132 questions
0
votes
0
answers
31
views
Error in OpenSIPS When Modifying Route Header to Remove Commas in BYE Method
I am working with OpenSIPS 3.1 and trying to modify the Route header in SIP messages for the BYE method. My goal is to remove all commas present in the Route header if the BYE message originates from ...
1
vote
1
answer
93
views
How to configure modules to use postgresql
I'm struggling to find any information on how to set up OpenSIPS with postgresql for modules like B2B, dialog, etc.
Example for the b2b_entities module I configure it as follows:
loadmodule "...
- 349
0
votes
0
answers
136
views
first opensips a simple call- receive a call from a sip client and send it to a sip server-
I have fresh installed opensips and lets say it is running on 192.168.10.50,
what i want is to receive a call from a sip client from an IP 1.1.1.1 and send it to a sip server10.10.10.50
i am receiving ...
1
vote
1
answer
146
views
Opensips get current value of Header [closed]
I'm new to opensips,
I have sip Flow like this:
A send Invite to Opensips Server X
X send Invite to devices IpPhone connect to extension 102
Devices Iphone still not pickup call yet, it transfer to ...
- 327
0
votes
1
answer
74
views
How to configure in build keepalived of opensips?
I am trying to configure opensips keepalived for nodes handling in cluster. I followed the instrustions mentioned in this guide. https://controlpanel.opensips.org/htmldoc_9_X_X/keepalived.html. After ...
- 1
0
votes
0
answers
124
views
Error connecting from React Native app to SIP server using react-native-jssip: "Handshake failed"
what i get when try to connect
@xml/network_security_config file.
I am trying to establish a connection from my React Native app to a SIP server using the react-native-jssip library (or any other ...
- 1
1
vote
1
answer
184
views
Opensips - Bad From Header when I was using failover in my script
my REQUEST_ROUTE:
...
route [FROM_GATEWAY] {
if ($fU=~"^999" && !$avp(new_caller)) {
for ($var(v) in $(json(foo)[*]))
$avp(new_caller) = $var(v);
}
...
- 23
2
votes
0
answers
233
views
Failed to connect to Opensips remote server from Zoiper
I have installed opensips on my google cloud VM instance, I have opensips control panel installed.
I am able to access control panel and create two users alice and bob . But the problem is I am unable ...
- 453
0
votes
1
answer
950
views
Opensips-cli -x command not working in opensips 3.3
Recently I am working on upgrading my opensips version manually from 2.2 to 3.3.
Upgradation is done from my side but in old opensips(2.2) I was able to show registered user(SIP) using opensipsctl ul ...
- 852
0
votes
1
answer
562
views
ACK message not routes when running opensips as mid_resistrar for freeswitch
This is my first time using opensips, so am probably missing something obvious.
I am trying to use opensips and mid_registrar in front of freeswitch. The opensips is behind a NAT so I am setting the ...
- 57
0
votes
2
answers
909
views
[RTPEngine]Failed to init DTLS connection: key values mismatch
i got the connection like this
sip flow
freeswitch server --(sip)-- > opensips ----(wss)---> sip client in chrome with jssip/webrtc
the rtp flow
freeswitch server ---- > rtpengine----...
- 21
0
votes
0
answers
255
views
webrtc sip calling video web application
do any one can built a webrtc sip calling video chat application with explanation of each step to compile the web application as end to end complete demo ?
- 47
1
vote
0
answers
353
views
RTPEngine transcode Opus to PCMU - unable to change Opus settings from defaults
I am invoking RTPEngine from OpenSIPS to transcode incoming Opus call to PCMU. I`m passing these parameters to the RTPEngine daemon:
codec-mask-opus codec-set-opus/48000/2/32000//maxplaybackrate=32000;...
- 129
1
vote
1
answer
385
views
OpenSIPS, how to avoid duplicate invite when registering through push notifications?
This is my script. I found that for a register, E_UL_CONTACT_INSERT will be triggered one or more times. I don’t want to invite the same client multiple times.
route[route_to_user] {
...
- 55
0
votes
1
answer
1k
views
kamailio: how to extract IP from received route header
Could any help me to find way to extract IP & Port from route header received (which is on the SIP INVITE received from a remote server).
KAMAILO ----> A Server
A - Server is appending router ...
- 234
0
votes
1
answer
2k
views
kamailio: how to send replies with port number in VIA header
using kamailio version 5.4v
Please help me with a way to send replies/forwards to include VIA header with port, because i can see that its just sending IP but not the port in VIA header.
my kamailio ...
- 234
1
vote
1
answer
278
views
Opensips - variables and strings
Dears,
I use Opensips 2.4
How to use mid_registrar_save("$(rd{ip.resolve})") to mean variable $(rd{ip.resolve}) instead of string "$(rd{ip.resolve})"
Thank you for your help.
User ...
- 23
1
vote
0
answers
228
views
SIP auth with ejabberd
I am trying to relay a SIP SUBSCRIBE message from OpenSIPS to my ejabberd server. However ejabberd insists on challenging this with a 407 Proxy Authentication Required response.
I would like to ...
- 11
0
votes
1
answer
161
views
Sending Requests to Locally-hosted OpenSips Server using android
I am really new to opensips and lately I was able to install the OpenSip server on my VirtualBox based VM ( Debian 10). Now I want to create an android application which enables SIP calling (Push to ...
1
vote
0
answers
116
views
PSTN Call is not working over opensip instalaltion
we have already installed opensip over asterisk but i want to move PSTN calls over asterisk .
Opensip working veru fine in extension to extension call . But i want The Public Switched Telephone ...
- 11
1
vote
1
answer
426
views
Not able to access OpenSIPS CP web interface
I have installed opensips-2.1.2 in on vmware (ubuntu 14.4.6).
I can see opensips installation went fine, but not able to access the web interface for the same.
After installation i have greped the ...
- 71
0
votes
1
answer
1k
views
VoIP delta spikes below 20ms, causing the jitter to change
I am trying to do some measurements on VoIP. I am using OpenSIPS, RTPProxy, and SIPp for testing.
Everything works fine as expected, but I only have a question regarding the delta time.
Below is a ...
- 35
1
vote
1
answer
2k
views
Opensips 2.4 can't make outbound call
I followed tutorial on https://www.powerpbx.org/content/opensips-v24-debian-v8-mariadb-apache-v1
With minor tweaks for "Debian Buster" I have installed all components, services starts with ...
2
votes
1
answer
2k
views
install and configure opensips as load-balancer for freeswitch [closed]
I am trying to configure opensips as load-balancer for freeswitch by following below link but the procedure written there seems very old and many issues are faced while following the steps mentioned ...
- 21
0
votes
1
answer
620
views
Is there a way to retrieve the sessions allocated by rtpproxy?
I have an OpenSIPS proxy running with RTPProxy. The calls are working fine and the media is relayed to RTPProxy. Is there a way to figure out what are the allocations used by the server? Can I query ...
- 35
1
vote
1
answer
363
views
OpenSIPs Control Panel errors
I have setup OpenSIPs control panel and I can successfully complete basic functions like adding users. Problem is I keep getting this error when clicking on most features in the control panel.
MI ...
- 2,087
0
votes
2
answers
434
views
OpenSIPS fork=yes required when listening on multiple ports?
If I set OpenSIPS to listen on multiple ports on the same IP, do I need to set fork=yes or is that an old option no longer needed? Thanks
- 2,087
1
vote
1
answer
1k
views
PSTN to OpenSIPS to next SIP destination
I have worked with Asterisk for years but I am very new to OpenSIPS. What I need is to have calls come in from our DID provider to the OpenSIPS server then redirect them to another SIP URI.
Something ...
- 2,087
1
vote
1
answer
243
views
Speed up Mysql queries with Hash Table in OPENSIPS
Dears,
I have got an Opensips server which make queries to an mysql server. I need to optimice these queries at maximum.
One way could be: using mysql table with engine=memory and index=hash, but ¿...
- 11
0
votes
1
answer
675
views
Can't create database 'opensips'; database exists
When trying to create a database for OpenSIPS I got this error:
root@root:/etc/opensips# opensipsdbctl create
MySQL password for root:
INFO: test server charset
INFO: creating database opensips ...
...
1
vote
1
answer
290
views
script variable should not be used for call processing?
sir,
I am trying to create stateful proxy in opensips 2.4.
I just wanted a variable to hold received message information and process it.
So i checked "core variable" in opensips manual.it says, ...
0
votes
0
answers
320
views
opnsips start gives " ERROR:core:main_loop: failed to fork module processes" and exits
starting opensips in aws t2 micro ( t2.micro Variable ECUs, 1 vCPUs, 2.5 GHz, Intel Xeon Family, 1 GiB memory, EBS only) in ubuntu gives " ERROR:core:main_loop: failed to fork module processes" and ...
- 1,363
0
votes
1
answer
243
views
Opensips suddenly crash in two-three days running
I am using opensips, it is working fine but after 2-3 days it suddenly crash. Don't understand following log
CRITICAL:core:receive_fd: EOF on 17
INFO:core:handle_sigs: child process 14090 exited by a ...
- 2,399
-1
votes
1
answer
843
views
Asterisk 13 PJSIP sometime sound coming sometime not coming
I recently set up my asterisk 13 with PJSIP and database. All working fine, but sometimes I get no voice, where most of the time I get a voice. So I need RTP software? following is detail log, I am ...
- 2,399
1
vote
1
answer
414
views
compiling Opensips on Clion: missing tap.h, which library is missing?
I am trying to compile Opensips on Clion as I want to debug and step through the code. I have tried adding environment variables through following lines in CMakeLists.txt:
set(CMAKE_C_FLAGS "-funroll-...
1
vote
1
answer
1k
views
OpenSIPS 2.4 call forbidden
I discovered OpenSIPS and all the possibilities a few days ago. I would just use it as a simple SIP proxy to get started. Proxy between my designated UAC and my UAS (asterisk, not natted). The goal is ...
- 176
2
votes
2
answers
1k
views
ERROR:mi_fifo:mi_fifo_check: security: fifo_check: inode/dev number differ: (/tmp/opensips_fifo)
I am new to opensips and have installed it a few days ago. I have got it to make calls. But i am facing a problem with mi_fifo module. It is giving the following error
ERROR:mi_fifo:mi_create_fifo: ...
1
vote
1
answer
360
views
Opensips avp_db_query can't compare null value
I am using avp_db_query to retrieve my table row, sometimes one field value is null. But when I use if condition it does not follow and move on.
avp_db_query("select status from orders where id = 1", ...
- 2,399
2
votes
1
answer
3k
views
Sip UPDATE method
I am new to sip protocol.I understand the normal sip mechanism like how it works.I know about sip re-invite method which is useful to update the SDP(Session Description Protocol) parameters.But ...
- 39
1
vote
1
answer
290
views
Saving cdrs manually using avp_db_query in Opensips
Is there a way to record cdrs manually using avp_db_query in opensips. I am using ACC table to record cdrs and than running procedure to transfer data to another table. But this put a lot of overhead ...
- 2,399
1
vote
0
answers
178
views
OpenSIPS Azure signalling does not forward call
I try to migrate an OpenSIPS implementation 2.4.1 from our regular datacenter to Azure.
I use standard residential script with NAT enabled, registrant and dynamic routing module. Calls come in fine ...
- 21
1
vote
1
answer
99
views
Passing AVP to prefix core function
I am working what appears to be a simple function for opensips 2.2.3, however cannot seem to get it working..
Essentially, extract the groupID from permissions module and add a prefix to R-URI on ...
- 15
1
vote
1
answer
610
views
How can do accounting manually in opensips
I am using Opensips 2.3 and already doing accounting. But I have a very different database, where I already configure to do missing/channel exceed CDRS manually using avp_db.
Is there a way to do ...
- 2,399
0
votes
0
answers
795
views
SSL_error_SSL error on tls_read
In a production setup, randomly a opensips error comes up indicating tls_read failed due to SSL_error_SSL error.
Opensips fails the tls/tcp session and a new session is created and it works fine.
...
- 67
0
votes
1
answer
818
views
SIP Redirect Server
I'm trying to create a SIP redirect server for LCR routing purpose, I have a basic question here:
is opensips is the best solution, and is there other free easier solutions?
Any info on this will be ...
- 61
0
votes
1
answer
512
views
Opensips & Freeswitch IP LAN & WAN Config
I have setup a dedicated Opensips Server on a public ip xx.xx.xx.xx & Dedicated Freeswitch Server 192.168.1.2, a dedicated MySql Database Server 192.168.1.3. My router is on 192.168.1.1. I have ...
1
vote
1
answer
422
views
How to convert string in URI
I setup OpenSips 2.3 proxy server, so any call come on server, my script grabs sip URI from DB, and forward call to that uri. When I get value I used AVP to get value and save it in $avp(didnumber), ...
- 2,399
0
votes
1
answer
124
views
Json in Perl error in opensips
What is the difference between json and json::PP in Perl?
I meet this error when use Json and Json:PP when writing perl script in opensips
ERROR:core:XS_OpenSIPS__Message_log:
perl warning: ...
- 1
2
votes
0
answers
331
views
Send stop recording request to RTPEngine from outside current session
I have an Opensip / RTPEngine setup where Opensips sends a start recording request to rtpengine when callers are connected. But we have an unfortunet situation where one of the callers can disapear ...
- 595
1
vote
1
answer
325
views
openSIPS setup an onreply route if the call is picked up
I'm wondering if is it possible to set a condition for call answered/picked up in an onreply_route
something like this
onreply_route {
if(call picked up) {
xlog("ON AIR");
}
}
- 595