Re: [SR-Users] Kam 4.1 Crash: HELP PLZ!!!!

[Daniel-Constantin Mierla]

Are the error messages before those with realloc  related to failing 
sending on branch 0? Any error messages related to sending "an empty 
buffer"?


Daniel

On 22/04/14 15:34, Samuel D Ware wrote:
Apr  3 13:14:32 tel-vc-fs03 /usr/local/sbin/kamailio[7589]: :  
[mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed 
pointer (0x7feaca6c9738), called from : mem/shm_mem.c: 
sh_realloc(88), first free : mem/shm_mem.c: sh_realloc(88) - 
aborting
Apr  3 14:38:58 tel-vc-fs03 /usr/local/sbin/kamailio[3974]: :  
[mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed 
pointer (0x7f34d7d6b720), called from : mem/shm_mem.c: 
sh_realloc(88), first free : mem/shm_mem.c: sh_realloc(88) - 
aborting
Apr 10 14:16:34 tel-vc-fs03 /usr/local/sbin/kamailio[31354]: :  
[mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed 
pointer (0x7f8c00319b40), called from : mem/shm_mem.c: 
sh_realloc(88), first free : mem/shm_mem.c: sh_realloc(88) - 
aborting
Apr 10 14:28:00 tel-vc-fs03 /usr/local/sbin/kamailio[31350]: :  
[mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed 
pointer (0x7f8c3bcddf00), called from : mem/shm_mem.c: 
sh_realloc(88), first free : mem/shm_mem.c: sh_realloc(88) - 
aborting
Apr 10 16:00:08 tel-vc-fs03 /usr/local/sbin/kamailio[31357]: :  
[mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed 
pointer (0x7f8c1e5ba128), called from : mem/shm_mem.c: 
sh_realloc(88), first free : mem/shm_mem.c: sh_realloc(88) - 
aborting
Apr 10 17:26:53 tel-vc-fs03 /usr/local/sbin/kamailio[31368]: :  
[mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed 
pointer (0x7f8c09557a80), called from tm: h_table.c: free_cell(159), 
first free tm: t_reply.c: reply_received(2260) - aborting
Apr 10 17:37:02 tel-vc-fs03 /usr/local/sbin/kamailio[31368]: :  
[mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed 
pointer (0x7f8c19c22f10), called from tm: h_table.c: free_cell(186), 
first free tm: h_table.c: free_cell(186) - aborting
Apr 10 17:50:22 tel-vc-fs03 /usr/local/sbin/kamailio[31356]: :  
[mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed 
pointer (0x7f8c4dc899c8), called from : mem/shm_mem.c: 
sh_realloc(88), first free : mem/shm_mem.c: sh_realloc(88) - 
aborting
Apr 10 18:16:50 tel-vc-fs03 /usr/local/sbin/kamailio[31359]: :  
[mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed 
pointer (0x7f8c034dcc58), called from : mem/shm_mem.c: 
sh_realloc(88), first free : mem/shm_mem.c: sh_realloc(88) - 
aborting
Apr 10 18:21:56 tel-vc-fs03 /usr/local/sbin/kamailio[31351]: :  
[mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed 
pointer (0x7f8c4dc899c8), called from : mem/shm_mem.c: 
sh_realloc(88), first free : mem/shm_mem.c: sh_realloc(88) - 
aborting
Apr 10 18:21:56 tel-vc-fs03 /usr/local/sbin/kamailio[31359]: :  
[mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed 
pointer (0x7f8c3daa95a8), called from : mem/shm_mem.c: 
sh_realloc(88), first free : mem/shm_mem.c: sh_realloc(88) - 
aborting
Apr 10 19:18:38 tel-vc-fs03 /usr/local/sbin/kamailio[31359]: :  
[mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed 
pointer (0x7f8c4182b490), called from : mem/shm_mem.c: 
sh_realloc(88), first free : mem/shm_mem.c: sh_realloc(88) - 
aborting
Apr 10 19:21:50 tel-vc-fs03 /usr/local/sbin/kamailio[31364]: :  
[mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed 
pointer (0x7f8c020bbea8), called from : mem/shm_mem.c: 
sh_realloc(88), first free : mem/shm_mem.c: sh_realloc(88) - 
aborting
Apr 10 19:28:23 tel-vc-fs03 /usr/local/sbin/kamailio[31350]: :  
[mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed 
pointer (0x7f8c19be2f78), called from : mem/shm_mem.c: 
sh_realloc(88), first free : mem/shm_mem.c: sh_realloc(88) - 
aborting
Apr 10 19:38:07 tel-vc-fs03 /usr/local/sbin/kamailio[31363]: :  
[mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed 
pointer (0x7f8c467eb400), called from : mem/shm_mem.c: 
sh_realloc(88), first free : mem/shm_mem.c: sh_realloc(88) - 
aborting
Apr 10 19:40:56 tel-vc-fs03 /usr/local/sbin/kamailio[31354]: :  
[mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed 
pointer (0x7f8c68795760), called from : mem/shm_mem.c: 
sh_realloc(88), first free : mem/shm_mem.c: sh_realloc(88) - 
aborting
Apr 10 20:45:07 tel-vc-fs03 /usr/local/sbin/kamailio[31362]: :  
[mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed 
pointer (0x7f8c50863060), called from : mem/shm_mem.c: 
sh_realloc(88), first free : mem/shm_mem.c: sh_realloc(88) - 
aborting
Apr 10 20:48:49 tel-vc-fs03 /usr/local/sbin/kamailio[31363]: :  
[mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed 
pointer (0x7f8c751ef678), called from : mem/shm_mem.c: 
sh_realloc(88), first free : mem/shm_mem.c: sh_realloc(88) - 
aborting
Apr 10 21:18:21 tel-vc-fs03 /usr/local/sbin/kamailio[31362]: :  
[mem/q_malloc.c:468]: qm_

Re: [SR-Users] LDAP and Kerberos backend for authentification / Or PAM / SASL ?

[Daniel-Constantin Mierla]

On 23/04/14 08:06, Yoann Gini wrote:


Le 22 avr. 2014 à 11:03, Daniel-Constantin Mierla > a écrit :


the constraint to have access to text password or HA1 format (md5 
over username, password, realm) comes from WWW Authentication 
mechanism which is used by SIP.


Writing something different in kamailio would be possible (it is open 
source), but you don't have phones able to do it and provide the 
adequate details.


Are you sure?

Sure you can't be sure of anything but this one.

You’ve a Radius backend for example, Radius don’t allow you to access 
to clear text password, it isn’t?
Look at digest module for (free)radius, if there is something like that 
for ldap, then you may get it working with some patch. But it still 
requires access to plain text password or HA1.



The only www authentication mechanism who must have access to clear 
text password is the DIGEST auth. But if we consider using SIP over 
TLS, we may be able to use BASIC authentication…


What do you think ? Does the authentication kind is negotiated during 
the communication?
If you use tls and give a signed certificate to client, then you can 
simply authenticate it by trusting the certificate and checking the 
owner fields to match sip headers.


If you control the client and develop it, you can add any authentication 
mechanism. You will eventually have to add to kamailio appropriate 
authentication support, but that's easy, it's open source.


However, SIP RFC enforces www digest authentication and it is what all 
the phones I am aware of in the wild support now.


--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

[SR-Users] Can anyone Help with RTP Proxy Recording

[sam]

Hi,

I am running a SIP Service using Kamailio.

I have activated the RTP Proxy Recording.

I need two things to be done.

1. RTP files saved need to have caller and callee usernames as file names.
2. Need to convert RTP files to mp3.

Can anyone assist on this?

Thanks
SamG
DS Media
___
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] Pick domain name of a user from location list of siremis

[aawaise]

Hello all,

I want to pick the domain name of a particular online user from location
list of siremis and then perform some action on this basis. How can it be
picked ??

Secondly how can the user name of called user be picked from the INVITE
packet received by the server ??

Any help will be appreciated.

Thanks,
Regards,
Aawaise.



--
View this message in context: 
http://sip-router.1086192.n5.nabble.com/Pick-domain-name-of-a-user-from-location-list-of-siremis-tp127058.html
Sent from the Users mailing list archive at Nabble.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] Loadbalancer/dispatcher with dialog/cnxcc

[Oliver Roth]

Hi all

Following situation

1 dispatcher/loadbalancer getting all the inbound traffic and sending it to 3 
different gateway (round robin).
The loadbalancer has no (or even very few) business logic.
Just "in" - split to different gateway - "out"

3 sip gateway doing all the business logic like auth, modifications on header, 
different checks, ...
These 3 gateways use the same database (cluster) with routing tables and so on.
A call gets terminated to a carrier through these sip gateway.

Now I would like to implement call limiting (no of calls) by using either only 
dialog module or cnxcc module based on "source ip" or later on "cli".

My problem:
A call from one source ip can be sent to the (3) different sip gateway - so not 
all calls are processed by the same sip gateway.
How can I ensure, that only a certain number of calls are allowed - even if 
they are split up on the 3 different gateway?

Or do I need to implement this kind of business logic on the 
dispatcher/loadbalancer?
That would not make much sense, because this is just a "stupid" machine...

Thanks for helping
Oli
___
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 from_uri help

[Juha Heinanen]

Geoffrey Mina writes:

> Right now, I would think that with our Priority of 0 on the route with the
> from_URI match, the algorithm should always look at that one first and
> route the calls accordingly.
> 
> Since we have the priority correctly set to evaluate in the order we want,
> what next can we look to?

i made same kind of test that you have and routing worked as expected,
i.e., when request uri matched a common prefix, the one with lower
priority was tried first.

if you remove from uri constraint from the rule that had it, is lower
priority gw then selected first or doesn't priorities work at all for
you?

-- juha



___
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] DRouting, routeid is not triggered

[Maciej Bylica]

Hi Daniel,

Here is debug you requested.

DEBUG:  [parser/msg_parser.c:623]: parse_msg(): SIP Request:
DEBUG:  [parser/msg_parser.c:625]: parse_msg():  method:  
DEBUG:  [parser/msg_parser.c:627]: parse_msg():  uri: <
sip:43111223344@10.10.10.5>
DEBUG:  [parser/msg_parser.c:629]: parse_msg():  version: 
DEBUG:  [parser/parse_via.c:1284]: parse_via_param(): Found param
type 235,  = ; state=6
DEBUG:  [parser/parse_via.c:1284]: parse_via_param(): Found param
type 232,  = ; state=16
DEBUG:  [parser/parse_via.c:2672]: parse_via(): end of header
reached, state=5
DEBUG:  [parser/msg_parser.c:513]: parse_headers(): parse_headers:
Via found, flags=2
DEBUG:  [parser/msg_parser.c:515]: parse_headers(): parse_headers:
this is the first via
DEBUG:  [receive.c:152]: receive_msg(): After parse_msg...
DEBUG:  [receive.c:193]: receive_msg(): preparing to run routing
scripts...
DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 69
DEBUG: maxfwd [maxfwd.c:161]: process_maxfwd_header(): value 69 decreased
to 16
DEBUG:  [parser/parse_addr_spec.c:893]: parse_addr_spec(): end of
header reached, state=10
DEBUG:  [parser/msg_parser.c:190]: get_hdr_field(): DEBUG:
get_hdr_field:  [32]; uri=[sip:43111223344@10.10.10.5]
DEBUG:  [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [<
sip:43111223344@10.10.10.5>#015#012]
DEBUG:  [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field:
cseq : <58787375> 
DEBUG:  [parser/msg_parser.c:204]: get_hdr_field(): DEBUG:
get_hdr_body : content_length=203
DEBUG:  [parser/msg_parser.c:106]: get_hdr_field(): found end of
header
DEBUG:  [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG:
add_param: tag=1eQFK719e4cyS
DEBUG:  [parser/parse_addr_spec.c:893]: parse_addr_spec(): end of
header reached, state=29
DEBUG: sanity [mod_sanity.c:255]: w_sanity_check(): sanity checks result: 1
DEBUG: siputils [checks.c:103]: has_totag(): no totag
DEBUG: tm [t_lookup.c:1072]: t_check_msg(): DEBUG: t_check_msg: msg id=1
global id=0 T start=0x
DEBUG: tm [t_lookup.c:527]: t_lookup_request(): t_lookup_request: start
searching: hash=49678, isACK=0
DEBUG: tm [t_lookup.c:485]: matching_3261(): DEBUG: RFC3261 transaction
matching failed
DEBUG: tm [t_lookup.c:709]: t_lookup_request(): DEBUG: t_lookup_request: no
transaction found
DEBUG: tm [t_lookup.c:1141]: t_check_msg(): DEBUG: t_check_msg: msg id=1
global id=1 T end=(nil)
DEBUG:  [socket_info.c:583]: grep_sock_info(): grep_sock_info -
checking if host==us: 12==9 && [10.10.10.5] == [127.0.0.1]
DEBUG:  [socket_info.c:587]: grep_sock_info(): grep_sock_info -
checking if port 5060 (advertise 0) matches port 5060
DEBUG:  [socket_info.c:583]: grep_sock_info(): grep_sock_info -
checking if host==us: 12==12 && [10.10.10.5] == [10.10.10.5]
DEBUG:  [socket_info.c:587]: grep_sock_info(): grep_sock_info -
checking if port 5060 (advertise 0) matches port 5060
DEBUG: registrar [lookup.c:158]: lookup(): '43111223344' Not found in usrloc
DEBUG: tm [t_lookup.c:1373]: t_newtran(): DEBUG: t_newtran: msg id=1 ,
global msg id=1 , T on entrance=(nil)
DEBUG: tm [t_lookup.c:527]: t_lookup_request(): t_lookup_request: start
searching: hash=49678, isACK=0
DEBUG: tm [t_lookup.c:485]: matching_3261(): DEBUG: RFC3261 transaction
matching failed
DEBUG: tm [t_lookup.c:709]: t_lookup_request(): DEBUG: t_lookup_request: no
transaction found
DEBUG: tm [t_hooks.c:374]: run_reqin_callbacks_internal(): DBG:
trans=0x7f9ae79e2c10, callback type 1, id 0 entered
DEBUG: tm [t_hooks.c:374]: run_reqin_callbacks_internal(): DBG:
trans=0x7f9ae79e2c10, callback type 1, id 0 entered
DEBUG:  [md5utils.c:67]: MD5StringArray(): DEBUG: MD5 calculated:
60b78f5b572d3477887c1e8305c94b0a
DEBUG: drouting [drouting.c:720]: do_routing(): using dr group 10
DEBUG: drouting [prefix_tree.c:87]: internal_check_rt(): found rgid 10
(rule list 0x7f9ae79e2a58)
DEBUG: drouting [drouting.c:895]: do_routing(): setting attr [] as for ruri
DEBUG: drouting [drouting.c:912]: do_routing(): setting the gw [0] as ruri "
sip:43111223344@10.10.10.9"
DEBUG: tm [t_lookup.c:1373]: t_newtran(): DEBUG: t_newtran: msg id=1 ,
global msg id=1 , T on entrance=0x7f9ae79e2c10
DEBUG: tm [t_lookup.c:1378]: t_newtran(): DEBUG: t_newtran: transaction
already in process 0x7f9ae79e2c10
DEBUG: tm [t_funcs.c:347]: t_relay_to(): SER: new INVITE
DEBUG:  [msg_translator.c:204]: check_via_address():
check_via_address(10.10.5.5, 10.10.5.5, 0)
DEBUG:  [mem/shm_mem.c:111]: _shm_resize(): WARNING:vqm_resize:
resize(0) called
DEBUG: tm [t_reply.c:728]: _reply_light(): DEBUG: reply sent out.
buf=0x7f9afe08a608: SIP/2.0 100 trying -..., shmem=0x7f9ae79e5860: SIP/2.0
100 trying -
DEBUG: tm [t_reply.c:738]: _reply_light(): DEBUG: _reply_light: finished
DEBUG: 

Re: [SR-Users] Drouting

[Keith]

Hi,

Or is it possible to use a regular expression for the user?

I basically want certain users to use a certain route without having to add
them all in.

Keith
___
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] test if record_route() has been called already?

[Juha Heinanen]

is there any means to test if record_route() has been called already on
the request?

if it is called more than once, error message is issued.

-- juha

___
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] RTPProxy and premature RTP

[Daniel Tryba]

I'm having some troubles with a provider sending RTP before a 183 Session 
Progress or 200 OK (I see up to 1s of rtp prematurely). The machine is running 
rtpproxy and apparently rtpproxy buffers these rtp packets and flushed them in 
one burst when the 183/200 arrive, this creates havock in some endpoints but 
is undesirable in all cases IMHO if the delay of the 182/200 is to high 
(>0.1s).

Is there a way to control buffering/flushing from kamailio? Is rebuilding 
rtpproxy with a smaller buffer an option? Or should I switch to an other proxy 
module? Any thoughts about this subject (apart from getting the provider to 
stop sending premature rtp)?

-- 

POCOS B.V. - Croy 9c - 5653 LC Eindhoven
Telefoon: 040 293 8661 - Fax: 040 293 8658
http://www.pocos.nl/   - http://www.sipo.nl/
K.v.K. Eindhoven 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] RTPProxy and premature RTP

[Alex Balashov]

On 04/23/2014 07:36 AM, Daniel Tryba wrote:


I'm having some troubles with a provider sending RTP before a 183
Session Progress or 200 OK (I see up to 1s of rtp prematurely). The
machine is running rtpproxy and apparently rtpproxy buffers these rtp
packets and flushed them in one burst when the 183/200 arrive, this
creates havock in some endpoints but is undesirable in all cases IMHO
if the delay of the 182/200 is to high (>0.1s).


Just of curiosity, how exactly does the "havoc" manifest itself? What 
are the symptoms of this being a problem?


--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.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] Kamailio-Asterisk - route "FROMASTERISK" not found

[Patrik Kristel]

Hi,

thanks for help! I solved it, I have not define route(FROMASTERISK) in my
kamailio.cfg


Thank you again!

Patrik


On Tue, Apr 22, 2014 at 11:05 AM, Daniel-Constantin Mierla <
mico...@gmail.com> wrote:

>  Hello,
>
> if you search for FROMASTERISK, do you find it on kamailio.cfg?
>
> Check also that you have the line:
>
> #!define WITH_ASTERISK
>
> somewhere at the top of kamailio.cfg.
>
> Cheers,
> Daniel
>
>
> On 20/04/14 08:38, Patrik Kristel wrote:
>
> Hello all,
>
>  I'm trying to implement Kamailio 4.1 with Asterisk 12.1.0 regarging this
> tutorial:
>
>>
>> http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
>
>
>  And when I try to compile kamaili.cfg, I still got this error:
> ERROR route "FROMASTERISK" not found
>
>  I have loaded all modules like in tutorial.
>
>  I tried to find some solve this issue, but with no result.
>
>
> ___
> 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 Mierla - http://www.asipto.comhttp://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
>
>
___
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] Realtime integration: Unregistered clients showing as registered?

[Olli Heiskanen]

Hello,

Gracias Pedro, kiitos Mikko.

It's good to know I have configured Kamailio correctly. I added the type
into my table but so far no luck having asterisk see the clients
registered, at least on cli. I do see that asterisk adds registration data
into the table. I'll work on this for a bit and ask in the asterisk list on
more tricks on asterisk side. I'll post back here if I find out what the
problem was, in case someone is having similar issues.

Thanks again,
Olli



2014-04-22 21:06 GMT+03:00 Pedro Niño :

> Don't forget to include peer type (friend), and The callbacknumber In The
> table.
>
> It happened to me and asterisk/kamailio behavior was wayyy to weird  until
> made sure both parameters were there.
>
> -
>
> In this setup I have SIP peers in an asterisk table added like this:
>
> INSERT INTO sippeers (name, defaultuser, host, sippasswd, fromuser,
> fromdomain) VALUES ('660', '660', 'dynamic', 'password', '660', '
> testers.com');
>
> --
>  El abr 19, 2014 1:17 PM, "Olli Heiskanen" 
> escribió:
>
>>
>> Hello,
>>
>> One of the tests I've been working with is Asterisk realtime integration
>> according to Daniel's guide here:
>> http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
>>
>> Weird thing is the client looks registered but I'm not sure if it really
>> is registered. If I'm not mistaken I should see the peers when I issue 'sip
>> show peers' on asterisk cli. Instead I get this:
>>
>> *CLI> sip show peers
>> Name/username  Host  Dyn Forcerport Comedia  ACL Port
>>  Status  Description  Realtime
>> 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0
>> offline]
>>
>>
>> Also, calling between clients will fail; in Asterisk cli I get:
>> *CLI>
>>   == Using SIP RTP TOS bits 184
>>   == Using SIP RTP CoS mark 5
>> -- Executing [661@default:1] NoOp("SIP/660-", "Testing:
>> Dialed 661") in new stack
>> -- Executing [661@default:2] Dial("SIP/660-",
>> "SIP/661,3600,rt") in new stack
>>   == Using SIP RTP TOS bits 184
>>   == Using SIP RTP CoS mark 5
>> -- Called SIP/661
>>   == Everyone is busy/congested at this time (1:0/0/1)
>> -- Executing [661@default:3] Hangup("SIP/660-", "") in new
>> stack
>>   == Spawn extension (default, 661, 3) exited non-zero on
>> 'SIP/660-'
>>
>>
>> In this setup I have SIP peers in an asterisk table added like this:
>> INSERT INTO sippeers (name, defaultuser, host, sippasswd, fromuser,
>> fromdomain) VALUES ('660', '660', 'dynamic', 'password', '660', '
>> testers.com');
>>
>> I have Kamailio and Asterisk on the same machine where Kamailio listens
>> port 5060 and Asterisk listens 5070. Things that differ from the guide are
>> Kamailio and Asterisk versions, which in my case are newer. Also, for
>> another testing case I have MULTIDOMAIN enabled in Kamailio, does this
>> interfere with the realtime integration? I'm using only one domain though.
>>
>> Please let me know if any configs or traces I can provide will help
>> figure out what's going on.
>>
>> cheers,
>> Olli
>>
>> ___
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users@lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

[SR-Users] TCP reset

[Daniel Ciprus]

Hi,

Is there any way to remove registration entry from registrar (Kamailio IMS 
built on stable 4.1) detected on p-cscf by incoming TCP RST from the network ? 
What's happening is that client is loosing connections/crashing and TCP RST is 
sent back to pcscf which is not propagating changes back to registrar. This is 
likely not possible with my version of kamailio but at least logging which 
would indicate change of the status will be helpful.

thanks for any hints


--
Daniel Ciprus
Integration engineer
http://www.acision.com

9954 Mayland Dr
Suite 3100
Richmond, VA 23233
USA
T: +1 804 762 5601
E: daniel.cip...@acision.com


This e-mail and any attachment is for authorised use by the intended 
recipient(s) only. It may contain proprietary material, confidential 
information and/or be subject to legal privilege. It should not be copied, 
disclosed to, retained or used by, any other party. If you are not an intended 
recipient then please promptly delete this e-mail and any attachment and all 
copies and inform the sender. Thank you for understanding.

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] TCP reset

[Carsten Bock]

Hi Daniel,

what you want, is basically already implemented in the "Standard"
usrloc module, see:

http://kamailio.org/docs/modules/devel/modules/usrloc.html#usrloc.p.handle_lost_tcp

We haven't implemented that feature yet in the IMS modules

Schöne Grüße,
Carsten


2014-04-23 15:47 GMT+02:00 Daniel Ciprus :
> Hi,
>
> Is there any way to remove registration entry from registrar (Kamailio IMS
> built on stable 4.1) detected on p-cscf by incoming TCP RST from the network
> ? What's happening is that client is loosing connections/crashing and TCP
> RST is sent back to pcscf which is not propagating changes back to
> registrar. This is likely not possible with my version of kamailio but at
> least logging which would indicate change of the status will be helpful.
>
> thanks for any hints
>
>
> --
> Daniel Ciprus
> Integration engineer
> http://www.acision.com
>
> 9954 Mayland Dr
> Suite 3100
> Richmond, VA 23233
> USA
> T: +1 804 762 5601
> E: daniel.cip...@acision.com
>
> 
> This e-mail and any attachment is for authorised use by the intended
> recipient(s) only. It may contain proprietary material, confidential
> information and/or be subject to legal privilege. It should not be copied,
> disclosed to, retained or used by, any other party. If you are not an
> intended recipient then please promptly delete this e-mail and any
> attachment and all copies and inform the sender. Thank you for
> understanding.
>
>
> ___
> 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
>



-- 
Carsten Bock
CEO (Geschäftsführer)

ng-voice GmbH
Schomburgstr. 80
D-22767 Hamburg / Germany

http://www.ng-voice.com
mailto:cars...@ng-voice.com

Office +49 40 34927219
Fax +49 40 34927220

Sitz der Gesellschaft: Hamburg
Registergericht: Amtsgericht Hamburg, HRB 120189
Geschäftsführer: Carsten Bock
Ust-ID: DE279344284

Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
http://www.ng-voice.com/imprint/

___
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] RTPProxy and premature RTP

[Daniel Tryba]

On Wednesday 23 April 2014 13:41:42 Alex Balashov wrote:
> Just of curiosity, how exactly does the "havoc" manifest itself? What 
> are the symptoms of this being a problem?

Havoc might be a bit strong (though "great confusion and disorder" fits well). 
Bad sound quality (if any sound) for a longer period due to multiple jitter 
buffer overflows for that one channel and if the endpoint is on a limited 
circuit drops in the other channels as well. It manifests on a particular PBX 
(Swyx), most endpoints are Patton Smartnodes and they seem to cope well.

-- 

POCOS B.V. - Croy 9c - 5653 LC Eindhoven
Telefoon: 040 293 8661 - Fax: 040 293 8658
http://www.pocos.nl/   - http://www.sipo.nl/
K.v.K. Eindhoven 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] TCP reset

[Daniel Ciprus]

Thanks Carsten,

Any plans to do so in near future ? :-)

On 04/23/2014 09:52 AM, Carsten Bock wrote:

Hi Daniel,

what you want, is basically already implemented in the "Standard"
usrloc module, see:

http://kamailio.org/docs/modules/devel/modules/usrloc.html#usrloc.p.handle_lost_tcp

We haven't implemented that feature yet in the IMS modules

Schöne Grüße,
Carsten


2014-04-23 15:47 GMT+02:00 Daniel Ciprus 
:


Hi,

Is there any way to remove registration entry from registrar (Kamailio IMS
built on stable 4.1) detected on p-cscf by incoming TCP RST from the network
? What's happening is that client is loosing connections/crashing and TCP
RST is sent back to pcscf which is not propagating changes back to
registrar. This is likely not possible with my version of kamailio but at
least logging which would indicate change of the status will be helpful.

thanks for any hints


--
Daniel Ciprus
Integration engineer
http://www.acision.com

9954 Mayland Dr
Suite 3100
Richmond, VA 23233
USA
T: +1 804 762 5601
E: daniel.cip...@acision.com


This e-mail and any attachment is for authorised use by the intended
recipient(s) only. It may contain proprietary material, confidential
information and/or be subject to legal privilege. It should not be copied,
disclosed to, retained or used by, any other party. If you are not an
intended recipient then please promptly delete this e-mail and any
attachment and all copies and inform the sender. Thank you for
understanding.


___
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 Ciprus
Integration engineer
http://www.acision.com

9954 Mayland Dr
Suite 3100
Richmond, VA 23233
USA
T: +1 804 762 5601
E: daniel.cip...@acision.com


This e-mail and any attachment is for authorised use by the intended 
recipient(s) only. It may contain proprietary material, confidential 
information and/or be subject to legal privilege. It should not be copied, 
disclosed to, retained or used by, any other party. If you are not an intended 
recipient then please promptly delete this e-mail and any attachment and all 
copies and inform the sender. Thank you for understanding.

___
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] Realtime integration: Unregistered clients showing as registered?

[Pedro Niño]

Try to install sngrep In both hosts (Kamailio and asterisk). It will help
you figure what is happening more clear.

I'll double check my own config, and Let you know the needed fields, At
least for my case.

I used the same integration  guide,  and splitted the model in 3 servers.
One for kamailio, one for databases and one for media server (asterisk).

It's now hosting 350 users with an avg of 15 concurrent calls,  planning to
take it to 1200 users in a course of 6 months.
 El abr 23, 2014 8:29 AM, "Olli Heiskanen" 
escribió:

> Hello,
>
> Gracias Pedro, kiitos Mikko.
>
> It's good to know I have configured Kamailio correctly. I added the type
> into my table but so far no luck having asterisk see the clients
> registered, at least on cli. I do see that asterisk adds registration data
> into the table. I'll work on this for a bit and ask in the asterisk list on
> more tricks on asterisk side. I'll post back here if I find out what the
> problem was, in case someone is having similar issues.
>
> Thanks again,
> Olli
>
>
>
> 2014-04-22 21:06 GMT+03:00 Pedro Niño :
>
>> Don't forget to include peer type (friend), and The callbacknumber In The
>> table.
>>
>> It happened to me and asterisk/kamailio behavior was wayyy to weird
>> until made sure both parameters were there.
>>
>> -
>>
>> In this setup I have SIP peers in an asterisk table added like this:
>>
>> INSERT INTO sippeers (name, defaultuser, host, sippasswd, fromuser,
>> fromdomain) VALUES ('660', '660', 'dynamic', 'password', '660', '
>> testers.com');
>>
>> --
>>  El abr 19, 2014 1:17 PM, "Olli Heiskanen" <
>> ohjelmistoarkkite...@gmail.com> escribió:
>>
>>>
>>> Hello,
>>>
>>> One of the tests I've been working with is Asterisk realtime integration
>>> according to Daniel's guide here:
>>> http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
>>>
>>> Weird thing is the client looks registered but I'm not sure if it really
>>> is registered. If I'm not mistaken I should see the peers when I issue 'sip
>>> show peers' on asterisk cli. Instead I get this:
>>>
>>> *CLI> sip show peers
>>> Name/username  Host  Dyn Forcerport Comedia  ACL Port
>>>  Status  Description  Realtime
>>> 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0
>>> offline]
>>>
>>>
>>> Also, calling between clients will fail; in Asterisk cli I get:
>>> *CLI>
>>>   == Using SIP RTP TOS bits 184
>>>   == Using SIP RTP CoS mark 5
>>> -- Executing [661@default:1] NoOp("SIP/660-", "Testing:
>>> Dialed 661") in new stack
>>> -- Executing [661@default:2] Dial("SIP/660-",
>>> "SIP/661,3600,rt") in new stack
>>>   == Using SIP RTP TOS bits 184
>>>   == Using SIP RTP CoS mark 5
>>> -- Called SIP/661
>>>   == Everyone is busy/congested at this time (1:0/0/1)
>>> -- Executing [661@default:3] Hangup("SIP/660-", "") in new
>>> stack
>>>   == Spawn extension (default, 661, 3) exited non-zero on
>>> 'SIP/660-'
>>>
>>>
>>> In this setup I have SIP peers in an asterisk table added like this:
>>> INSERT INTO sippeers (name, defaultuser, host, sippasswd, fromuser,
>>> fromdomain) VALUES ('660', '660', 'dynamic', 'password', '660', '
>>> testers.com');
>>>
>>> I have Kamailio and Asterisk on the same machine where Kamailio listens
>>> port 5060 and Asterisk listens 5070. Things that differ from the guide are
>>> Kamailio and Asterisk versions, which in my case are newer. Also, for
>>> another testing case I have MULTIDOMAIN enabled in Kamailio, does this
>>> interfere with the realtime integration? I'm using only one domain though.
>>>
>>> Please let me know if any configs or traces I can provide will help
>>> figure out what's going on.
>>>
>>> cheers,
>>> Olli
>>>
>>> ___
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users@lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>> ___
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users@lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Realtime integration: Unregistered clients showing as registered?

[Pedro Niño]

Ahhh also, don't forget to place the *rtcachefriends*=*yes* in your
sip.conf (asterisk) to show the realtime peers
El abr 23, 2014 8:29 AM, "Olli Heiskanen" 
escribió:

> Hello,
>
> Gracias Pedro, kiitos Mikko.
>
> It's good to know I have configured Kamailio correctly. I added the type
> into my table but so far no luck having asterisk see the clients
> registered, at least on cli. I do see that asterisk adds registration data
> into the table. I'll work on this for a bit and ask in the asterisk list on
> more tricks on asterisk side. I'll post back here if I find out what the
> problem was, in case someone is having similar issues.
>
> Thanks again,
> Olli
>
>
>
> 2014-04-22 21:06 GMT+03:00 Pedro Niño :
>
>> Don't forget to include peer type (friend), and The callbacknumber In The
>> table.
>>
>> It happened to me and asterisk/kamailio behavior was wayyy to weird
>> until made sure both parameters were there.
>>
>> -
>>
>> In this setup I have SIP peers in an asterisk table added like this:
>>
>> INSERT INTO sippeers (name, defaultuser, host, sippasswd, fromuser,
>> fromdomain) VALUES ('660', '660', 'dynamic', 'password', '660', '
>> testers.com');
>>
>> --
>>  El abr 19, 2014 1:17 PM, "Olli Heiskanen" <
>> ohjelmistoarkkite...@gmail.com> escribió:
>>
>>>
>>> Hello,
>>>
>>> One of the tests I've been working with is Asterisk realtime integration
>>> according to Daniel's guide here:
>>> http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
>>>
>>> Weird thing is the client looks registered but I'm not sure if it really
>>> is registered. If I'm not mistaken I should see the peers when I issue 'sip
>>> show peers' on asterisk cli. Instead I get this:
>>>
>>> *CLI> sip show peers
>>> Name/username  Host  Dyn Forcerport Comedia  ACL Port
>>>  Status  Description  Realtime
>>> 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0
>>> offline]
>>>
>>>
>>> Also, calling between clients will fail; in Asterisk cli I get:
>>> *CLI>
>>>   == Using SIP RTP TOS bits 184
>>>   == Using SIP RTP CoS mark 5
>>> -- Executing [661@default:1] NoOp("SIP/660-", "Testing:
>>> Dialed 661") in new stack
>>> -- Executing [661@default:2] Dial("SIP/660-",
>>> "SIP/661,3600,rt") in new stack
>>>   == Using SIP RTP TOS bits 184
>>>   == Using SIP RTP CoS mark 5
>>> -- Called SIP/661
>>>   == Everyone is busy/congested at this time (1:0/0/1)
>>> -- Executing [661@default:3] Hangup("SIP/660-", "") in new
>>> stack
>>>   == Spawn extension (default, 661, 3) exited non-zero on
>>> 'SIP/660-'
>>>
>>>
>>> In this setup I have SIP peers in an asterisk table added like this:
>>> INSERT INTO sippeers (name, defaultuser, host, sippasswd, fromuser,
>>> fromdomain) VALUES ('660', '660', 'dynamic', 'password', '660', '
>>> testers.com');
>>>
>>> I have Kamailio and Asterisk on the same machine where Kamailio listens
>>> port 5060 and Asterisk listens 5070. Things that differ from the guide are
>>> Kamailio and Asterisk versions, which in my case are newer. Also, for
>>> another testing case I have MULTIDOMAIN enabled in Kamailio, does this
>>> interfere with the realtime integration? I'm using only one domain though.
>>>
>>> Please let me know if any configs or traces I can provide will help
>>> figure out what's going on.
>>>
>>> cheers,
>>> Olli
>>>
>>> ___
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users@lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>> ___
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users@lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

[SR-Users] rtpproxy_manage() issues in rtpproxy-ng

[Alex Balashov]

Hello,

I'm running the latest pull of Kamailio 4.1 (last commit 
be187e135b0b9b28136817c3569ab5c0abcc5b3f) and am using rtpproxy-ng with 
a recent mediaproxy-ng master (commit 
cb6990e43864b077dd6a24acfbdf5ef76c1a427e).


For no apparent reason, Kamailio has stopped sending 'offer' commands to 
it when I use rtpproxy_manage(). My use of it is in this setting:


  if(isflagset(PROXY_MEDIA) && !isflagset(PROXY_MEDIA_SET) &&
   has_body("application/sdp")) {
set_rtp_proxy_set("1");
rtpproxy_manage("o");

add_rr_param(";proxy_media=yes");

setflag(PROXY_MEDIA_SET);
  }

I put in a log message to confirm that this block is being run. It is.

But, there's no offer command going to the mediaproxy-ng on the other 
box, as confirmed by packet captures and logs. However, the 
mediaproxy-ng is getting lots of 'answer' and 'delete' commands that it 
doesn't know what to do with, since the call was never initialised with 
'offer'.


When I change rtpproxy_manage() to rtpproxy_offer(), it works perfectly.

I've been using rtpproxy_manage() forever, almost as long as it's 
existed. Does it not work with rtpproxy-ng, despite what the 
documentation says?


--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.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] rtpproxy_manage() issues in rtpproxy-ng

[Alex Balashov]

If it helps, here are my rtpproxy-ng modparams:

modparam("rtpproxy-ng", "rtpproxy_sock", "1 == udp:xxx.xxx.xxx.xxx:5050")
modparam("rtpproxy-ng", "rtpproxy_disable_tout", 120)
modparam("rtpproxy-ng", "rtpproxy_tout", 1)
modparam("rtpproxy-ng", "rtpproxy_retr", 2)

--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.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] rtpproxy_manage() issues in rtpproxy-ng

[Richard Fuchs]

On 04/23/14 11:53, Alex Balashov wrote:
> Hello,
> 
> I'm running the latest pull of Kamailio 4.1 (last commit
> be187e135b0b9b28136817c3569ab5c0abcc5b3f) and am using rtpproxy-ng with
> a recent mediaproxy-ng master (commit
> cb6990e43864b077dd6a24acfbdf5ef76c1a427e).
> 
> For no apparent reason, Kamailio has stopped sending 'offer' commands to
> it when I use rtpproxy_manage(). My use of it is in this setting:
> 
>   if(isflagset(PROXY_MEDIA) && !isflagset(PROXY_MEDIA_SET) &&
>has_body("application/sdp")) {
> set_rtp_proxy_set("1");
> rtpproxy_manage("o");
> 
> add_rr_param(";proxy_media=yes");
> 
> setflag(PROXY_MEDIA_SET);
>   }
> 
> I put in a log message to confirm that this block is being run. It is.
> 
> But, there's no offer command going to the mediaproxy-ng on the other
> box, as confirmed by packet captures and logs. However, the
> mediaproxy-ng is getting lots of 'answer' and 'delete' commands that it
> doesn't know what to do with, since the call was never initialised with
> 'offer'.

Can you please clarify: does it not send anything at all, or does it
send an answer instead of an offer?

cheers




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] rtpproxy_manage() issues in rtpproxy-ng

[Alex Balashov]

On 04/23/2014 12:08 PM, Richard Fuchs wrote:


Can you please clarify: does it not send anything at all, or does it
send an answer instead of an offer?


Well, I suppose I can't say for sure, but it looks like it's not sending 
anything at all. When I grep a particular Call-ID, the first thing I get is:


Apr 23 15:28:37 sd-rtp01 mediaproxy-ng[7910]: Got valid command from
xxx.xxx.xxx.xxx:43733: answer - { "sdp": "v=0#015#012o=- 1398266077
1398266077 IN IP4 xxx.xxx.xxx.xxx#015#012s=VOS2009#015#012c=IN IP4
yyy.yyy.yyy.yyy#015#012t=0 0#015#012m=audio 33576 RTP/AVP 18
101#015#012a=rtpmap:18 G729/8000#015#012a=fmtp:18
annexb=no#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101
0-16#015#012", "replace": [ "origin" ], "call-id":
"15109646-3607255707-160030@xxx", "received-from": [
"IP4", "xxx.xxx.xxx.xxx" ], "from-tag": "3607255707-160035", "to-tag":
"Fa1Fe4ZSvBQSF", "command": "answer" }

If my understanding of the control protocol is correct, this is actually 
an answer, not a mislabeled offer, since the offer would contain some 
other attributes.


But maybe not?

I expect an answer because I have an rtpproxy_manage() in an 
onreply_route for 18x/2xx+SDP responses to this INVITE, of course.


--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.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] rtpproxy_manage() issues in rtpproxy-ng

[Alex Balashov]

Maybe it's not a bug. I think I've got a theory about what may be 
happening. Admittedly, it's not trivial to follow, but bear with me.


The invocation of rtpproxy_manage() is happening in a REQUEST_ROUTE that 
is actually being triggered out of a FAILURE_ROUTE, because we are 
pulling routing info from a redirect server. So, it looks like this 
(obviously, greatly simplified):


   # Main request route.

   route {
  ...

  t_on_failure("FAIL");
  t_relay();
   }

   route[PROCESS_CALL] {
  rtpproxy_manage("o");
   }

   failure_route[FAIL] {
  if($T_rpl($rs) == 302) {
 ...

 route(PROCESS_CALL);
  }
   }

Now, "PROCESS_CALL" is a request route and the things that are done 
inside it are all safe to do inside a FAILURE_ROUTE, e.g. no stateless 
replies. However, because it's being called from a FAILURE_ROUTE, I'll 
bet what's happening is that the evaluative context tells 
rtpproxy_manage() that it's dealing with a _reply_ (the 302 redirect), 
not a _request_, so it should be sending an 'answer' on that basis.


Does that sound reasonable? I don't have an easy way of testing this 
thesis right now since it's production.


--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.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] rtpproxy_manage() issues in rtpproxy-ng

[Alex Balashov]

The reason I had not previously considered this possibility is because 
the documentation says--or, at least to my lackadaisical 
interpretation--that rtpproxy_manage() will only call rtpproxy_answer() 
if it is operating on a 1xx/2xx reply with SDP, otherwise it'll send 
rtpproxy_offer(), or send a delete command if it's a >= 300 reply.


On 04/23/2014 12:51 PM, Alex Balashov wrote:


Maybe it's not a bug. I think I've got a theory about what may be
happening. Admittedly, it's not trivial to follow, but bear with me.

The invocation of rtpproxy_manage() is happening in a REQUEST_ROUTE that
is actually being triggered out of a FAILURE_ROUTE, because we are
pulling routing info from a redirect server. So, it looks like this
(obviously, greatly simplified):

# Main request route.

route {
   ...

   t_on_failure("FAIL");
   t_relay();
}

route[PROCESS_CALL] {
   rtpproxy_manage("o");
}

failure_route[FAIL] {
   if($T_rpl($rs) == 302) {
  ...

  route(PROCESS_CALL);
   }
}

Now, "PROCESS_CALL" is a request route and the things that are done
inside it are all safe to do inside a FAILURE_ROUTE, e.g. no stateless
replies. However, because it's being called from a FAILURE_ROUTE, I'll
bet what's happening is that the evaluative context tells
rtpproxy_manage() that it's dealing with a _reply_ (the 302 redirect),
not a _request_, so it should be sending an 'answer' on that basis.

Does that sound reasonable? I don't have an easy way of testing this
thesis right now since it's production.




--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.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] rtpproxy_manage() issues in rtpproxy-ng

[Richard Fuchs]

On 04/23/14 12:59, Alex Balashov wrote:
> The reason I had not previously considered this possibility is because
> the documentation says--or, at least to my lackadaisical
> interpretation--that rtpproxy_manage() will only call rtpproxy_answer()
> if it is operating on a 1xx/2xx reply with SDP, otherwise it'll send
> rtpproxy_offer(), or send a delete command if it's a >= 300 reply.

Actually the logic is a bit more complicated than that. You can look at
rtpproxy_manage() in rtpproxy.c for the full details. Main selection
criterion is whether the message is a request or a reply, second
criterion is the SIP method (taken from the CSeq) and/or the response
code in case of a reply. The route type is only marginally relevant. So
it really depends on what kind of SIP message you're acting upon.

You should actually see the same behaviour with rtpproxy module as well,
as this part of the code hasn't been changed.

cheerse



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] rtpproxy_manage() issues in rtpproxy-ng

[Alex Balashov]

On 04/23/2014 01:22 PM, Richard Fuchs wrote:


Main selection criterion is whether the message is a request or a
reply, second criterion is the SIP method (taken from the CSeq)
and/or the response code in case of a reply. The route type is only
marginally relevant.


Yeah, so the key question is: what is the message we are acting upon?

It it is my theory that in a request route that is called from a 
failure_route that is triggered by a 302 reply, the message being 
operated on is actually the 302 reply, and not an initial INVITE. And 
that's why it doesn't produce the offer command as expected.


The legacy rtpproxy module may well behave the same way. I hadn't tried 
to use rtpproxy_manage() in this scope before, which is why I was 
imagining it to have "always worked".


--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.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] rtpproxy_manage() issues in rtpproxy-ng

[Richard Fuchs]

On 04/23/14 13:24, Alex Balashov wrote:
> On 04/23/2014 01:22 PM, Richard Fuchs wrote:
> 
>> Main selection criterion is whether the message is a request or a
>> reply, second criterion is the SIP method (taken from the CSeq)
>> and/or the response code in case of a reply. The route type is only
>> marginally relevant.
> 
> Yeah, so the key question is: what is the message we are acting upon?
> 
> It it is my theory that in a request route that is called from a
> failure_route that is triggered by a 302 reply, the message being
> operated on is actually the 302 reply, and not an initial INVITE. And
> that's why it doesn't produce the offer command as expected.

Correct, it would be sending a delete to the proxy. I'm not certain that
instead sending an offer is indeed the expected behaviour.

cheers



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

[SR-Users] IBM SIP server has implemented proxy managed overload protection

[Yang Hong]

Hello Folks.
 
According the new update on Feb 5, 2014, IBM has implemented proxy managed 
overload protection (PMOP) at SIP proxy server.
http://pic.dhe.ibm.com/infocenter/wasinfo/v7r0/index.jsp?topic=%2Fcom.ibm.websphere.nd.doc%2Finfo%2Fae%2Fae%2Fcjpx_sippxovldpro.html
 
The above IBM link lists IBM SIP proxy server custom properties for SIP 
overload protection/control.
•burstResetFactor
•deflatorRatio
•dropOverloadPackets
•inDialogAveragingPeriod
•maxThroughputFactor
•outDialogAveragingPeriod
•perSecondBurstFactor
•proxyTransitionPeriod
•sipProxyStartupDelay
 
IBM SIP overload protection mechanism can be regarded as "Local SIP Overload 
Protection/Control" (Proceedings of WWIC, June 2013).
 
I am moderator of open-source FreeRDP-WebConnect project at GitHub.
https://github.com/FreeRDP/FreeRDP-WebConnect/issues/36
 
I will try to integrate two implicit SIP overload protection/control algorithms 
(RRRC, IEEE Globecom 2010 and RTDC, IEEE ICC 2011) into Kamailio (OpenSER) in 
the future.
 
A survey on SIP overload protection/control algorithms (including IETF RFC "SIP 
Overload Control") can be downloaded from the following ResearchGate link.
Y. Hong, C. Huang, and J. Yan, “A Comparative Study of SIP Overload Control 
Algorithms,"Network and Traffic Engineering in Emerging Distributed Computing 
Applications, Edited by J. Abawajy, M. Pathan, M. Rahman, A.K. Pathan, and M.M. 
Deris, IGI Global, 2012, pp. 1-20.
http://www.researchgate.net/publication/231609451_A_Comparative_Study_of_SIP_Overload_Control_Algorithms
http://arxiv.org/abs/1210.1505
 
This survey paper provides a short review on SIP Express Router (SER). As we 
know, SIP Express Router (SER) and Kamailio (OpenSER) are open-source SIP 
router.
"SIP Express Router (SER) provides a load balancing module to mitigate the 
overload caused by large subscriber populations or abnormal operational 
conditions (IP Telecommunications Portal, 2011).
http://www.igi-global.com/chapter/comparative-study-sip-overload-control/67496
 
The presentation slides for two implicit SIP overload protection/control 
algorithms (RRRC and RTDC) are available for your download.
 
Redundant Retransmission Ratio Control (RRRC) - implicit SIP overload 
protection/control algorithm (IEEE Globecom 2010 Slides) can be downloaded from 
the following ResearchGate link.
https://www.researchgate.net/publication/258555827_Mitigating_SIP_Overload_Using_a_Control-Theoretic_Approach
 
Round-Trip Delay Control (RTDC) - implicit SIP overload protection/control 
algorithm (IEEE ICC 2011 Slides) can be downloaded from the following 
ResearchGate link.
https://www.researchgate.net/publication/257945199_Round-Trip_Delay_Control_(RTDC)_For_Mitigating_SIP_Overload_(IEEE_ICC_2011_Slides)
 
The paper with Redundant Retransmission Ratio Control (RRRC, implicit SIP 
overload protection/control) algorithm can be downloaded from the following 
ResearchGate link.
Y. Hong, C. Huang, and J. Yan, "Mitigating SIP Overload Using a 
Control-Theoretic Approach," Proceedings of IEEE Globecom, Miami, FL, U.S.A, 
December 2010.
https://www.researchgate.net/publication/221284946_Miigating_SIP_Overload_Using_a_Control-Theoretic_Approach
http://ieeexplore.ieee.org/xpls/abs_all.jsp?arnumber=5683124
 
Redundant Retransmission Ratio Control (RRRC, implicit SIP overload 
protection/control) algorithm has been quickly adopted by The Central Weather 
Bureau of Taiwan for their early earthquake warning system.
Ting-Yun Chi, Chun-Hao Chen, Han-Chieh Chao, and Sy-Yen Kuo, "An Efficient 
Earthquake Early Warning Message Delivery Algorithm Using an in Time 
Control-Theoretic Approach", 2011.
http://link.springer.com/chapter/10.1007%2F978-3-642-23641-9_15#
http://www.ipv6.org.tw/docu/elearning8_2011/1010004798p_3-7.pdf
 
Short review and comments on RRRC implicit SIP overload protection/control 
algorithm by former IEEE TAC Associate Editor S. Mascolo: 
Luca De Cicco, Giuseppe Cofano, and Saverio Mascolo, "Local SIP Overload 
Control", Proceedings of WWIC, June 2013.
http://link.springer.com/chapter/10.1007%2F978-3-642-38401-1_16#
http://c3lab.poliba.it/images/2/2a/SipOverload_WWIC13.pdf
 
The paper with Round-Trip Delay Control (RTDC, implicit SIP overload 
protection/control) algorithm can be downloaded from the following ResearchGate 
link.
Y. Hong, C. Huang, and J. Yan, "Design Of A PI Rate Controller For Mitigating 
SIP Overload," Proceedings of IEEE ICC, Kyoto, Japan, June 2011. 
https://www.researchgate.net/publication/224249824_Design_of_a_PI_Rate_Controller_for_Mitigating_SIP_Overload
http://ieeexplore.ieee.org/xpls/abs_all.jsp?arnumber=5963029
 
Round-Trip Delay Control (RTDC, implicit SIP overload protection/control) 
algorithm has been recommended as White Paper by TechRepublic (CBS Interactive)
http://www.techrepublic.com/whitepapers/design-of-a-pi-rate-controller-for-mitigating-sip-overload/25142469
 
Control theoretic approaches have been applied to model the interactions 
between an overloaded SIP server and its u

Re: [SR-Users] LDAP and Kerberos backend for authentification / Or PAM / SASL ?

[Yoann Gini]

Le 23 avr. 2014 à 09:50, Daniel-Constantin Mierla  a écrit :

> However, SIP RFC enforces www digest authentication and it is what all the 
> phones I am aware of in the wild support now.

Thanks for all this informations.

That explain me why all SIP product I see on the market have this really big 
issue of requiring a distinct PIN code for SIP account.

As a sys admin who maintain a unique identity for all enterprise services, it’s 
hard to accept to make an exception for SIP…

I don’t understand how it's possible to end up on a RFC like that…

The good point for me is, on OS X Server, I’ve a private API who can provide me 
DIGEST challenge, so something is possible. But for my FreeBSD based server, 
I’m stuck…



TLS authentication is harder to deploy in SMB. That mean a internal CA and a 
overhead to ensure that each client certificate are well secured.


The solution of BASIC authentication over TLS connection (with certificate only 
on the server) is widely used by HTTPS based software or event e-mail protocols 
to allow add-on services to be connected to existing directory services without 
requiring access to clear text password.


Cheers,
Yoann

smime.p7s
Description: S/MIME cryptographic 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

[SR-Users] Processing on Invite Packet

[aawaise]

Hello,

On arrival of INVITE packet to the Kamailio server, which module is used to
take in response to Invite packet. Other than record_route, which records
the route.
And what to do if we want to add some actions in response to INVITE packet.

Cheers.



--
View this message in context: 
http://sip-router.1086192.n5.nabble.com/Processing-on-Invite-Packet-tp127097.html
Sent from the Users mailing list archive at Nabble.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] Difference between route(RELAY) and t_relay

[aawaise]

Hi all,

I want to know while coding kamailio.cfg. What factor determines that
route(RELAY) should be used or t_relay ??

Thanks.

Cheers.



--
View this message in context: 
http://sip-router.1086192.n5.nabble.com/Difference-between-route-RELAY-and-t-relay-tp127098.html
Sent from the Users mailing list archive at Nabble.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] Problem accessing git repository

[arun Jayaprakash]

Is there a problem with the git repository? When I try to download kamailio 
(git clone --depth 1
git://git.sip-router.org/sip-router kamailio ) I get an error saying that 
git.sip-router.org can not be reached. Please let me know, thank you.

Arun___
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] Difference between route(RELAY) and t_relay

[Olle E. Johansson]

On 24 Apr 2014, at 06:58, aawaise  wrote:

> Hi all,
> 
> I want to know while coding kamailio.cfg. What factor determines that
> route(RELAY) should be used or t_relay ??

route(RELAY) is a call to a routeblock that calls t_relay.

Check that route block - what it does in your configuration. Usually it handles 
t_relay errors and is generally more useful than calling t_relay() directly.

/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] Planning release of v4.1.3

[Daniel-Constantin Mierla]

Hello,

soon I will start packaging 4.1.3 -- if anyone has backport to push in 
branch 4.1, reply to sr-dev mailing list.


Cheers,
Daniel

On 22/04/14 09:00, Daniel-Constantin Mierla wrote:

Hello,

short reminder about upcoming release of v4.1.3. If there are patches 
left to be backported or issues that you are aware, report them.


Cheers,
Daniel

On 16/04/14 12:07, Daniel-Constantin Mierla wrote:

Hello,

I am considering releasing v4.1.3 by mid of next week, on Wednesday 
or Thursday (April 23 or 24). If there are issues you are aware of 
and not reported to the bug tracker, add them there asap to 
investigate them.


Also, if you noticed some fixes in the master branch not backported 
yet, report them to the mailing lists.


Cheers,
Daniel





--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda


___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users