Re: [OpenSIPS-Users] need advice on B2b
HI Anca
i am trying to use the b2b_request + b2b_reply
route{
...
if(is_method(INVITE) !(src_ip == 192.168.1.249 src_port ==5060))
{
if (! t_newtran()){
sl_reply_error();
exit;
};
b2b_init_request(top hiding);
exit;
};
route(1);
}
route[1] {
if(is_method(INVITE)) {
route(b2b_request);
t_on_reply(2);
}
else
if(status==200)
route(b2b_reply);
if (!t_relay()) {
sl_reply_error();
};
exit;
}
route[b2b_request] {
xlog(b2b_request cucku ($ci)\n);
force_rtp_proxy();
}
route[b2b_reply] {
xlog(b2b_reply cucku ($ci)\n);
force_rtp_proxy();
}
i get the errors :
ERROR:nathelper:force_rtp_proxy: Unable to parse body
and
DBG:tm:utimer_routine: timer routine:4,tl=0xb615e9a8 next=(nil),
timeout=2900
DBG:tm:timer_routine: timer routine:3,tl=0xb615c6f4 next=(nil), timeout=29
DBG:tm:delete_handler: removing 0xb615c690
DBG:tm:delete_cell: delete_cell 0xb615c690: can't delete -- still reffed (1)
===
DBG:core:get_hdr_field: cseq CSeq: 2 INVITE
DBG:core:parse_headers: flags=8
DBG:tm:t_reply_matching: hash 21530 label 76806763 branch 0
DBG:tm:t_reply_matching: REF_UNSAFE: after is 2
DBG:tm:t_reply_matching: reply matched (T=0xb615e85c)!
DBG:tm:t_check: end=0xb615e85c
DBG:tm:reply_received: org. status uas=0, uac[0]=100 local=2 is_invite=1)
DBG:tm:t_should_relay_response: T_code=0, new_code=180
DBG:tm:local_reply: branch=0, save=0, winner=0
DBG:tm:local_reply: Passing provisional reply 180 to FIFO application
DBG:tm:run_trans_callbacks: trans=0xb615e85c, callback type 1024, id 0 entered
DBG:b2b_entities:b2b_parse_key: hash_index = [111] - local_index= [0]
DBG:core:parse_headers: flags=
DBG:core:get_hdr_field: content_length=0
DBG:core:get_hdr_field: found end of header
DBG:b2b_entities:b2b_tm_cback: Received a reply with statuscode = 180
DBG:core:parse_headers: flags=
DBG:b2b_entities:b2b_new_dlg: 'To' header ALREADY PARSED:
sip:1...@192.168.1.249
DBG:b2b_entities:b2b_new_dlg: Not an initial request
DBG:core:parse_to_param: tag=bfad35cdb22f09f741816636d344f54b-19f0
DBG:core:parse_to: end of header reached, state=29
DBG:core:parse_to: display={}, ruri={sip:0873000...@192.168.1.249;user=phone}
DBG:core:print_rr_body: current rr is sip:192.168.1.249;lr=on
DBG:core:print_rr_body: out rr [sip:192.168.1.249;lr=on]
DBG:core:print_rr_body: we have 1 records
DBG:b2b_entities:b2b_tm_cback: Created new dialog structure 0xb61618c0
DBG:core:print_rr_body: current rr is sip:192.168.1.249;lr=on
DBG:core:print_rr_body: out rr [sip:192.168.1.249;lr=on]
DBG:core:print_rr_body: we have 1 records
DBG:b2b_logic:b2bl_parse_key: hash_index = [623] - local_index= [0]
DBG:core:parse_headers: flags=
DBG:b2b_entities:b2b_parse_key: hash_index = [346] - local_index= [0]
DBG:core:parse_headers: flags=
DBG:core:check_ip_address: params 192.168.1.4, 192.168.1.4, 0
DBG:tm:t_reply_with_body: buffer computed
DBG:tm:_reply_light: reply sent out. buf=0x81c70b8: SIP/2.0 1...,
shmem=0xb615e534: SIP/2.0 1
DBG:tm:_reply_light: finished
b2b_reply cucku (B2B.111.0.1262765386)
DBG:core:parse_headers: flags=
DBG:core:parse_headers: flags=1000
DBG:core:parse_content_type_hdr: missing Content-Type header
ERROR:nathelper:force_rtp_proxy: Unable to parse body
Thank you
Ha`
--- On Mon, 1/4/10, Anca Vamanu a...@opensips.org wrote:
From: Anca Vamanu a...@opensips.org
Subject: Re: [OpenSIPS-Users] need advice on B2b
To: OpenSIPS users mailling list users@lists.opensips.org
Date: Monday, January 4, 2010, 3:04 AM
Hi Ha`,
There is a very simple example in the documentation:
route[b2b_request] {
xlog(b2b_request ($ci)\n);
}
route[b2b_reply] {
xlog(b2b_reply ($ci)\n);
}
You can call in these routes any function that you call in a request route.
Regards,
--
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www.voice-system.ro
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Re: [OpenSIPS-Users] Need help on flag in usrloc
Hi Bogdan
got it :)
1 more question about the flag
modparam(nathelper, sipping_bflag, 7)
modparam(usrloc, nat_bflag, 7)
the modem ADSL will close the port after 3 mins(some minutes), so Opensips
should send OPTION message(sip ping) to modem to keep port that should open for
UA
the sipping_blag of nathelper module should be the same value as nat_bflag of
usrloc ? or the cflag of usrloc just has a value??
Thank you
Ha`
--- On Tue, 1/5/10, Bogdan-Andrei Iancu bog...@voice-system.ro wrote:
From: Bogdan-Andrei Iancu bog...@voice-system.ro
Subject: Re: [OpenSIPS-Users] Need help on flag in usrloc
To: OpenSIPS users mailling list users@lists.opensips.org
Date: Tuesday, January 5, 2010, 7:20 AM
Hi Ha,
the NAT branch flag you use is 7 (nat_bflag) and in usrloc you find the
branch flags in the cflags (contact flags) field. The cflags is a mask
with all the branch flags: 192 = 128 (2^7) + 64 (2^6)
Regards,
Bogdan
ha do wrote:
Hi all
i am successfull to check the UA behind NAT but i dont know what value
of the flag will be stored in the usrloc
Could someone please let me know the value of Nated UA flag, that is
stored in usrloc
my config :
modparam(nathelper, natping_interval,180)
modparam(nathelper, ping_nated_only, 1)
modparam(nathelper, sipping_bflag, 7)
modparam(nathelper, sipping_from, sip:cu...@kamailio.org)
modparam(registrar|nathelper, received_avp, $avp(i:80))
modparam(usrloc, nat_bflag, 7)
route{
route(4);
if (method==REGISTER)
{
if (isflagset(5)) {
setbflag(6);
setbflag(7);
}
if (!save(location))
sl_reply_error();
exit;
}
}
route[4]{
force_rport();
if (nat_uac_test(19)) {
if (method==REGISTER) {
fix_nated_register();
} else {
fix_nated_contact();
}
setflag(5);
}
return;
}
mysql select * from location\G
*** 1. row ***
id: 12
username: 1000
domain: NULL
contact: sip:1...@192.168.1.2:12280;rinstance=a4752f45bdc3ddd3
received: sip:210.245.35.150:12280
path: NULL
expires: 2010-01-05 17:48:56
q: -1.00
callid: YjE1NWJmYWYyYWMzZDg2ZDc5MjY0NDQyMGE5NDEwM2E.
cseq: 2
last_modified: 2010-01-05 16:48:56
flags: 0
cflags: 192
user_agent: eyeBeam release 1004p stamp 31962
socket: udp:118.69.193.198:5060
methods: 5951
Thank you
Ha`
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www.voice-system.ro
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Re: [OpenSIPS-Users] need advice on B2b
Hi Ha`,
The excerpt from your script shows that you don't have a good
understanding of the opensips scripting logic. First, the *route* block
will only be called for SIP Requests.
Calling this is not right:
/ if(status==200)
route(b2b_reply);
/The replies will go into reply route blocks. You can have a default
reply route ( one without an index or with index 0), or you can specify
a certain reply route for a request by calling t_on_reply.
Second, you don't understand what happens with b2b request and replies.
It is explained in the documentation:
/The requests and replies that are received by the B2BUA server,
belonging to the dialogs it is handling will not go into the script as
normal request do. The reason for this is that this are not normal
requests where the server is a proxy, but the server is an endpoint in
the dialog and therefore they should not go through the same routes.
However, it is normal for this request to be seen from the script and
allow the script writer to do the processing it desires based on them.
For this, it is possible to define two special B2B routes - one for
requests and one for replies. The routes are of type *route* and have
their name defined in the modules parameters *script_req_route* and
*script_reply_route*. /
In other words, there are two important things:
1. the B2B requests/replies will not go into the default request/reply
route block.
2. the b2b_request/b2_reply route will be called automatically for every
request/reply targeted to the b2b agent
So, for your script, you don't need this lines:
/ if(is_method(INVITE)) {
route(b2b_request);
t_on_reply(2);
}
else
if(status==200)
route(b2b_reply);/
Hope this made things a bit clearer.
Regards,
--
Anca Vamanu
www.voice-system.ro
ha do wrote:
HI Anca
i am trying to use the b2b_request + b2b_reply
route{
...
if(is_method(INVITE) !(src_ip == 192.168.1.249 src_port ==5060))
{
if (! t_newtran()){
sl_reply_error();
exit;
};
b2b_init_request(top hiding);
exit;
};
route(1);
}
route[1] {
if(is_method(INVITE)) {
route(b2b_request);
t_on_reply(2);
}
else
if(status==200)
route(b2b_reply);
if (!t_relay()) {
sl_reply_error();
};
exit;
}
route[b2b_request] {
xlog(b2b_request cucku ($ci)\n);
force_rtp_proxy();
}
route[b2b_reply] {
xlog(b2b_reply cucku ($ci)\n);
force_rtp_proxy();
}
i get the errors :
ERROR:nathelper:force_rtp_proxy: Unable to parse body
and
DBG:tm:utimer_routine: timer routine:4,tl=0xb615e9a8 next=(nil),
timeout=2900
DBG:tm:timer_routine: timer routine:3,tl=0xb615c6f4 next=(nil),
timeout=29
DBG:tm:delete_handler: removing 0xb615c690
DBG:tm:delete_cell: delete_cell 0xb615c690: can't delete -- still
reffed (1)
===
DBG:core:get_hdr_field: cseq CSeq: 2 INVITE
DBG:core:parse_headers: flags=8
DBG:tm:t_reply_matching: hash 21530 label 76806763 branch 0
DBG:tm:t_reply_matching: REF_UNSAFE: after is 2
DBG:tm:t_reply_matching: reply matched (T=0xb615e85c)!
DBG:tm:t_check: end=0xb615e85c
DBG:tm:reply_received: org. status uas=0, uac[0]=100 local=2 is_invite=1)
DBG:tm:t_should_relay_response: T_code=0, new_code=180
DBG:tm:local_reply: branch=0, save=0, winner=0
DBG:tm:local_reply: Passing provisional reply 180 to FIFO application
DBG:tm:run_trans_callbacks: trans=0xb615e85c, callback type 1024, id
0 entered
DBG:b2b_entities:b2b_parse_key: hash_index = [111] - local_index= [0]
DBG:core:parse_headers: flags=
DBG:core:get_hdr_field: content_length=0
DBG:core:get_hdr_field: found end of header
DBG:b2b_entities:b2b_tm_cback: Received a reply with statuscode = 180
DBG:core:parse_headers: flags=
DBG:b2b_entities:b2b_new_dlg: 'To' header ALREADY PARSED:
sip:1...@192.168.1.249
DBG:b2b_entities:b2b_new_dlg: Not an initial request
DBG:core:parse_to_param: tag=bfad35cdb22f09f741816636d344f54b-19f0
DBG:core:parse_to: end of header reached, state=29
DBG:core:parse_to: display={},
ruri={sip:0873000...@192.168.1.249;user=phone}
DBG:core:print_rr_body: current rr is sip:192.168.1.249;lr=on
DBG:core:print_rr_body: out rr [sip:192.168.1.249;lr=on]
DBG:core:print_rr_body: we have 1 records
DBG:b2b_entities:b2b_tm_cback: Created new dialog structure 0xb61618c0
DBG:core:print_rr_body: current rr is sip:192.168.1.249;lr=on
DBG:core:print_rr_body: out rr [sip:192.168.1.249;lr=on]
DBG:core:print_rr_body: we have 1 records
DBG:b2b_logic:b2bl_parse_key: hash_index = [623] - local_index= [0]
DBG:core:parse_headers: flags=
DBG:b2b_entities:b2b_parse_key: hash_index = [346] - local_index= [0]
Re: [OpenSIPS-Users] Need help on flag in usrloc
Hi Ha,
the two flags are different and may have different values - one is used
as NAT marker, the other one is used as SIP-based pinging marker.
so, you can use different flags and both of them will be saved in cflag
mask.
Regards,
Bogdan
ha do wrote:
Hi Bogdan
got it :)
1 more question about the flag
modparam(nathelper, sipping_bflag, 7)
modparam(usrloc, nat_bflag, 7)
the modem ADSL will close the port after 3 mins(some minutes), so
Opensips should send OPTION message(sip ping) to modem to keep port
that should open for UA
the sipping_blag of nathelper module should be the same value as
nat_bflag of usrloc ? or the cflag of usrloc just has a value??
Thank you
Ha`
--- On *Tue, 1/5/10, Bogdan-Andrei Iancu /bog...@voice-system.ro/*
wrote:
From: Bogdan-Andrei Iancu bog...@voice-system.ro
Subject: Re: [OpenSIPS-Users] Need help on flag in usrloc
To: OpenSIPS users mailling list users@lists.opensips.org
Date: Tuesday, January 5, 2010, 7:20 AM
Hi Ha,
the NAT branch flag you use is 7 (nat_bflag) and in usrloc you
find the
branch flags in the cflags (contact flags) field. The cflags is a
mask
with all the branch flags: 192 = 128 (2^7) + 64 (2^6)
Regards,
Bogdan
ha do wrote:
Hi all
i am successfull to check the UA behind NAT but i dont know what
value
of the flag will be stored in the usrloc
Could someone please let me know the value of Nated UA flag,
that is
stored in usrloc
my config :
modparam(nathelper, natping_interval,180)
modparam(nathelper, ping_nated_only, 1)
modparam(nathelper, sipping_bflag, 7)
modparam(nathelper, sipping_from, sip:cu...@kamailio.org
/mc/compose?to=cu...@kamailio.org)
modparam(registrar|nathelper, received_avp, $avp(i:80))
modparam(usrloc, nat_bflag, 7)
route{
route(4);
if (method==REGISTER)
{
if (isflagset(5)) {
setbflag(6);
setbflag(7);
}
if (!save(location))
sl_reply_error();
exit;
}
}
route[4]{
force_rport();
if (nat_uac_test(19)) {
if (method==REGISTER) {
fix_nated_register();
} else {
fix_nated_contact();
}
setflag(5);
}
return;
}
mysql select * from location\G
*** 1. row ***
id: 12
username: 1000
domain: NULL
contact: sip:1...@192.168.1.2:12280;rinstance=a4752f45bdc3ddd3
received: sip:210.245.35.150:12280
path: NULL
expires: 2010-01-05 17:48:56
q: -1.00
callid: YjE1NWJmYWYyYWMzZDg2ZDc5MjY0NDQyMGE5NDEwM2E.
cseq: 2
last_modified: 2010-01-05 16:48:56
flags: 0
cflags: 192
user_agent: eyeBeam release 1004p stamp 31962
socket: udp:118.69.193.198:5060
methods: 5951
Thank you
Ha`
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www.voice-system.ro
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[OpenSIPS-Users] How to limit channel on bunch of called DIDs?
Hi! I'm a newbie with OpenSIPS administration and configuration and I searched on the mail archives regarding limiting the channels but only found the site regarding Concurrent calls limitation. I've been trying to grasp the whole idea about AVPops and dialog module but unfortunately I'm having a hard time understanding it. If ever, does anyone here be kind enough to point me to any available documents or sample configurations file that will help me limit the channels on inbound calls to a single or group of DIDs? Implementation will be done static first and hopefully once I got the whole idea, will be implementing a dynamic one. Thank you very much. --conpaj-- ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] How to increase cache in opensips tables
Hi,
I'm using permission module's function check_source_address(), the problem
I'm facing is that I can not add not than more 8 IPs in address table, but I
want to permit more than 100 IPs. I only want to use these IPs on group 0
what I am using. When I enter more than 8 IPs in address table and make a
call, I observe a message i.e. not found in hash table.My opensips.cfg
configuration for check_source_address() is listed below;
if (is_method(INVITE) check_source_address(0)) {
log(INVITE###);
ds_select_domain(1,4);
forward();
route(1);
log(#END);
setflag(1);
}
Kindly advise me how to increase cache of OpenSIPs database tables so I can
reslove my case. Further added, how can I enter domain name in 'ip' column
section of address table i.e. abc.com can't be used and gives me an error,
kindly advise this well.
--
Regards,
Ahmed Munir
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[OpenSIPS-Users] INVITE as a variable?
Hi list, Im trying to find a way to extract and the re-write the number in the INVITE itself, Mera and a few other switch manufacturers are all of a sudden are trying to route on INVITE and not the To field, when the INVITE has a prefix in it they fail to complete the call then. I can't seem to find any built in variables or way to do that, is it just not possible to rewrite the INVITE or am I missing something? It would seem pretty straight forward but no way to do this, am I just blind? Thanks for any help ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] INVITE as a variable?
On the contrary, rewriting the Request URI (which is what I assume you mean by rewrite the number in the INVITE itself), including the user part of the Request URI (the part before the @), is one of the most basic functions the proxy *can* perform. See the pseudovariables $ru (and for the user part specifically, $rU, or $rd for the domain and $rp for the port). They are mutable: $rU = 39824234234; $rd = sip.biloxi.net; -- Alex On 01/06/2010 09:52 AM, Brad Bendy wrote: Hi list, Im trying to find a way to extract and the re-write the number in the INVITE itself, Mera and a few other switch manufacturers are all of a sudden are trying to route on INVITE and not the To field, when the INVITE has a prefix in it they fail to complete the call then. I can't seem to find any built in variables or way to do that, is it just not possible to rewrite the INVITE or am I missing something? It would seem pretty straight forward but no way to do this, am I just blind? Thanks for any help ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to limit channel on bunch of called DIDs?
Hi Johnson, The idea is to use dialog profiles to keep trace of the ongoing calls (per resource). As a first step take a look at the following tutorial: http://www.opensips.org/Resources/DocsTutConcurrentCalls Regards, Bogdan Johnson Pajayat wrote: Hi! I'm a newbie with OpenSIPS administration and configuration and I searched on the mail archives regarding limiting the channels but only found the site regarding Concurrent calls limitation. I've been trying to grasp the whole idea about AVPops and dialog module but unfortunately I'm having a hard time understanding it. If ever, does anyone here be kind enough to point me to any available documents or sample configurations file that will help me limit the channels on inbound calls to a single or group of DIDs? Implementation will be done static first and hopefully once I got the whole idea, will be implementing a dynamic one. Thank you very much. --conpaj-- ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] nathelper+rtpproxy between IPv6 client and IPv4 client
Hi, for bridging IPv4 and IPv6 you can use opensips (nathelper) for doing the signalling part and rtpproxy for doing the RTP part. You need to configure opensips to listen both on an IPv4 and IPv6 interface (and do the routing between). Also RTPproxy does bridging between 2 interface (ipv4 and ipv6). Regards, Bogdan agung nugroho wrote: I want to ask you all about nathelper modules. Is it can be using with rtpproxy?? i reasearch about how to connect IPv6 client with IPv4 client using IMS. And they can communicate with tripleplay. what is client software that i can use?? coz i use monster as a client but it cannnot connect to the server. can anyone help me??? sorry for my bad english. Thanks before. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPS 1.6 drouting: drouting:dr_load_routing_info: route 1 does not exist
Hi All I'm trying to get drouting working, the issue is I can not work out which tables and field its complaining about when it give the error. 'drouting:dr_load_routing_info: route 1 does not exist' In 'dr_rules' I have a routeid of 1 for a number of rows, and if I change this to a 2 or 5 then the above error changes. So the issue must be that I have not place a route some where with the id of 1, problem I can not find where to add this record. Any help greatly appreciated. Mike ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to limit channel on bunch of called DIDs?
Hello Bogdan,
I appreciate a lot your response regarding my inquiry. I've been reading
that tutorial as well as the AVPops and dialog modules documentation for
about a month now. I tried to adapt that route block for inbound calls and
here's a portion of what I have on our OpenSIPS 1.5 config file:
---
modparam(dialog, dlg_flag, 4)
modparam(dialog, profiles_with_value, inbound)
..
} else if (uri=~sip:1234567...@.*) {
route(3);
rewritehost(111.222.111.222);
...
route[3]
{
## have we done our checking on this call?
if(!isflagset(31))
{
# user has max channel limit set as preference
if(is_avp_set($avp(s:channels)/n)
avp_check($avp(s:channels), gt/i:0))
{
# get the current calls for DID
get_profile_size(inbound,$tU,$var(calls));
# check within limit
if($avp(s:channels) $var(calls))
{
xlog(L_INFO, Call control: DID '$tU'
currently has
'$var(calls)' of '$avp(s:channels)'
active calls before this one\n);
$var(setprofile) = 1;
}
else
{
xlog(L_INFO, Call control: DID '$tU'
channel limit exceeded [$var(calls)/$avp(s:channels)]
\n);
send_reply(487, Request Terminated:
Channel limit exceeded\n);
exit;
}
}
else
{
$var(setprofile) = 0;
}
call_control();
switch ($retcode)
{
case 2:
# Call with no limit
case 1:
# Call with a limit under callcontrol
management (either prepaid or postpaid)
break;
case -1:
# Not enough credit (prepaid call)
xlog(L_INFO, Call control: not enough
credit for prepaid call\n);
acc_rad_request(402);
sl_send_reply(402, Not enough credit);
exit;
break;
case -2:
# Locked by call in progress (prepaid call)
xlog(L_INFO, Call control: prepaid call
locked by another call in progress\n);
acc_rad_request(403);
sl_send_reply(403, Call locked by another
call in progress);
exit;
break;
default:
# Internal error (message parsing,
communication, ...)
xlog(L_INFO, Call control: internal
server error\n);
acc_rad_request(500);
sl_send_reply(500, Internal server
error);
exit;
}
if($var(setprofile) 0)
{
create_dialog();
set_dlg_profile(inbound,$tU);
}
## mark checking done
setflag(31);
}
}
And here are the logs appearing on /var/log/messages:
Jan 6 05:53:22 opensips cdrtool[20998]: MaxSessionTime Duration=36000
callid=28a35ce1-4d7a175-25a58...@124.123.234.229 From=si
p:12135551...@124.123.234.229
p%3a12135551...@124.123.234.229Gateway=124.123.234.229
To=sip:1234567...@124.123.234.241:5060
;user=phone
Jan 6 05:53:22 opensips cdrtool[20998]: MaxSessionTime=unlimited
Type=postpaid callid=28a35ce1-4d7a175-25a58...@124.123.234.229 BillingParty=
12135551...@124.123.234.229
Jan 6 05:53:22 opensips call-control[21263]: Call id
28a35ce1-4d7a175-25a58...@64.77.234.229 of 12135551...@124.123.234.229 to
sip:1234567...@124.123.234.241:5060;user=phone is postpaid not limited
Jan 6 05:53:22 opensips /usr/local/sbin/opensips[1636]: new branch at
sip:1234567...@111.222.111.222:5060;user=phone
Jan 6 05:53:22 opensips /usr/local/sbin/opensips[1640]: incoming reply
Jan 6 05:53:22 opensips /usr/local/sbin/opensips[1636]: incoming reply
Jan 6 05:53:22 opensips /usr/local/sbin/opensips[1636]: ACC: transaction
answered: timestamp=1262786002;method=INVITE;from_t
ag=14fe61da-25a58684-e5ea4d40;to_tag=as6a53f46c;call_id=
28a35ce1-4d7a175-25a58...@64.77.234.229;code=200;reason=OK
Jan 6 05:53:22 opensips
