be found here http://dopensource.com/least-cost-routing-with-kamailio-v4-1/ 2015-08-12 0:41 GMT+03:00 Alexandru Covalschi 568...@gmail.com: First of all I'd suggest to use http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb guide in combination with http://saevolgo.blogspot.com/2011/11/how-to-increasing-voip-services.html But, assuming your platform is behind NAT, you need: 1st. Use rtpengine instead of rtpproxy. You can read about how to advertise your external public adress on rtpengine git page. 2nd. In Kamailio configuration when you define listen, you should use listen - advertise construction ( http://www.kamailio.org/wiki/cookbooks/4.0.x/core#listen). 3d. Be sure to leave secret column empty on asterisk database, otherwise all users registered on asterisks won't have OK status, what can cause problems with queues etc. 2015-08-12 0:19 GMT+03:00 Bruno d4rks...@gmail.com: Hello, i'm on my first try with kamailio. I need to build a SIP balancer that should keep SIP registration from VoIP provider and route the calls to the asterisk boxes where an IVR will take care to answer. Here's my network topology: +--- [asterisk1] [public_ip] |10.50.10.131 [router] ---NAT--- [kamailio] ---+ 10.50.10.110.50.10.120| +--- [asterisk2] 10.50.10.132 In my setup i planned to use UAC and DISPATCHER modules. I started from the kamailio-basic.cfg and added some extra lines to handle UAC and DISPATCHER. All is working fine when i do a test call from a softphone inside network 10.50.10.0/24. When a call is coming from the sip carrier, troubles occurs because asterisk boxes are sending their internal ip in SDP. I understand that i need to rewrite SDP in that case, but i actually don't know how/where. I've attached kamailio configuration and a sip trace taken with sngrep where the problem is visible. For security reasons, i would like to force the RTP through RTPProxy. I'm missing something, and need your help me to understand my errors. Best Regards, Bruno ___ 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 -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/ -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.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] force_send_socket/$fs behaviour when binding to INADDR_ANY
Hello, inaddr any is special as a single socket receiving all traffic, but then it has its own limitations when attempting to send. Kamailio is building the list of sockets at startup and use that at runtime, so it is not possible to have a new socket added at runtime. However, depending on what exactly you do, there can be variants at ip/network layer. Keep adding new sockets to an application can become inefficient, not matter how it is done. So, as alternative, for example in the case of clients connecting via vpn tunnels, the best solution is to have kamailio listening on a single IP, then have masquerading of the tunnels to this ip. In other words, the clients will look at being behind nat from sip server point of view. The route to the IP of the sip server is pushed by th vpn client when connecting. The overall system has really low overhead, given that is practically ip forwarding. That was the example with vpn, if you have other scenario, maybe you can give more details in order to see if there are other solutions. Cheers, Daniel On 10/08/15 23:34, Alex Balashov wrote: Hello, When binding Kamailio to 0.0.0.0, Kamailio no longer recognises any specific IP address homed on the system as being a socket for purposes of forcing traffic out of any specific interface (i.e. setting $fs), or any other purpose for which ingress and egress sockets are tracked (e.g. double RR). As I understand it, this is because Kamailio considers only addresses explicitly specified via the 'listen' core config directive to be valid send socket arguments. Kamailio _will_ accept addresses already attached to the interface if it is bound interface-wise, e.g. listen=eth1:5060 but not when addresses are added to the interface without restarting the proxy: # ip addr add dev eth1 172.30.110.10/24 Aug 10 17:31:35 centosity6 /usr/local/sbin/kamailio[28627]: WARNING: pv [pv_core.c:2285]: pv_set_force_sock(): no socket found to match [udp:172.30.110.10:5060] Is there any straightforward way to modify this behaviour, so that it would be possible to dynamically add addresses to existing physical interfaces and get Kamailio to utilise them as if they were bound at boot time? Thanks, -- Alex -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - 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
[SR-Users] Trailing newline with {re.subst,expression}
I'm unable to get rid of the \n in multiline content of a var.
xlog(var: $var(sdp));
$var(sdp)=$(var(sdp){re.subst,/^a=rtcp:.*//});
xlog(var: $var(sdp));
a=sendrecv#015#012a=rtcp:31543#015#012a=X-foo:bar
becomes:
a=sendrecv#015#012#012a=X-foo:bar
Using the s flag results in no substitution at all, /^a=rtcp:.*//s should
result in
a=sendrecv#015#012
according to
http://www.kamailio.org/wiki/cookbooks/4.3.x/transformations#resubst_expression
Is this a bug or a feature? If it is a feature, how to get rid of the newline
(without additional regexps substitutions).
___
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] sdp manipulation in connection with rtpengine_offer/answer calls
Le Mon, 10 Aug 2015 18:33:47 +0200, Daniel Tryba d.tr...@pocos.nl a écrit : Only strange thing is that $var(sdp) is set to 0 on on the second passing of below during branch routing after handling a 302 redirect I don't know the full context here, but since $var()s are process-local, is the process reading the 0 value the same one that has set it? -- Camille ___ 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 Routing - Cost based routing
Hi Alexandru, Well the concept is simple: on INVITE, you will issue a special format evapi request. You will receive the list of suppliers in the reply (which is received in a separate route) and you continue processing from there. These suppliers are simple tags so your creativity will be the limit of what you can do with them - eg: direct ip addresses received or tags of suppliers which you can further translate with drouting module or any alias related one. For more information on LCR please find this year's presentation on Kamailio World (which I strongly recommend to be attended): https://www.youtube.com/watch?v=Hsvcwleb-fY You can also find the kamailio configuration which I have used in the worshop here: https://github.com/cgrates/cgrates/tree/master/data/tutorials/kamevapi/kamailio/etc/kamailio Let me know if further questions (or join our irc or mailing list so we do not create too much noise here). 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
Re: [SR-Users] kamailio as SIP Agent
Hi Sandeep,
what is the problem here ? Kamailio just sends a 404 on its own or is
really sending calls to MSC and MSC is replying with 404 ?
On Mon, Aug 10, 2015 at 12:33 PM, Sandeep Chakravarthi
ivschakravar...@gmail.com wrote:
Hi ,
Sorry for the delayed reply.
I have configured my Asterisk and kamailio server, but when i initiate one
outbound call from my asterisk server to kamailio server, kamailio server
is initiating the call to MSC.
Please find the attached pcap details for your reference.
Below is my kamailio debug log and kamailio.cfg file.
Please check the pcap and below cfg file and log file and let me know
whether to change anything in cfg file or not.
request_route {
# per request initial checks
route(REQINIT);
# NAT detection
route(NATDETECT);
# CANCEL processing
if (is_method(CANCEL))
{
if (t_check_trans()) {
route(RELAY);
}
exit;
}
# handle requests within SIP dialogs
route(WITHINDLG);
### only initial requests (no To tag)
t_check_trans();
# authentication
route(AUTH);
# record routing for dialog forming requests (in case they are
routed)
# - remove preloaded route headers
remove_hf(Route);
if (is_method(INVITE|SUBSCRIBE))
record_route();
# account only INVITEs
if (is_method(INVITE))
{
setflag(FLT_ACC); # do accounting
}
route(TOASTERISK);
# dispatch requests to foreign domains
route(SIPOUT);
### requests for my local domains
# handle presence related requests
route(PRESENCE);
# handle registrations
route(REGISTRAR);
if ($rU==$null)
{
# request with no Username in RURI
sl_send_reply(484,Address Incomplete);
exit;
}
# dispatch destinations to PSTN
route(PSTN);
# user location service
route(LOCATION);
}
route[TOASTERISK] {
if(ds_is_from_list(2)) {
#Call from Telco Should goto Asterisk pool in Loadbalanced mode
if(!ds_select_dst(1, 4)) {
sl_send_reply(500, Service Unavailable);
xlog(L_INFO,[$fU@$si:$sp]{$rm} No destinations
available for $rd \n);
exit;
}
}if(ds_is_from_list(1)) {
#Call from Asterisk servers pool, send it to telco using LoadBalancer
if(!ds_select_dst(2, 4)) {
sl_send_reply(500, Service Unavailable);
xlog(L_INFO,[$fU@$si:$sp]{$rm} No destinations
available for $rd \n);
exit;
}
}
}
+
Debug log
8(1186) DEBUG: core [parser/msg_parser.c:623]: parse_msg(): SIP Request:
8(1186) DEBUG: core [parser/msg_parser.c:625]: parse_msg(): method:
INVITE
8(1186) DEBUG: core [parser/msg_parser.c:627]: parse_msg(): uri:
sip:0730092190@172.22.14.12
8(1186) DEBUG: core [parser/msg_parser.c:629]: parse_msg(): version:
SIP/2.0
8(1186) DEBUG: core [parser/parse_via.c:1284]: parse_via_param(): Found
param type 232, branch = z9hG4bK3c5fb091; state=16
8(1186) DEBUG: core [parser/parse_via.c:2672]: parse_via(): end of
header reached, state=5
8(1186) DEBUG: core [parser/msg_parser.c:513]: parse_headers():
parse_headers: Via found, flags=2
8(1186) DEBUG: core [parser/msg_parser.c:515]: parse_headers():
parse_headers: this is the first via
8(1186) DEBUG: core [receive.c:152]: receive_msg(): After parse_msg...
8(1186) DEBUG: core [receive.c:193]: receive_msg(): preparing to run
routing scripts...
8(1186) DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 70
8(1186) DEBUG: core [parser/parse_addr_spec.c:898]: parse_addr_spec():
end of header reached, state=10
8(1186) DEBUG: core [parser/msg_parser.c:190]: get_hdr_field(): DEBUG:
get_hdr_field: To [31]; uri=[sip:0730092190@172.22.14.12]
8(1186) DEBUG: core [parser/msg_parser.c:192]: get_hdr_field(): DEBUG:
to body [sip:0730092190@172.22.14.12
]
8(1186) DEBUG: core [parser/msg_parser.c:170]: get_hdr_field():
get_hdr_field: cseq CSeq: 102 INVITE
8(1186) DEBUG: core [parser/msg_parser.c:204]: get_hdr_field(): DEBUG:
get_hdr_body : content_length=327
8(1186) DEBUG: core [parser/msg_parser.c:106]: get_hdr_field(): found
end of header
8(1186) DEBUG: core [parser/parse_addr_spec.c:176]: parse_to_param():
DEBUG: add_param: tag=as4decf975
8(1186) DEBUG: core [parser/parse_addr_spec.c:898]: parse_addr_spec():
end of header reached, state=29
8(1186) DEBUG: sanity [mod_sanity.c:255]: w_sanity_check(): sanity checks
result: 1
8(1186) DEBUG:
Re: [SR-Users] kamailio as SIP Agent
Hi,
Kamailio is sending 404 Response and its not MSC.
If you see the pcap file , Kamailio has to forward the SIP invite packet to
MSC which it got from Asterisk server. But it is not happening.
I am attaching the pcap one more time for your reference.
In my pcap, below are the server details
172.22.14.12 - Kamailio server
172.22.14.17 - Asterisk server
172.22.0.68 - MSC
Regards,
Sandeep
Warm Regards,
Sandeep Chakravarthi.
On Tue, Aug 11, 2015 at 7:10 PM, SamyGo govoi...@gmail.com wrote:
Hi Sandeep,
what is the problem here ? Kamailio just sends a 404 on its own or is
really sending calls to MSC and MSC is replying with 404 ?
On Mon, Aug 10, 2015 at 12:33 PM, Sandeep Chakravarthi
ivschakravar...@gmail.com wrote:
Hi ,
Sorry for the delayed reply.
I have configured my Asterisk and kamailio server, but when i initiate
one outbound call from my asterisk server to kamailio server, kamailio
server is initiating the call to MSC.
Please find the attached pcap details for your reference.
Below is my kamailio debug log and kamailio.cfg file.
Please check the pcap and below cfg file and log file and let me know
whether to change anything in cfg file or not.
request_route {
# per request initial checks
route(REQINIT);
# NAT detection
route(NATDETECT);
# CANCEL processing
if (is_method(CANCEL))
{
if (t_check_trans()) {
route(RELAY);
}
exit;
}
# handle requests within SIP dialogs
route(WITHINDLG);
### only initial requests (no To tag)
t_check_trans();
# authentication
route(AUTH);
# record routing for dialog forming requests (in case they are
routed)
# - remove preloaded route headers
remove_hf(Route);
if (is_method(INVITE|SUBSCRIBE))
record_route();
# account only INVITEs
if (is_method(INVITE))
{
setflag(FLT_ACC); # do accounting
}
route(TOASTERISK);
# dispatch requests to foreign domains
route(SIPOUT);
### requests for my local domains
# handle presence related requests
route(PRESENCE);
# handle registrations
route(REGISTRAR);
if ($rU==$null)
{
# request with no Username in RURI
sl_send_reply(484,Address Incomplete);
exit;
}
# dispatch destinations to PSTN
route(PSTN);
# user location service
route(LOCATION);
}
route[TOASTERISK] {
if(ds_is_from_list(2)) {
#Call from Telco Should goto Asterisk pool in Loadbalanced mode
if(!ds_select_dst(1, 4)) {
sl_send_reply(500, Service Unavailable);
xlog(L_INFO,[$fU@$si:$sp]{$rm} No
destinations available for $rd \n);
exit;
}
}if(ds_is_from_list(1)) {
#Call from Asterisk servers pool, send it to telco using LoadBalancer
if(!ds_select_dst(2, 4)) {
sl_send_reply(500, Service Unavailable);
xlog(L_INFO,[$fU@$si:$sp]{$rm} No
destinations available for $rd \n);
exit;
}
}
}
+
Debug log
8(1186) DEBUG: core [parser/msg_parser.c:623]: parse_msg(): SIP
Request:
8(1186) DEBUG: core [parser/msg_parser.c:625]: parse_msg(): method:
INVITE
8(1186) DEBUG: core [parser/msg_parser.c:627]: parse_msg(): uri:
sip:0730092190@172.22.14.12
8(1186) DEBUG: core [parser/msg_parser.c:629]: parse_msg(): version:
SIP/2.0
8(1186) DEBUG: core [parser/parse_via.c:1284]: parse_via_param():
Found param type 232, branch = z9hG4bK3c5fb091; state=16
8(1186) DEBUG: core [parser/parse_via.c:2672]: parse_via(): end of
header reached, state=5
8(1186) DEBUG: core [parser/msg_parser.c:513]: parse_headers():
parse_headers: Via found, flags=2
8(1186) DEBUG: core [parser/msg_parser.c:515]: parse_headers():
parse_headers: this is the first via
8(1186) DEBUG: core [receive.c:152]: receive_msg(): After parse_msg...
8(1186) DEBUG: core [receive.c:193]: receive_msg(): preparing to run
routing scripts...
8(1186) DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 70
8(1186) DEBUG: core [parser/parse_addr_spec.c:898]: parse_addr_spec():
end of header reached, state=10
8(1186) DEBUG: core [parser/msg_parser.c:190]: get_hdr_field(): DEBUG:
get_hdr_field: To [31]; uri=[sip:0730092190@172.22.14.12]
8(1186) DEBUG: core [parser/msg_parser.c:192]: get_hdr_field(): DEBUG:
to body [sip:0730092190@172.22.14.12
]
8(1186) DEBUG: core [parser/msg_parser.c:170]: get_hdr_field():
get_hdr_field: cseq CSeq: 102 INVITE
8(1186)
Re: [SR-Users] kamailio as SIP Agent
Thats because your configuration file is not sending packet out (RELAY) to
MSC instead it is only doing a Loadbalancer / destination lookup in
TOASTERISK route and comes out of it, processes the following routes in
order
route(SIPOUT);
route(PRESENCE);
route(REGISTRAR);
route(PSTN);
route(LOCATION);
Where finally in LOCATION route it tries to find the destination user
0730092190 online locally on Kamailio, which it can't find and says 404 Not
Found.
You should modify your TOASTERISK route as follow:
route[TOASTERISK] {
if(ds_is_from_list(2)) {
#Call from Telco Should goto Asterisk pool in Loadbalanced mode
if(!ds_select_dst(1, 4)) {
sl_send_reply(500, Service Unavailable);
xlog(L_INFO,[$fU@$si:$sp]{$rm} No destinations
available for $rd \n);
exit;
}
route(RELAY);
}if(ds_is_from_list(1)) {
#Call from Asterisk servers pool, send it to telco using LoadBalancer
if(!ds_select_dst(2, 4)) {
sl_send_reply(500, Service Unavailable);
xlog(L_INFO,[$fU@$si:$sp]{$rm} No destinations
available for $rd \n);
exit;
}
route(RELAY);
}
}
This will immediately route the packet out towards the new $du after the
loadbalancer function ds_select_dst(...)
On Tue, Aug 11, 2015 at 10:48 AM, Sandeep Chakravarthi
ivschakravar...@gmail.com wrote:
Hi,
Kamailio is sending 404 Response and its not MSC.
If you see the pcap file , Kamailio has to forward the SIP invite packet
to MSC which it got from Asterisk server. But it is not happening.
I am attaching the pcap one more time for your reference.
In my pcap, below are the server details
172.22.14.12 - Kamailio server
172.22.14.17 - Asterisk server
172.22.0.68 - MSC
Regards,
Sandeep
Warm Regards,
Sandeep Chakravarthi.
On Tue, Aug 11, 2015 at 7:10 PM, SamyGo govoi...@gmail.com wrote:
Hi Sandeep,
what is the problem here ? Kamailio just sends a 404 on its own or is
really sending calls to MSC and MSC is replying with 404 ?
On Mon, Aug 10, 2015 at 12:33 PM, Sandeep Chakravarthi
ivschakravar...@gmail.com wrote:
Hi ,
Sorry for the delayed reply.
I have configured my Asterisk and kamailio server, but when i initiate
one outbound call from my asterisk server to kamailio server, kamailio
server is initiating the call to MSC.
Please find the attached pcap details for your reference.
Below is my kamailio debug log and kamailio.cfg file.
Please check the pcap and below cfg file and log file and let me know
whether to change anything in cfg file or not.
request_route {
# per request initial checks
route(REQINIT);
# NAT detection
route(NATDETECT);
# CANCEL processing
if (is_method(CANCEL))
{
if (t_check_trans()) {
route(RELAY);
}
exit;
}
# handle requests within SIP dialogs
route(WITHINDLG);
### only initial requests (no To tag)
t_check_trans();
# authentication
route(AUTH);
# record routing for dialog forming requests (in case they are
routed)
# - remove preloaded route headers
remove_hf(Route);
if (is_method(INVITE|SUBSCRIBE))
record_route();
# account only INVITEs
if (is_method(INVITE))
{
setflag(FLT_ACC); # do accounting
}
route(TOASTERISK);
# dispatch requests to foreign domains
route(SIPOUT);
### requests for my local domains
# handle presence related requests
route(PRESENCE);
# handle registrations
route(REGISTRAR);
if ($rU==$null)
{
# request with no Username in RURI
sl_send_reply(484,Address Incomplete);
exit;
}
# dispatch destinations to PSTN
route(PSTN);
# user location service
route(LOCATION);
}
route[TOASTERISK] {
if(ds_is_from_list(2)) {
#Call from Telco Should goto Asterisk pool in Loadbalanced mode
if(!ds_select_dst(1, 4)) {
sl_send_reply(500, Service Unavailable);
xlog(L_INFO,[$fU@$si:$sp]{$rm} No
destinations available for $rd \n);
exit;
}
}if(ds_is_from_list(1)) {
#Call from Asterisk servers pool, send it to telco using LoadBalancer
if(!ds_select_dst(2, 4)) {
sl_send_reply(500, Service Unavailable);
xlog(L_INFO,[$fU@$si:$sp]{$rm} No
destinations available for $rd \n);
exit;
}
}
}
Re: [SR-Users] sdp manipulation in connection with rtpengine_offer/answer calls
Le Mon, 10 Aug 2015 16:25:36 +0300, Juha Heinanen j...@tutpro.com a écrit : i think it would be useful especially if sdpops functions could also be made operate on the pv. then there would no need to call msg_apply_changes(). the pv could be initialized from $rb and when all calls that manipulate the body have been made, the pv would be assigned to $rb. Hi, i've created a merge request on the master branch to add the rtpengine input SDP pv. I'm not sure if modifying sdpops to work also on variables would be straightforward, but anyway, most operations can be done using transformations. -- Camille ___ 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 SIP Agent
1 - Take a look at the Kamailio transformations and psuedo-variable page.
change the $td to the IP of the MSC; modify the $ru as $rU + @
172.22.12.100:5060 where this is IP of MSC side.
2 - Wireshark guys could've said it SIP-3 - point is it doesnt matter at
this point since you know your MSC is replying back and talking to you.
On Tue, Aug 11, 2015 at 1:16 PM, Sandeep Chakravarthi
ivschakravar...@gmail.com wrote:
Yes, You are right and done the changes as you suggested.
Kamailio server is forwarding the call to MSC. But two issues are there.
1 .In the INVITE packet which is being sent from kamailio server to MSC,
it is coming Request-Line: INVITE sip:0730092190@*172.22.14.12*
That is my kamailio server IP and it should be MSC IP(172.28.0.68) and
as of now call is failing as MSC is sending 404 error.
2. Other issue is , in the pcap file it is coming SIP/SDP as protocol and
it is not coming SIP-I.
Please find the latest attached pcap.
Regards,
Sandeep
Warm Regards,
Sandeep Chakravarthi.
On Tue, Aug 11, 2015 at 9:47 PM, SamyGo govoi...@gmail.com wrote:
Thats because your configuration file is not sending packet out (RELAY)
to MSC instead it is only doing a Loadbalancer / destination lookup in
TOASTERISK route and comes out of it, processes the following routes in
order
route(SIPOUT);
route(PRESENCE);
route(REGISTRAR);
route(PSTN);
route(LOCATION);
Where finally in LOCATION route it tries to find the destination user
0730092190 online locally on Kamailio, which it can't find and says 404 Not
Found.
You should modify your TOASTERISK route as follow:
route[TOASTERISK] {
if(ds_is_from_list(2)) {
#Call from Telco Should goto Asterisk pool in Loadbalanced mode
if(!ds_select_dst(1, 4)) {
sl_send_reply(500, Service Unavailable);
xlog(L_INFO,[$fU@$si:$sp]{$rm} No
destinations available for $rd \n);
exit;
}
route(RELAY);
}if(ds_is_from_list(1)) {
#Call from Asterisk servers pool, send it to telco using LoadBalancer
if(!ds_select_dst(2, 4)) {
sl_send_reply(500, Service Unavailable);
xlog(L_INFO,[$fU@$si:$sp]{$rm} No
destinations available for $rd \n);
exit;
}
route(RELAY);
}
}
This will immediately route the packet out towards the new $du after the
loadbalancer function ds_select_dst(...)
On Tue, Aug 11, 2015 at 10:48 AM, Sandeep Chakravarthi
ivschakravar...@gmail.com wrote:
Hi,
Kamailio is sending 404 Response and its not MSC.
If you see the pcap file , Kamailio has to forward the SIP invite packet
to MSC which it got from Asterisk server. But it is not happening.
I am attaching the pcap one more time for your reference.
In my pcap, below are the server details
172.22.14.12 - Kamailio server
172.22.14.17 - Asterisk server
172.22.0.68 - MSC
Regards,
Sandeep
Warm Regards,
Sandeep Chakravarthi.
On Tue, Aug 11, 2015 at 7:10 PM, SamyGo govoi...@gmail.com wrote:
Hi Sandeep,
what is the problem here ? Kamailio just sends a 404 on its own or is
really sending calls to MSC and MSC is replying with 404 ?
On Mon, Aug 10, 2015 at 12:33 PM, Sandeep Chakravarthi
ivschakravar...@gmail.com wrote:
Hi ,
Sorry for the delayed reply.
I have configured my Asterisk and kamailio server, but when i initiate
one outbound call from my asterisk server to kamailio server, kamailio
server is initiating the call to MSC.
Please find the attached pcap details for your reference.
Below is my kamailio debug log and kamailio.cfg file.
Please check the pcap and below cfg file and log file and let me know
whether to change anything in cfg file or not.
request_route {
# per request initial checks
route(REQINIT);
# NAT detection
route(NATDETECT);
# CANCEL processing
if (is_method(CANCEL))
{
if (t_check_trans()) {
route(RELAY);
}
exit;
}
# handle requests within SIP dialogs
route(WITHINDLG);
### only initial requests (no To tag)
t_check_trans();
# authentication
route(AUTH);
# record routing for dialog forming requests (in case they are
routed)
# - remove preloaded route headers
remove_hf(Route);
if (is_method(INVITE|SUBSCRIBE))
record_route();
# account only INVITEs
if (is_method(INVITE))
{
setflag(FLT_ACC); # do accounting
}
route(TOASTERISK);
# dispatch requests to foreign domains
route(SIPOUT);
### requests for my local domains
# handle presence related requests
Re: [SR-Users] kamailio as SIP Agent
Yes, You are right and done the changes as you suggested.
Kamailio server is forwarding the call to MSC. But two issues are there.
1 .In the INVITE packet which is being sent from kamailio server to MSC, it
is coming Request-Line: INVITE sip:0730092190@*172.22.14.12*
That is my kamailio server IP and it should be MSC IP(172.28.0.68) and
as of now call is failing as MSC is sending 404 error.
2. Other issue is , in the pcap file it is coming SIP/SDP as protocol and
it is not coming SIP-I.
Please find the latest attached pcap.
Regards,
Sandeep
Warm Regards,
Sandeep Chakravarthi.
On Tue, Aug 11, 2015 at 9:47 PM, SamyGo govoi...@gmail.com wrote:
Thats because your configuration file is not sending packet out (RELAY) to
MSC instead it is only doing a Loadbalancer / destination lookup in
TOASTERISK route and comes out of it, processes the following routes in
order
route(SIPOUT);
route(PRESENCE);
route(REGISTRAR);
route(PSTN);
route(LOCATION);
Where finally in LOCATION route it tries to find the destination user
0730092190 online locally on Kamailio, which it can't find and says 404 Not
Found.
You should modify your TOASTERISK route as follow:
route[TOASTERISK] {
if(ds_is_from_list(2)) {
#Call from Telco Should goto Asterisk pool in Loadbalanced mode
if(!ds_select_dst(1, 4)) {
sl_send_reply(500, Service Unavailable);
xlog(L_INFO,[$fU@$si:$sp]{$rm} No destinations
available for $rd \n);
exit;
}
route(RELAY);
}if(ds_is_from_list(1)) {
#Call from Asterisk servers pool, send it to telco using LoadBalancer
if(!ds_select_dst(2, 4)) {
sl_send_reply(500, Service Unavailable);
xlog(L_INFO,[$fU@$si:$sp]{$rm} No destinations
available for $rd \n);
exit;
}
route(RELAY);
}
}
This will immediately route the packet out towards the new $du after the
loadbalancer function ds_select_dst(...)
On Tue, Aug 11, 2015 at 10:48 AM, Sandeep Chakravarthi
ivschakravar...@gmail.com wrote:
Hi,
Kamailio is sending 404 Response and its not MSC.
If you see the pcap file , Kamailio has to forward the SIP invite packet
to MSC which it got from Asterisk server. But it is not happening.
I am attaching the pcap one more time for your reference.
In my pcap, below are the server details
172.22.14.12 - Kamailio server
172.22.14.17 - Asterisk server
172.22.0.68 - MSC
Regards,
Sandeep
Warm Regards,
Sandeep Chakravarthi.
On Tue, Aug 11, 2015 at 7:10 PM, SamyGo govoi...@gmail.com wrote:
Hi Sandeep,
what is the problem here ? Kamailio just sends a 404 on its own or is
really sending calls to MSC and MSC is replying with 404 ?
On Mon, Aug 10, 2015 at 12:33 PM, Sandeep Chakravarthi
ivschakravar...@gmail.com wrote:
Hi ,
Sorry for the delayed reply.
I have configured my Asterisk and kamailio server, but when i initiate
one outbound call from my asterisk server to kamailio server, kamailio
server is initiating the call to MSC.
Please find the attached pcap details for your reference.
Below is my kamailio debug log and kamailio.cfg file.
Please check the pcap and below cfg file and log file and let me know
whether to change anything in cfg file or not.
request_route {
# per request initial checks
route(REQINIT);
# NAT detection
route(NATDETECT);
# CANCEL processing
if (is_method(CANCEL))
{
if (t_check_trans()) {
route(RELAY);
}
exit;
}
# handle requests within SIP dialogs
route(WITHINDLG);
### only initial requests (no To tag)
t_check_trans();
# authentication
route(AUTH);
# record routing for dialog forming requests (in case they are
routed)
# - remove preloaded route headers
remove_hf(Route);
if (is_method(INVITE|SUBSCRIBE))
record_route();
# account only INVITEs
if (is_method(INVITE))
{
setflag(FLT_ACC); # do accounting
}
route(TOASTERISK);
# dispatch requests to foreign domains
route(SIPOUT);
### requests for my local domains
# handle presence related requests
route(PRESENCE);
# handle registrations
route(REGISTRAR);
if ($rU==$null)
{
# request with no Username in RURI
sl_send_reply(484,Address Incomplete);
exit;
}
# dispatch destinations to PSTN
route(PSTN);
# user location service
route(LOCATION);
}
route[TOASTERISK] {
if(ds_is_from_list(2)) {
#Call from Telco
