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
The Tag column size is 64. If I make this larger in the database, will it be truncated once it is loaded into memory? 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? 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? 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. From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Vitaliy Aleksandrov Sent: Wednesday, April 29, 2015 12:27 PM To: sr-users@lists.sip-router.org Subject: 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.orgmailto: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.orgmailto: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.orgmailto: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
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? How fast does it select a gateway after load_gws using next_gw()? Is it O(N) again? 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? 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.orgmailto: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] Measuring subscriber latency
Hello, On 28/04/15 20:42, Jon Bonilla (Manwe) wrote: Hi all I'm replacing an Asterisk based system with a kamailio based one. One of the features the legacy system has is showing the subscriber the latency obtained from the qualify option of sip.conf Now, I'd like to measure the latency but I'm not sure how to do it. AFAIK nathelper module sends the OPTIONS keepalive messages stateless mode and there's no information there. I was thinking on triggering a route send_options via the timer module, save the timestamp of the relay and the timestamp of the response in onreply_route but it doesn't look elegant. Creating my own daemon in an external server and reading the info from the location module seems to be another option. I guess I'm not the first one with this need so I wonder if there's an already existing solution or an elegant way of dealing with it. any ideas? it looks like you are the first wanting this, or at least the first that has expressed it. It might not be hard to code it in c, it will require to extend the usrloc structure to have with two timestamps, one to be set when sending the options and one when receiving the reply. At this moment there is only one timestamp when the reply is received, see the function ul_refresh_keepalive() from usrloc, the field in ucontact_t structure is last_keepalive. So half is done more or less. You can open a feature request on tracker and probably will get into 4.4. The development for 4.3 is frozen. 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
Re: [SR-Users] Measuring subscriber latency
On 29 Apr 2015, at 10:04, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, On 28/04/15 20:42, Jon Bonilla (Manwe) wrote: Hi all I'm replacing an Asterisk based system with a kamailio based one. One of the features the legacy system has is showing the subscriber the latency obtained from the qualify option of sip.conf Now, I'd like to measure the latency but I'm not sure how to do it. AFAIK nathelper module sends the OPTIONS keepalive messages stateless mode and there's no information there. I was thinking on triggering a route send_options via the timer module, save the timestamp of the relay and the timestamp of the response in onreply_route but it doesn't look elegant. Creating my own daemon in an external server and reading the info from the location module seems to be another option. I guess I'm not the first one with this need so I wonder if there's an already existing solution or an elegant way of dealing with it. any ideas? it looks like you are the first wanting this, or at least the first that has expressed it. As Jon said, this is a feature that has been in Asterisk for a very long time and we need in Kamailio. I think many of us has looked for it, but never mailed about it since we still have Asterisk in there. Since Kamailio has grown so much and we now can build Asterisk-free solutions, I think this would be a valuable feature both for dispatcher and for usrloc. /O It might not be hard to code it in c, it will require to extend the usrloc structure to have with two timestamps, one to be set when sending the options and one when receiving the reply. At this moment there is only one timestamp when the reply is received, see the function ul_refresh_keepalive() from usrloc, the field in ucontact_t structure is last_keepalive. So half is done more or less. You can open a feature request on tracker and probably will get into 4.4. The development for 4.3 is frozen. 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] 2 phones behind same NAT, Kamailio on Pub IP One Way Audio
Hello, if both phones are behind same nat and no nat processing enabled in kamailio.cfg, the audio should work fine. Be sure there is no ALG in your nat router or firewall. The best is to look at singaling, on the server you can use: ngrep -d any -qt -W byline sip port 5060 If the phones are behind nat, you need to enable nat traversal in kamailio config by defining WITH_NAT, install it and configure to listen for control commands on the socket specified to rtpproxy kamailio module parameters (see the comments at the top of default kamailio.cfg). Cheers, Daniel On 29/04/15 05:03, Todd R. wrote: This is my first go round' with Kamailio but I have been messing with it off and on for a few weeks. Running latest version on latest CentOS 7.x. Kamailio installed on VM with public IP in one location, 2 SIP phones behind same NAT, both registered to Kamailio fine. I can call one phone to the other and it rings and I can answer it but get one way audio most of the time, sometimes no audio at all. I just did a standard install with MySQL and no other modules. What am I missing, do I need RTPPROXY? Will Kamailio allow extension to extension calls from/to phones behind NATS without any additional modules? I tried installing it with yum, it installed but it won't start. If I need it, I will remove the YUM version and install from source or GIT or some other method. I do NOT want any media passing through this server, that's one big reason I am learning something OTHER than Asterisk which I am very familiar with. I see all these RTP modules, music on hold etc but I specifically don't want to install anything that causes media to pass through the box. Finding the learning curve much steeper than Asterisk back in the day and a REALLY tough time find step by step examples to get started, especially on CentOS. I bought the draft of the book and it will be a great resource but at the moment, it's not helping me get started or get my first instance up and running. I need to add SIP trunks to originate and terminate calls but I can't even think about that since I can't even get audio on extension to extension calls yet. I also see that Kamailio fails to startup at boot because it can't connect to MySQL, when I restart Kamailio it starts fine. I think the issue is that Kamailio is trying to start before MySQL is up and running, I guess I can do a delayed start of Kamailio to fix that later. Any help would be appreciate by this NEWB. Thanks in advance for any assistance. ___ 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://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
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] Measuring subscriber latency
Am 29.04.2015 um 10:21 schrieb Olle E. Johansson: On 29 Apr 2015, at 10:04, Daniel-Constantin Mierla mico...@gmail.com wrote: it looks like you are the first wanting this, or at least the first that has expressed it. As Jon said, this is a feature that has been in Asterisk for a very long time and we need in Kamailio. I think many of us has looked for it, but never mailed about it since we still have Asterisk in there. Since Kamailio has grown so much and we now can build Asterisk-free solutions, I think this would be a valuable feature both for dispatcher and for usrloc. Agreed, this is a nice feature to have. -Sven signature.asc Description: OpenPGP digital signature ___ 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] Running Kamailio with IMS
Hello, On 29/04/15 10:06, Mihail Dakov wrote: Hi all, I am trying to run Kamailio server with IMS modules for which I have gone through the following tutorials: http://nil.uniza.sk/ngnims/kamailio-ims/preparing-debian-operating-system-kamailio-4x-platform http://www.kamailio.org/wiki/tutorials/ims/installation-howto The problem I have is that using 'service kamailio start' I can only run one IMS component, say PCSCF. But I need to run all P/S/I-CSCFs. How do I proceed to get running P/S/I-CSCF on the same machine with kamailio? you can run multiple instances of kamailio on the same system, just use different ports for each instance. Or, if you can allocate many IP addresses to the system, you can run one per IP. Just specify the ip and port to listen on via global parameter: listen=1.2.3.4:5080 If you questions is about combining all those configs, perhaps is possible, but not something I can assist with. 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
[SR-Users] sht_rm_name_re() question
Hi I've hit a problem with sht_rm_name_re() in htable module. I was calling it like this: sht_rm_name_re(Dlg=$var(callid)::tenant); But when I used sipp to generate 600 concurrent calls for example, I called this function when receiving BYE. But it removed more entries than it should. Seems when removing removing some entry with callid 1-3798@192.168.60.80, it also removed entries for [1-9]1-3798@192.168.60.80. How do I add '^' in front and '$' at the end of the regexp string when calling this function please? Here I don't really need regexp in fact, just want to do an exact match. But I didn't find any other functions for deleting entries. Thank you! Yufei ___ 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] UAC Module
Hi all. I have this setup. Trunk---KamailioFreeSWITCH I have a trunk from a sip provided and registered successfully with the UAC module. Incoming is working fine. I need to make out going through kamailio too. I have it in the dialplan to forward the invite to kamailio from FreeSWITCH. I can see it the logs that it reaches kamailio. Now how do I make the call via the trunk? Basically this is what I'm trying to workout FSkamailiotrunk. Any help will be much appreciated. Thanks. AJ ___ 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] issue with TLS and 2 NIC interfaces
Hello, if i have two interfaces (eth0 and eth0:0) I set kamailio to listen to the IP on eth0:0 This works great except when I try using TLS, kamailio routes traffic from eth0:0 to eth0 then to correct destination It should be just doing eth0:0 - correct destination If I use UDP/TCP I am not seeing this behavior. If I set kamailio to listen to eth0 and remove eth0;0, TLS works fine. I'm running kamailio compiled from source git master a few days ago. Is anyone else having this problem? I'm wondering if it's a bug. Thanks, V ___ 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] [SR_USers] Authenticate asterisk-kamailio
Hello All.
My English is bad so I hope you can understand.
I have been working with Kamailio some time, I following some of Guides of
http://kb.asipto.com, http://saevolgo.blogspot.com and http://nil.uniza.sk.
To Authenticate Asterisk sipusers I'm not have problems, but subscriber of
Kamailio authenticate is my problem.
Autenticate are denied to subscriber Kamailio.
In spanish I can explain it better :-)
best regards
my Settings:
###
# - auth_db params -
#!ifdef WITH_AUTH
modparam(auth_db, calculate_ha1, yes)
modparam(auth_db, load_credentials, )
#!ifdef WITH_ASTERISK
modparam(auth_db, user_column, name)
modparam(auth_db, password_column, sippasswd)
modparam(auth_db, db_url, DBASTURL)
modparam(auth_db, version_table, 0)
#!else
modparam(auth_db, db_url, DBURL)
modparam(auth_db, password_column, password)
modparam(auth_db, use_domain, MULTIDOMAIN)
#!endif
#!ifdef WITH_AUTH
#!ifdef WITH_ASTERISK
# do not auth traffic from Asterisk - trusted!
if(route(FROMASTERISK))
return;
#!endif
#!ifdef WITH_IPAUTH
if((!is_method(REGISTER)) allow_source_address())
{
# source IP allowed
return;
}
#!endif
if (is_method(REGISTER) || from_uri==myself)
{
# authenticate requests
#!ifdef WITH_ASTERISK
if (!auth_check($fd, sipusers, 1)) {
Aquí autentifica sin problemas en caso de ser un usuario asterisk
#!else
Pero hace caso omiso al switch.
if (!auth_check($fd, subscriber, 1)) {
No autentifica subscriber de la base de datos de kamailio
#!endif
auth_challenge($fd, 0);
exit;
}
# user authenticated - remove auth header
if(!is_method(REGISTER|PUBLISH))
consume_credentials();
}
#!endif
return;
}
#
DEBUG KAMAILIO
Apr 29 10:40:48 labkamailio kamailio[11304]: DEBUG: core
[parser/msg_parser.c:623]: parse_msg(): SIP Request:
Apr 29 10:40:48 labkamailio kamailio[11304]: DEBUG: core
[parser/msg_parser.c:625]: parse_msg(): method: REGISTER
Apr 29 10:40:48 labkamailio kamailio[11304]: DEBUG: core
[parser/msg_parser.c:627]: parse_msg(): uri: sip:192.168.65.132
Apr 29 10:40:48 labkamailio kamailio[11304]: DEBUG: core
[parser/msg_parser.c:629]: parse_msg(): version: SIP/2.0
Apr 29 10:40:48 labkamailio kamailio[11304]: DEBUG: core
[parser/parse_addr_spec.c:898]: parse_addr_spec(): end of header reached,
state=10
Apr 29 10:40:48 labkamailio kamailio[11304]: DEBUG: core
[parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: To [29];
uri=[sip:107@192.168.65.132]
Apr 29 10:40:48 labkamailio kamailio[11304]: DEBUG: core
[parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body
[107sip:107@192.168.65.132#015#012]
Apr 29 10:40:48 labkamailio kamailio[11304]: DEBUG: core
[parser/parse_via.c:1284]: parse_via_param(): Found param type 232, branch
= z9hG4bK-d87543-733510592-1--d87543-; state=6
Apr 29 10:40:48 labkamailio kamailio[11304]: DEBUG: core
[parser/parse_via.c:1284]: parse_via_param(): Found param type 235, rport
= n/a; state=17
Apr 29 10:40:48 labkamailio kamailio[11304]: DEBUG: core
[parser/parse_via.c:2672]: parse_via(): end of header reached, state=5
Apr 29 10:40:48 labkamailio kamailio[11304]: DEBUG: core
[parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found,
flags=2
Apr 29 10:40:48 labkamailio kamailio[11304]: DEBUG: core
[parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first
via
Apr 29 10:40:48 labkamailio kamailio[11304]: DEBUG: core [receive.c:154]:
receive_msg(): After parse_msg...
Apr 29 10:40:48 labkamailio kamailio[11304]: DEBUG: core [receive.c:197]:
receive_msg(): preparing to run routing scripts...
Apr 29 10:40:48 labkamailio kamailio[11304]: DEBUG: core
[parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq CSeq: 1
REGISTER
Apr 29 10:40:48 labkamailio kamailio[11304]: DEBUG: maxfwd [mf_funcs.c:85]:
is_maxfwd_present(): value = 70
Apr 29 10:40:48 labkamailio kamailio[11304]: DEBUG: core
[parser/msg_parser.c:204]: get_hdr_field(): DEBUG: get_hdr_body :
content_length=0
Apr 29 10:40:48 labkamailio kamailio[11304]: DEBUG: core
[parser/msg_parser.c:106]: get_hdr_field(): found end of header
Apr 29 10:40:48 labkamailio kamailio[11304]: DEBUG: core
[parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param:
tag=306ac106
Apr 29 10:40:48 labkamailio kamailio[11304]: DEBUG: core
Re: [SR-Users] Running Kamailio with IMS
Mihail, Are you able to share your configurations for each of the elements (P-CSCF, S-CSCF and I-CSCF) ? Cheers, Abdul Hakeem -Original Message- From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Mihail Dakov Sent: Wednesday, April 29, 2015 1:03 PM To: mico...@gmail.com; Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Running Kamailio with IMS Hi All, I solved the issue by compiling kamailio from src and running different instances for each component P-CSCF, S-CSCF, and I-CSCF with different configuration respectively. Thank you all for the support. On 04/29/2015 12:04 PM, Daniel-Constantin Mierla wrote: Hello, On 29/04/15 10:06, Mihail Dakov wrote: Hi all, I am trying to run Kamailio server with IMS modules for which I have gone through the following tutorials: http://nil.uniza.sk/ngnims/kamailio-ims/preparing-debian-operating-system-kamail io-4x-platform http://www.kamailio.org/wiki/tutorials/ims/installation-howto The problem I have is that using 'service kamailio start' I can only run one IMS component, say PCSCF. But I need to run all P/S/I-CSCFs. How do I proceed to get running P/S/I-CSCF on the same machine with kamailio? you can run multiple instances of kamailio on the same system, just use different ports for each instance. Or, if you can allocate many IP addresses to the system, you can run one per IP. Just specify the ip and port to listen on via global parameter: listen=1.2.3.4:5080 If you questions is about combining all those configs, perhaps is possible, but not something I can assist with. 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] Running Kamailio with IMS
Hi Abdul, I followed this tutorial: https://loadmultiplier.com/node/76 and the configuration is the same but ips, ports and aliases. Have you tried that? br, m.dakov On 04/29/2015 04:07 PM, Abdul Hakeem wrote: Mihail, Are you able to share your configurations for each of the elements (P-CSCF, S-CSCF and I-CSCF) ? Cheers, Abdul Hakeem -Original Message- From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Mihail Dakov Sent: Wednesday, April 29, 2015 1:03 PM To: mico...@gmail.com; Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Running Kamailio with IMS Hi All, I solved the issue by compiling kamailio from src and running different instances for each component P-CSCF, S-CSCF, and I-CSCF with different configuration respectively. Thank you all for the support. On 04/29/2015 12:04 PM, Daniel-Constantin Mierla wrote: Hello, On 29/04/15 10:06, Mihail Dakov wrote: Hi all, I am trying to run Kamailio server with IMS modules for which I have gone through the following tutorials: http://nil.uniza.sk/ngnims/kamailio-ims/preparing-debian-operating-system-kamail io-4x-platform http://www.kamailio.org/wiki/tutorials/ims/installation-howto The problem I have is that using 'service kamailio start' I can only run one IMS component, say PCSCF. But I need to run all P/S/I-CSCFs. How do I proceed to get running P/S/I-CSCF on the same machine with kamailio? you can run multiple instances of kamailio on the same system, just use different ports for each instance. Or, if you can allocate many IP addresses to the system, you can run one per IP. Just specify the ip and port to listen on via global parameter: listen=1.2.3.4:5080 If you questions is about combining all those configs, perhaps is possible, but not something I can assist with. 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] UAC Module
Hi Jibran,
Here is an old thread as reference:
http://lists.sip-router.org/pipermail/sr-users/2013-August/079336.html
I wouldn't want to do the whole handshake of INVITE,PROXY-AUTH,INVITE with
username/password on a Provider for huge number of calls..imagine sending
thousands of call to that provider and for each call going through the
trouble of exchanging authentication.
Thats why its usually recommended to go with IP-Authentication only. Send
INVITE and Provider says Lets do this call,simple and easy.
From the configuration perspective this is my idea of still using UAC.
- Call coming from FS on kamailio
- Rewrite the from-uri (so the provider receives calls from the registered
username)
- modify the to-domain part to contain the IP address of the provider
- set the $du to ip of the provider, and t_relay() the call.
- Most likely the Provider would say Proxy-Auth required..that can be
caught in failure_route[]
- There you can call the uac_auth() function to have username.password
attached to the response of above.
http://kamailio.org/docs/modules/4.3.x/modules/uac.html#uac.f.uac_auth()
- once this function is successful send the INVITE again to the provider.
Last three steps can be the following snippet of code(reference from here
http://opensips.org/pipermail/users/2010-August/013947.html):
failure_route[2] {
if (t_check_status(40[17])) {
xlog(got challenged \n);
if (uac_auth()) {
xlog(auth was succesful \n);
t_relay(udp:ip_addr:5060); //provider's IP_ADDR
}
}
I hope you get IP Auth from the provider, and find the reply useful.
Regards,
On Wed, Apr 29, 2015 at 4:49 PM, Ali Jibran alijib...@vividtech.io wrote:
Hi all.
I have this setup.
Trunk---KamailioFreeSWITCH
I have a trunk from a sip provided and registered successfully with the
UAC module. Incoming is working fine. I need to make out going through
kamailio too.
I have it in the dialplan to forward the invite to kamailio from
FreeSWITCH. I can see it the logs that it reaches kamailio. Now how do I
make the call via the trunk?
Basically this is what I'm trying to workout
FSkamailiotrunk.
Any help will be much appreciated. Thanks.
AJ
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