thank you! I'll give that a shot!
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Ok, so hopefully I get some help in this forum, I'm trying to get a faster
PDD by going down the LCR list faster. I'm trying to use the fr_inv_timeout
to skip the carrier that has slow PDD. So I did the code below so that if I
don't get a 180/183 after invite within 5 second, I will terminate the
Just curious, I am using opensips 1.11 and have the tm module set as:
modparam("tm", "fr_timer", 5)
modparam("tm", "fr_inv_timer", 120)
modparam("tm", "restart_fr_on_each_reply", 0)
modparam("tm", "onreply_avp_mode", 1)
When I do wireshark on a call that the far end device is not responding (or
d
Ok, thx! Let me try that!
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U
So I guess I am trying to get that going again. I am currently running 1.11
opensips and here is what my code:
loadmodule "tm.so"
modparam("tm", "fr_timer", 10)
modparam("tm", "fr_inv_timer", 10)
modparam("tm", "restart_fr_on_each_reply", 0)
modparam("tm", "onreply_avp_mode", 1)
...
route{
...
thank you!
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Is there a way to force a re-register time on a particular user (ua)? I
don't want to do it to all, just a particular (or specified) user. I am
running 1.11.3. thx!
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thx for the reply, I think I fixed the issue, but do you know why am I seeing
so many of these error messages? Is it just means some of the SIP message
opensips gets can not be parsed or have some weird stuff in it? I checked
these IP, they are from VoIP devices manufactured by Grandstream (e.g.
I am currently running Opensips 1.9 on Debian Whizzy, but as stated on my
other thread, I am getting some weird error which opensips would suddenly
stop working.
So, I am starting a new installation of Opensips 1.11 on Centos 6.6 using
the yum install method:
1. I was able to install yum opensi
Thx for the reply, but what is the function of rtpproxy? I use asterisk to
handle my rtp (media). So can I remove the module from opensip being
loaded? Also network congestion, but is on the localhost which is itself,
how can there be network congestion?
Thx.
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I don't know how to describe the issue. But we've been running on 1.9 for
years without any issue. Lately, it start acting up "weird". My setup is
that I am using opensips as the signalling gateway and behind it, the
asterisk as media gateway. So lately, asterisk would suddenly deem opensips
to
I have 2 opensips setup as HA. The opensips have 2 interface, one public and
one private. The public is the "floating IP" between the 2 HA pair opensip.
I've learn to use the following python call to enable or disable a node on
load balancer remotely:
opensips = xmlrpclib.ServerProxy('http://19
Nevermind, I finally get it. So is NOT PHP, I know I may sound stupid, but
to linux rookie, is very hard.
So just for future reference, or anybody would like to know:
1. Install make sure you have python install in your OS
2. Make a file with extension of .py (e.g. the following):
#!/usr/bin/py
Ok, so I figured that you can use xmlrpc to call fifo function remotely, but
the issue is I am no linux expert, so I've been reading things here and
there.
I guess the thing I need to do is to write a xxx.php file on Asterisk and
then use the php -h xxx.php to execute it to send a command to ope
Is there a place I can download the mi_xmlrpc.so? I looked under my
/usr/local/lib64/opensips/modules directory, the file is not there.
Thx!
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I guess further searching I need to install xmlrpc module, but I read on doc
that it needs 0.9.10 but I tried to do apt-get install libxmlrpc-c3=0.9.10
it says it can not find that version. If I simply install, it will install
version 1.16, will that be ok?
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Is there a way to execute lb_status remotely to disable a node from load
balancer?
So I have an inbound Opensips that takes all the calls and load balance to
multiple asterisk. And then I have another obound opensips that will
terminate call from the asterisk server. The call flow goes:
Inbou
I've asked this issue before and trying to do some search, but I am still not
able to accomplish what I need in my opensips script.
So, on the "outbound" opensips where I send call to the carriers, the sip
flow would be
1. INVITE (from opensips server)
2. 100 TRYING (from far end carrier)
3. 18x
Ok! Thank you very much, so currently I have:
listen=udp:99.99.99.1:5060
where my NIC card is actually 99.99.99.1
So now I can change my NIC card to 192.168.1.100 and then put in opensip
config file as:
listen=udp:192.168.1.100:5060 as 99.99.99.1:5060
And that I simply do 1 to 1 nat on the fi
I am currently running Opensips using public IP (on NIC and also in config
file). So is there a "quick" way I can convert that to "private" IP? e.g.
maybe some parameter in the global section of the config file that I can do
so that Opensip will use a "public" IP in all is SIP messages? This way
This may be stupid question, but I have this code in my opensips script to
match number
if (uri=~"^sip:11211[0-9]*@")
(just side note 11121 or 11101 is internal routing code)
So how do I match on "+"? What I mean is I want opensips to say:
if 1112112135551212 do "this"
if 111011+12135551212
thank you, yes I finally found that out!
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I have Asterisk as media gateway and Opensip in front. So the issue I am
running into is that, in reference to Asterisk, it send calls to Opensips,
it is not aware of the carrier which Opensips is sending to.
Thus:
1. "INVITE" header would be the IP address of the Opensips
2. "From" header wou
Oh yes, I figured out, I just forgot to remove the thread. It seems the
database "data type" matters in this case. If that field in the DB table is
varchar (I am using MSSQL) avp variable under load balancer module will not
work. You have to use one of the "interger" data type in order for this
Actually after reading, so like you said $T_fr_inv_timeout is just a
"dynamic" way to change the fr_inv_timeout setting for the tm module. BUT
it seems by reading $T_fr_inv 'definition':
1.5.4. $T_fr_inv_timeout
$T_fr_inv_timeout (R/W) - the timeout for the final reply to an INVITE
request, afte
Is it possible to use avp variable instead? I've tried, it seems not
working, I have to put some numeric value for it to work. Or is my syntax
wrong somewhere?
load_balance("$avp(GroupID)","PSTN")
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thank you very much for the information! I'll try it out!
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Ok, so I tried this timer thinking that it would shorten my PDD out to
multiple carriers. But I guess this timer control when the call "answer"?
So anything after INVITE (e.g. 100 trying, 180 ringing or 183) it will
ignore? So if the timer set to 5 seconds, it will try that SIP route for 5
secon
Ya, that is what I thought, but I was wondering if there is a way to just
have the opensips validate on first try because first registration SIP
already have the username and password. The reason I ask is that it seems
something it will take more then 2 try for opensips to reply a 200OK. So
here
Why is it that when I use this function call on registration and opensips
will always reject the first registration request and then on the second one
it will sent back 200OK?
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Sen
So anyhow, I probabaly better off to go with "IF" statement then "switch" in
this type of scenario,huh...
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Really?! So the "exit" statement I put does not EXIT from the entire script?
Then what does "exit" do? Hey by the way, thanks again for taking the time
answering my question.
And if it go next block, the return code should been "-1" why it would
execute in code block in "-3" or "-2"?
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On top of that, I get one way voice... need to wireshark more...
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Hello Liviu,
thanks for your help again! Ok, so I tried that and is weird, is kinda
working but is working weirdly. Not sure, maybe I don't truely understand
how opensips works? But here is my code:
if (is_method("INVITE")) {
trace_dialog();
Hello Liviu,
thanks for the help. Although one more question, on your "sip:FRANK101@ip";
What is the "@ip"? is that the IP of the "domain" field in the subscriber
table, or the IP in the "contact" field of the location table?
here is an example of what in my "contact" field of the "location" t
sorry I type wrong the syntax for location, please ignore the "d", I got it
confused with the db_alias lookup command. I have in my code only
lookup("location","","FRANK101")
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Hello Bogdan,
I tried to use lookup("location","d","FRANK101") and it seems opensips can
not find the user, even though the username "FRANK101" is in the subcriber
table.
So it is my syntax is wrong on the "lookup" function call?
And please correct me if I am wrong also, I was under the impressi
Thank you! I'll give that a shot!
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Or will location("username") function do it?
So what I mean is that if I do
If(is_method("INVITE")) {
lookup("username");
t_relay();
}
So that if
1. asterisk server sent an INVITE to OpenSIP (with an username of unknown by
OpenSIPs)
2. Opensips goes into the code above
3. rewrite the INVIT
How do I get the IP and Port of registered user (probably a stupid question)?
I check the usrloc module, I don't see function like that.
The reason I need this is that when we were only using Asterisk server, we
were able to sent INVITE out with the DNIS. So what I mean is that I can
dial out
Have you use sql server 2008 with usrloc module? it seems with that I can
only do db_mode 3, db_mode 2 will cause it fail. Pleaes let me know if you
are able to do db_mode 2 with usrloc module with sql server 2008 backend.
thank you1
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no the problem is not create table command. I simply show the sql create
table command so that you guys can see what is my table (data type)
definition is. I do think there is some issue with the table definition
where sql server 2008 has different then mysql, but I thought some of you
know this
As I stated in my reply to dave, yes I have that setup, I have all module
running on sql server 2008 r2 via freetds, the only module does not load
correctly is the dialog module. That is why I am wondering what is wrong.
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I am using the freedts. It is working fine. Actually I have all of my
opensips module running using sql server 2008 r2 as backend right now (e.g.
avpops, sip trace, auth_db, and many others). the only module not playing
along is the dialog module. It only likes mysql but not sql server 2008.
Anybody tried to use the dialog module with sql server 2008 r2 back end?
Opensips will start ok if I use mysql as db for dialog module, but as soon
as I switch to sql server it give me this error in the log
Nov 7 19:59:50 OSIPIBD-2 /usr/local/sbin/opensips[16842]:
INFO:dialog:mod_init: Dialog mo
Yes, I know. But with that, I will end up with 1 HUGE context (script) in
the extensions.conf. So if I have 10 different customer, I would have to
have 1 HUGE context script to maintain. Vs. if I can direct differnet
Inbound calls to different context from beginning then the script is much
manag
sorry, again, I am not looking for to know the source ip "IN" asterisk
script. I need to "direct" the incoming call to "different" context within
asterisk by the source IP. I DO NOT want to have 1 big context script in
the extensions.conf. I want to be able to able to direct inbound call to
diff
thx for the suggestion, I don't think asterisk reads the IP from any of the
header or in any part of SIP message. I think asterisk read the IP from the
IP at the network layer. anyhow, if you like you can read my reply to Mike.
Thx again. Have you ever encounter the usr_loc module that stop upd
thx for the help, if you like you can read my reply to Mike.
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thx for the suggestion, but anything "after the fact" (what I mean is once
gets into asterisk script or context), is no good.
What I am trying to accomplish is to have asterisk pick the correct context
base on the sippeers table. The only way asterisk will identify by is the
user name in the UR
Just FYI, I tried, I insert your line in the method invite and right before
the routing, Asterisk didn't seem to care. It still care about the prior
Hop IP.
So what I mean is that
from 199.33.33.33 --> opensip 22.55.33.33 (and then I put your line) -->
Asterisk server.
Asterisk server identifie
cool, thx for that, I will try it! Thank you very much for your help!
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Are you sure? Can you tell my which function call in opensips? I know how
to manipulate the core variable, but $si is read only. And I think if you
define a "peering" resource in asterisk, it will try to match it by the
source IP at the network layer and not within the INVITE. Please tell me
wh
I've try to search on internet but not much info. I currently have Asterisk
server setup to have sip trunk with customers on a "peer" type. This way,
no registration need and that asterisk server will identify the inbound call
base on "IP address" matching. But now I would like to put OPENSIPS i
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