If I create a dialog with in-dilaog pings like create_dialog("pPB") . Service
got crashed in local route.
(gdb) bt
#0 0x in ?? ()
#1 0x7ff20c965b3d in replicate_dialog_cseq_updated
(dlg=dlg@entry=0x7ff211d34af8, leg=leg@entry=0) at dlg_replication.c:858
#2 0x7ff20c90b
Opensips Users,
Waiting for your comments/help/Suggestion.
Regards
Hamid R. Hashmi
From: Hamid Hashmi
Sent: Tuesday, July 30, 2019 12:40 PM
To: users@lists.opensips.org
Subject: Unable to parse JSON
Below is the JSON response received in an async rest post
Below is the JSON response received in an async rest post request (mod
rest_client).
xlog("L_INFO","[$fU $tU $ci $rm] PUSH Server responded with
$var(rcode):$json(response)");
{
"error": null,
"payload": {
"results": [
{
"error": {
"code": "messaging/registratio
Hi
I need to set advertised address in b2b_init_request("Top Hiding") function. so
that the request going out must have public IP while it uses private IP for
inter-network communication.
I am facing issue while PRACK is requested from Carrier.
1. if I don't set advertised IP with interface
SI
I am trying to load test TLS listener through sipp. REGISTER Request is
working on TLS but INVITE is not working. I changed the transport to TCP and
receive following logs
version: opensips 2.4.3 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC,
FAST_
Really Sorry. I did not see following lines before. :p
/bin/sh: 1: pkg-config: not found
/bin/sh: 1: pkg-config: not found
Regards
Hamid R. Hashmi
__
www.hrhashmi.blogspot.com
Mobile: +92 300 968 22 85 ; +92 322 636 32 66
Email: hamid.hashmi...@gmail.com; hamid2k
I have been facing following error while compiling
make[2]: Entering directory '/usr/local/src/opensips/modules/proto_tls'
/bin/sh: 1: pkg-config: not found
/bin/sh: 1: pkg-config: not found
Compiling proto_tls.c
In file included from /usr/local/ssl/include/openssl/ssl.h:1518:0,
fr
Please define following valuestls_ca_list = "/path/to/file"
tls_method = tlsv1for details please consult
http://www.opensips.org/html/docs/tutorials/tls-1.4.x.html
RegardsHamid R. Hashmi
Date: Thu, 7 Apr 2016 13:14:28 -0400
From: ali...@gmail.com
To: users@lists.opensips.org
Subject: [O
Trying to implement call hold feature and facing an issue of media timeout
after 45 sec on RTPproxy (-T 45). If I increase the media timeout time on
RTPporxy ( -T 3600) then it fulfill our call hold requirement but disturbs
media timeout settings for other calls.
Is there any other solution to
listening only on the private IP in EC2 ? do you use advertise ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 23.02.2016 07:33, Hamid Hashmi
wrote:
I am using opensips as an
edge
I am using opensips as an edge proxy on Amazon EC2 (server behind NAT) and it
works fine if It only works on UDP or TCP. but if I need to translate from TCP
-> UDP, ACK does not reach proxy server due to private IP in record-route.
EdgeProxy -> B#
INVITE sip:923003000200@182.185.200.42:46488;tr
d R. HashmiSoftware Engineer - VoIPVopium A/S
To: users@lists.opensips.org
From: raz...@opensips.org
Date: Fri, 19 Feb 2016 10:23:28 +0200
Subject: Re: [OpenSIPS-Users] What cause TCP connection reset ?
This situation appears when a SIP message is sent in more than 4
"chunks&qu
Why opensips send TCP connection reset packet ?
I have been facing an error of
ERROR:proto_tls:tcp_handle_req: Made 4 read attempts but message is not
complete yet - closing connection
while wireshark traces shows that a TCP connection reset packet was sent back
to UAC by opensips server.
or related to
TLS, but not directly related to configuration.
Nabeel
On 12 Feb 2016 6:45 am, "Hamid Hashmi" wrote:
Nabeel
I dont know how to present a certificate from client. I have tried using Xoiper
(Android - Free), SFLphone (Ubuntu) and CsipSimple (Android) but there was
with
modparam("proto_tls","require_cert", "1"), OpenSIPS misleadingly logs:
'failed to accept: rejected by client'. What it actually means is that the
client failed to present a certificate.
On 9 Feb 2016 6:06 am, "Hamid Hashmi" wrote:
It will be a great help if you please help me in configuring TLS. I have
followed this to configure TLS but could not able to verify certificates.
its working if disable following flags
modparam("proto_tls","verify_cert", "0")modparam("proto_tls","require_cert",
"0")
BUT not verifying certificate
Nabeel I have been using Linphone 3.6.1 with opensips for a long time. And its
working fine on both UDP and TCP. I have gone through your logs, there is log
line DBG:tm:matching_3261: RFC3261 transaction matching failed Please check
contact of your "to number" in location table.
Hamid R. Ha
What is the reason of the crash ? how it can be reproduced ?
Jan 31 16:17:00 prod-siplb SIPLB[24724]: WARNING:core:utimer_ticker: utimer
task already schedualed for 5216416740 ms (now 5216416840 ms), it
may overlap..
Jan 31 16:17:00 prod-siplb SIPLB[24728]: [udp:keepalive@192.168.26.237:7000]:
Try the following example. Change connection IP and codec order accordingly.
if (is_method("INVITE") && has_body("application/sdp")) {
$var(Session_owner) = $rb[1];
append_to_reply("Content-Type:
application/sdp\r\nv=0\r\n$var(Session_owner)\r\ns=call\r\nc=IN IP4
10.130.130.114\r\nt=0 0\r
Is there any opensource sip messenger for linux to test IM on SIP ?
Hamid R. HashmiSoftware Engineer - VoIPVopium A/S
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/list
Service is already listening on UDP. I just loaded TCP module with a line
listen=tcp:192.168.26.180:7000 and ended with following error. I also tried
memlog=1 but could not find the issue.WARNING:core:fm_malloc: Not enough free
memory, will attempt defragmentation
CRITICAL:core:init_io_wait: cou
I did not find any method to remove line from SDP so I am trying to replace it
with some other value but this solution is also NOT working.
$var(line1) = $(rb{sdp.line,b});
$var(line2) = $(rb{sdp.line,b,1});
replace_body("$var(line1)", "a=ptime:20");
replace_body("$var(line2)", "a=qos");
Hamid R.
rtpproxy -p /var/run/rtpproxy.pid -u rtpproxy -l x.y.z.w -s
udp:127.0.0.1:2 -F -P -n tcp:127.0.0.1:22233 -T 60 -W 120 -d DBUG
LOG_LOCAL1 -L 65535
insert in OpenSips script follow lines:
modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:2")
modparam("rtppr
uest
Start RTPproxy:
/bin/rtpproxy -p /var/run/rtpproxy.pid -u rtpproxy -l x.y.z.w -s
udp:127.0.0.1:2 -F -P -n tcp:127.0.0.1:22233 -T 60 -W 120 -d DBUG
LOG_LOCAL1 -L 65535
insert in OpenSips script follow lines:
modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.
rtpproxy_sock and rtpp_notify_socket like this
modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:2")
modparam("rtpproxy", "rtpp_notify_socket", "tcp:127.0.0.1:22233")
На 18.11.2015 г. 4:44 PM "Hamid Hashmi" написа:
According to
According to RTPproxy wiki Create/Update session request needs following
arguments.
U[opts] call_id remote_ip remote_port from_tag [to_tag] [notify_socket
notify_tag]but opensips v2.1.1 rtpproxy module not sending any socket
information in Create/Update session request.received command "29180_8
Please give me a direction to move further.
RegardsHamid R. Hashmi
From: hamid2kv...@hotmail.com
To: users@lists.opensips.org
Subject: Timeout Notification doesn't work through TCP sockets
Date: Wed, 11 Nov 2015 19:51:09 +0500
I am using opensips v2.1.1 and rtpproxy v2.0.0. RTPproxy delete the
I am using opensips v2.1.1 and rtpproxy v2.0.0. RTPproxy delete the session
after 30 seconds (-T 30) but opensips did not generate BYE locally and dialog
exists till $DLG_timeout. How to dig out the issue more in detail? or is there
any solution ?
RTPPROXY /usr/local/bin/rtpproxy -p /var/run/rt
Am i doing it right ?
mysql> select * from
dispatcher;++---+-++---++--+---+--+|
id | setid | destination | socket | state | weight | priority |
attrs | description
|++---+---
RegardsSasmita PandaNetwork Testing and Software Engineer3CLogic ,
ph:07827611765
On Mon, Oct 19, 2015 at 4:22 PM, Hamid Hashmi wrote:
SET $ru = "sip:2233@162.243.1.1:5506";
RegardsHamid R. Hashmi
Date: Mon, 19 Oct 2015 16:01:28 +0530
From: spa...@3clogic.com
To: users@lists.opensips.org
Subj
oftware
Engineer3CLogic , ph:07827611765
On Mon, Oct 19, 2015 at 3:56 PM, Hamid Hashmi wrote:
INVITE Request is going that IP because destination URI has been changed. Can
you Please share the URIs before and after t_relay();
RegardsHamid R. Hashmi
Date: Mon, 19 Oct 2015 15:26:39 +0530
From:
mp; RegardsSasmita
PandaNetwork Testing and Software Engineer3CLogic , ph:07827611765
On Mon, Oct 19, 2015 at 2:01 PM, Hamid Hashmi wrote:
Use t_relay(); instead on exit;
RegardsHamid R. Hashmi
From: hamid2kv...@hotmail.com
To: users@lists.opensips.org
Date: Mon, 19 Oct 2015 13:29:52 +0500
Use t_relay(); instead on exit;
RegardsHamid R. Hashmi
From: hamid2kv...@hotmail.com
To: users@lists.opensips.org
Date: Mon, 19 Oct 2015 13:29:52 +0500
Subject: Re: [OpenSIPS-Users] How to change the destination uri and forward the
call to that uri .
SET
$du = "sip:162.243.1.1:5506"
OR
$rd =
SET
$du = "sip:162.243.1.1:5506"
OR
$rd = "162.243.1.1";$rp = "5506";
RegardsHamid R. Hashmi
Date: Mon, 19 Oct 2015 13:37:46 +0530
From: spa...@3clogic.com
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] How to change the destination uri and forward the
call to that uri .
Hi All ,
ion, but do you have traffic on
that OpenSIPS ?
Can you do an "opensipsctl trap" and provide the output ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 23.09.2015 11:14, Hamid Hashmi
wro
On 23.09.2015 10:38, Hamid Hashmi
wrote:
Hello
I am upgrading opensips from v1.9 to v2.1.1. What is the
alternate of own_timer_proc in latest version v2.1.1. I am
having following wearings continuously after startin
Hello
I am upgrading opensips from v1.9 to v2.1.1. What is the alternate of
own_timer_proc in latest version v2.1.1. I am having following wearings
continuously after starting the service.
..INFO:acc:mod_init: initializing...WARNING:acc:mod_init: Integer flags are now
deprecated! Use unique qu
,
I see the UAS is breaking the contact by re-writing it with the IP
of the proxy. Is the UAS an opensips too, by chance ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 09.09.2015 14:21, Hamid Hashmi
wrote
ertheless, the PATH does not interfere and change anything when
comes to the sequential requests.
Could you post the 200 OK between UAS-proxy and proxy-UAC ?
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-so
What is the time difference between duplicate SIP messages ? It happens due to
network latency.
RegardsHamid R. Hashmi
Date: Mon, 7 Sep 2015 21:19:16 +0500
From: aqsyou...@gmail.com
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] Multiple invite and bye for a single a call
Hi, users.
I
bogus Contact header, the whole in-dialog routing
(for the ACK) gets broken.
Best Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 03.09.2015 09:29, Hamid Hashmi
wrote:
D
penSIPS-Users] ACK did not Receive on Proxy and get in loop
I beleive you have to engage NAT corrections in reply route for 200OK.
I havent seen your SIP trace but if im Comtaact header the IP is of UAS and it
goes to the caller side ACK will get directly to that IP.
Just my guess. Need more
ACK is not receiving on Proxy, It directly goes to UAS and get in loop at Proxy
before relayed to bob. Can you explain why ACK is going directly to UAS while
all other responses going to proxy?
AliceProxy UAS Bob |
||
er and Developer
http://www.opensips-solutions.com
On 26.08.2015 18:06, Hamid Hashmi
wrote:
I am unable to use
'$RAD_REQUEST{'Digest-URI'}, $RAD_REQUEST{'Digest-Realm'}
etc, in my perl script in case
of
I am unable to use '$RAD_REQUEST{'Digest-URI'}, $RAD_REQUEST{'Digest-Realm'}
etc, in my perl script in case of service type equals to 'Call-Check else cases
are working fine like 'Sip-Sessions' etc. details are as follows.
$ opensips -Vversion: opensips 1.9.0-notls (x86_64/linux)flags: STATS: Of
On 13.05.2015 09:49, Hamid Hashmi
wrote:
Previously, I have tried `make clean` but now i
have also tried 'make proper' and 'make'. But faced the same
error.
root@Debian-Virtu
ounder and Developer
http://www.opensips-solutions.com
On 12.05.2015 13:32, Hamid Hashmi
wrote:
I have successfully installed v1.11.5 but when i
compile v2.1.0 on the same machine, have the following ERROR
I have successfully installed v1.11.5 but when i compile v2.1.0 on the same
machine, have the following ERROR
Compiling lex.yy.cCompiling cfg.tab.cLinking opensipsmem/q_malloc.o: In
function `qm_mem_check':/usr/local/src/opensips-2.1.0/mem/q_malloc.c:727:
undefined reference to `qm_debug_frag'c
RTPproxy replaces 'o' and 'c' in SDP with private IP in case of
rtpproxy_offer("OCNFWIE") while in case of rtpproxy_offer("OCNFWEI") it gives
an error
ERROR:rtpproxy:unforce_rtp_proxy_f: no available proxies.
UAC ---> Amazon Firewall -> Opensips/RTPproxy
How to remove dialog from table "dialog" ? OR is there any method to update the
timeout value in dialog without calling create_dialog("B") ?
RegardsHamid R. Hashmi
From: hamid2kv...@hotmail.com
To: users@lists.opensips.org
Date: Tue, 10 Mar 2015 17:43:15 +0500
Subject: [OpenSIPS-Users] How to rem
route[sip]{...t_on_failure("1");$DLG_timeout = 120;
create_dialog("B"); t_relay();}
failure_route[1]{... if(t_check_status("some reasson")){
route(pstn); }...}
route[pstn]{...t_on_failure("2");$DLG_timeout = 60;
create_dialog("B");t_relay();}
How
), there will be nothing sent back
to UAC (caller) to inform about this timeout event.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 10.02.2015 10:17, Hamid Hashmi wrote:
Hello
Hello,
On fr_ivr_timer timeout opensips1.11 sends CANCEL to Caller and I handle the
timout in failure route by 408 sip response. May I change method CANCEL to some
other response that can indicate a caller (UAC) that call is now diverting to
another gateway so that Caller can wait more.
failu
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