Hi Daniel,
I am attaching here my sip trace at client side.Request you to see this.
My client IP in register request is 125.16.231.74 and the server on which
kamailio is running is 185.122.205.178.
I want my contact header to be <sip:OTT919620649614@185.122.205.178:5070;lr>
I am using avp_subst() function to replace this.But kamailio is not starting.
Please reply if I am in wrong way or have different understanding as I am very
new to kamailio.
Thanks,
Narayan
________________________________
From: Narayan P
Sent: Friday, April 14, 2017 12:54:25 PM
To: sr-us...@lists.sip-router.org
Cc: moco...@gmail.com
Subject: RE: Not able to replace route_uri in contact header uri
Hi Daniel,
Thanks for your response.
I mean to say,
In my register request the route header i.e. route_uri is <sip:185.122.205.178>
and contact header is 3333@172.22.13.41.
I want my contact header to be 3333@185.122.205.178.
So how can I replace the contact header's only domain part with route_uri.the
user part of the contact header remains same.
Thanks,
Narayan
________________________________________
From: Narayan P
Sent: Thursday, April 13, 2017 11:13 AM
To: sr-us...@lists.sip-router.org
Subject: Re: Not able to replace route_uri in contact header uri
Hi ,
Can anybody help me how to replace the domain part of $route-uri in contact
header .
I am using avp_subst().But kamailio is not starting.
I am new to kamailio.Any help will be highly appreciated.
Thanks,
Narayan
________________________________
From: Narayan P
Sent: Wednesday, April 12, 2017 8:17:02 AM
To: sr-us...@lists.sip-router.org
Subject: Not able to replace route_uri in contact header uri
Hi All,
I am using pjsip client and kamailiio,both running on different servers.I want
to replace the uri on which pjsip is running with the uri on which kamailio is
running in contact header.
I put some logs on route[REGISTRAR] and able to see the $route_uri as kamailio
server uri($avp(s:fs) and $hdr(Contact) as pjsip server uri
($avp(s:contact).But I am not able to replace the uri with avp_subst() function.
Can anybody help me how to replace the uri or with any any other function.
Below I have put the snippet of config file.
# Handle SIP registrations
route[REGISTRAR] {
if (!is_method("REGISTER")) return;
if(isflagset(FLT_NATS)) {
setbflag(FLB_NATB);
#!ifdef WITH_NATSIPPING
# do SIP NAT pinging
setbflag(FLB_NATSIPPING);
#!endif
}
xlog("Narayan: before sending contact\n");
$avp(s:contact) = $hdr(Contact);
$avp(s:fs) = $route_uri;
xlog("Narayan: Forced socket is $avp(s:fs)\n");
xlog("Narayan: contact header is $avp(s:contact)\n");
avp_subst("$avp(s:contact)/avp(s:contac)/g", "/(.*)@(.*)/$route_uri/");
#avp_pushto("$ru/domain","$fd");
#rewritehostport("185.122.206.62:5060");
t_on_reply("MANAGE_REPLY");
t_relay();
#if (!save("location")) {
# sl_reply_error();
#}
exit;
}
Any suggestion will be highly appreciated.
Thanks,
Narayan
root@dharm-VirtualBox:/home/dharm/rudra/pjsip_comp/pjsip-apps/bin#
root@dharm-VirtualBox:/home/dharm/rudra/pjsip_comp/pjsip-apps/bin#
root@dharm-VirtualBox:/home/dharm/rudra/pjsip_comp/pjsip-apps/bin#
root@dharm-VirtualBox:/home/dharm/rudra/pjsip_comp/pjsip-apps/bin#
./pjsua-x86_64-unknown-linux-gnu --local-port=5067
--outbound=sip:185.122.205.178:5070 --config-file=config_cfg.cfg --no-tcp
11:49:40.373 os_core_unix.c !pjlib 2.3 for POSIX initialized
11:49:40.373 sip_endpoint.c .Creating endpoint instance...
11:49:40.373 pjlib .select() I/O Queue created (0xe95240)
11:49:40.373 sip_endpoint.c .Module "mod-msg-print" registered
11:49:40.373 sip_transport. .Transport manager created.
11:49:40.373 pjsua_core.c .PJSUA state changed: NULL --> CREATED
11:49:40.373 pjsua_core.c .pjsua_init
11:49:40.373 pjsua_core.c .rtpCompStatus 0 rtpEncStatus 0 sipCompressStatus
1 encryptionState 1
11:49:40.373 sip_endpoint.c .Module "mod-pjsua-log" registered
11:49:40.373 sip_endpoint.c .Module "mod-tsx-layer" registered
11:49:40.373 sip_endpoint.c .Module "mod-stateful-util" registered
11:49:40.373 sip_endpoint.c .Module "mod-ua" registered
11:49:40.373 sip_endpoint.c .Module "mod-100rel" registered
11:49:40.373 sip_endpoint.c .Module "mod-pjsua" registered
11:49:40.373 sip_endpoint.c .Module "mod-invite" registered
bt_audio_service_open: connect() failed: Connection refused (111)
bt_audio_service_open: connect() failed: Connection refused (111)
bt_audio_service_open: connect() failed: Connection refused (111)
bt_audio_service_open: connect() failed: Connection refused (111)
11:49:40.427 pa_dev.c ..PortAudio sound library initialized, status=0
11:49:40.428 pa_dev.c ..PortAudio host api count=2
11:49:40.428 pa_dev.c ..Sound device count=11
11:49:40.428 pjlib ..select() I/O Queue created (0xeddd98)
11:49:40.434 sip_endpoint.c .Module "mod-evsub" registered
11:49:40.436 sip_endpoint.c .Module "mod-presence" registered
11:49:40.436 sip_endpoint.c .Module "mod-mwi" registered
11:49:40.436 sip_endpoint.c .Module "mod-refer" registered
11:49:40.436 sip_endpoint.c .Module "mod-pjsua-pres" registered
11:49:40.436 sip_endpoint.c .Module "mod-pjsua-im" registered
11:49:40.436 sip_endpoint.c .Module "mod-pjsua-options" registered
11:49:40.436 pjsua_core.c .1 SIP worker threads created
11:49:40.436 pjsua_core.c .pjsua version 2.3 for
Linux-4.4.0.64/x86_64/glibc-2.19 initialized
11:49:40.436 pjsua_core.c .PJSUA state changed: CREATED --> INIT
11:49:40.436 sip_endpoint.c Module "mod-default-handler" registered
11:49:40.436 pjsua_core.c SIP UDP socket reachable at 172.22.13.41:5067
11:49:40.436 udp0xef1d80 SIP UDP transport started, published address is
172.22.13.41:5067
11:49:40.436 pjsua_acc.c Adding account: id=<sip:172.22.13.41:5067>
11:49:40.436 pjsua_acc.c .Account <sip:172.22.13.41:5067> added with id 0
11:49:40.436 pjsua_acc.c Modifying accunt 0
11:49:40.436 pjsua_acc.c Acc 0: setting online status to 1..
11:49:40.436 pjsua_acc.c Adding account:
id=sip:OTT919620649614@185.122.206.62
11:49:40.436 pjsua_acc.c .Account sip:OTT919620649614@185.122.206.62 added
with id 1
11:49:40.436 pjsua_acc.c .Acc 1: setting registration..
11:49:40.436 pjsua_core.c ...TX 628 bytes Request msg REGISTER/cseq=41800
(tdta0xef79f0) to UDP 185.122.205.178:5070:
REGISTER sip:185.122.206.62 SIP/2.0
Via: SIP/2.0/UDP
172.22.13.41:5067;rport;branch=z9hG4bKPj475160b1-4579-48ba-8c56-6858389d292f
Route: <sip:185.122.205.178:5070;lr>
Max-Forwards: 70
From:
<sip:OTT919620649614@185.122.206.62>;tag=35af1666-a2b1-4c6e-a7b6-8675845036e7
To: <sip:OTT919620649614@185.122.206.62>
Call-ID: 5bc6558f-1ffc-4990-bc99-d099c4fdcbcb
CSeq: 41800 REGISTER
User-Agent: PJSUA v2.3 Linux-4.4.0.64/x86_64/glibc-2.19
Contact: <sip:OTT919620649614@172.22.13.41:5067;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Content-Length: 0
--end msg--
11:49:40.436 pjsua_acc.c ..Acc 1: Registration sent
11:49:40.436 pjsua_acc.c Acc 1: setting online status to 1..
11:49:40.436 pjsua_core.c PJSUA state changed: INIT --> STARTING
11:49:40.436 sip_endpoint.c .Module "mod-unsolicited-mwi" registered
11:49:40.436 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING
11:49:40.436 main.c Ready: Success
>>>>
Account list:
[ 0] <sip:172.22.13.41:5067>: does not register
Online status: Online
*[ 1] sip:OTT919620649614@185.122.206.62: 100/In Progress (expires=0)
Online status: Online
Buddy list:
-none-
+=============================================================================+
| Call Commands: | Buddy, IM & Presence: | Account: |
| | | |
| m Make new call | +b Add new buddy .| +a Add new accnt |
| M Make multiple calls | -b Delete buddy | -a Delete accnt. |
| a Answer call | i Send IM | !a Modify accnt. |
| h Hangup call (ha=all) | s Subscribe presence | rr (Re-)register |
| H Hold call | u Unsubscribe presence | ru Unregister |
| v re-inVite (release hold) | t ToGgle Online status | > Cycle next ac.|
| U send UPDATE | T Set online status | < Cycle prev ac.|
| ],[ Select next/prev call +--------------------------+-------------------+
| x Xfer call | Media Commands: | Status & Config: |
| X Xfer with Replaces | | |
| # Send RFC 2833 DTMF | cl List ports | d Dump status |
| * Send DTMF with INFO | cc Connect port | dd Dump detailed |
| dq Dump curr. call quality | cd Disconnect port | dc Dump config |
| | V Adjust audio Volume | f Save config |
| S Send arbitrary REQUEST | Cp Codec priorities | |
+-----------------------------------------------------------------------------+
| q QUIT L ReLoad sleep MS echo [0|1|txt] n: detect NAT type |
+=============================================================================+
You have 0 active call
>>> 11:49:40.600 pjsua_core.c .RX 493 bytes Response msg
>>> 401/REGISTER/cseq=41800 (rdata0xef3808) from UDP 185.122.205.178:5070:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
172.22.13.41:5067;rport=25841;branch=z9hG4bKPj475160b1-4579-48ba-8c56-6858389d292f;received=125.16.231.74
From:
<sip:OTT919620649614@185.122.206.62>;tag=35af1666-a2b1-4c6e-a7b6-8675845036e7
To: <sip:OTT919620649614@185.122.206.62>
Call-ID: 5bc6558f-1ffc-4990-bc99-d099c4fdcbcb
CSeq: 41800 REGISTER
WWW-Authenticate: Digest
nonce="af7780946d2b72ddc5e765a68798e937",realm="185.122.206.62",opaque="",stale=FALSE,algorithm=MD5
Content-Length: 0
--end msg--
11:49:40.600 pjsua_acc.c ....IP address change detected for account 1
(172.22.13.41:5067 --> 125.16.231.74:25841). Updating registration (using
method 4)
11:49:40.600 pjsua_core.c ....TX 834 bytes Request msg REGISTER/cseq=41801
(tdta0xef79f0) to UDP 185.122.205.178:5070:
REGISTER sip:185.122.206.62 SIP/2.0
Via: SIP/2.0/UDP
125.16.231.74:25841;rport;branch=z9hG4bKPj592750e4-9a06-4662-a55e-a8a71dedb974
Route: <sip:185.122.205.178:5070;lr>
Max-Forwards: 70
From:
<sip:OTT919620649614@185.122.206.62>;tag=35af1666-a2b1-4c6e-a7b6-8675845036e7
To: <sip:OTT919620649614@185.122.206.62>
Call-ID: 5bc6558f-1ffc-4990-bc99-d099c4fdcbcb
CSeq: 41801 REGISTER
User-Agent: PJSUA v2.3 Linux-4.4.0.64/x86_64/glibc-2.19
Contact: <sip:OTT919620649614@125.16.231.74:25841;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Authorization: Digest username="OTT919620649614", realm="185.122.206.62",
nonce="af7780946d2b72ddc5e765a68798e937", uri="sip:185.122.206.62",
response="a4c0614fbc3c72ece619247de5766a4b", algorithm=MD5
Content-Length: 0
--end msg--
11:49:40.768 pjsua_core.c .RX 456 bytes Response msg 200/REGISTER/cseq=41801
(rdata0x7f1190001d08) from UDP 185.122.205.178:5070:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
125.16.231.74:25841;rport=25841;branch=z9hG4bKPj592750e4-9a06-4662-a55e-a8a71dedb974;received=125.16.231.74
From:
<sip:OTT919620649614@185.122.206.62>;tag=35af1666-a2b1-4c6e-a7b6-8675845036e7
Call-ID: 5bc6558f-1ffc-4990-bc99-d099c4fdcbcb
CSeq: 41801 REGISTER
To: <sip:OTT919620649614@185.122.206.62>;tag=1492409213989
Expires: 50
Contact: <sip:OTT919620649614@125.16.231.74:25841;ob>;expires=50
Content-Length: 0
--end msg--
11:49:40.769 pjsua_acc.c ....SIP outbound status for acc 1 is not active
11:49:40.769 pjsua_acc.c ....sip:OTT919620649614@185.122.206.62:
registration success, status=200 (OK), will re-register in 50 seconds
11:49:40.769 pjsua_acc.c ....Keep-alive timer started for acc 1,
destination:185.122.205.178:5070, interval:15s
^C
root@dharm-VirtualBox:/home/dharm/rudra/pjsip_comp/pjsip-apps/bin#
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