Re: [OpenSIPS-Users] Too many Hops error

[Francjos]

Hello, 

Yes I configure up to 100 sessions per server, but i still have the same
error on the second server.
I'm gonna explain the scenario once again, what i'm trying to do is to put
in place redundancy with Opensips servers and do load balancing.

I'm using sipsak to send OPTIONS to Opensips servers and load_balancer
module to do load balancing.
And i think that the error "service fuul" occurs when OPTIONS are sent,
because Opensips is supposed to sent OPTIONS to FeeSwitch servers only.

So how can i tell Opensips server to sent OPTIONS to the other Opensips
server without generating any error?

Thanks.





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Re: [OpenSIPS-Users] Too many Hops error

[Francjos]

Hello, 

I really failed to use lb_is_destination (.) function.
In fact, when i do this:

if(!load_balance("1", "calls", "1")){
sl_send_reply("500", "Service full");
exit;
}
without any other condiiton before load balancing, it success on the first
server, but it failed on the second server saying "Service Full". I think
this error is caused by the sipsak i'm using to send OPTIONS.

I dont understand why it is ok on one server and fails on the other?




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Re: [OpenSIPS-Users] Too many Hops error

[Francjos]

Thank you Johan,

In fact, i have this scenario:

1) i have to do load balancing 
2) i have to verify opensips servers using sipsak. sipsak is used to moniter
and verify the state of opensips servers.

So, how can i perform the two actions without generating errors?



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[OpenSIPS-Users] Too many Hops error

[Francjos]

I've configured two opensips to load balance incoming calls to freeswitch
servers.One of the two opensips works welle and routes calls to the
destination, but another causes the error "Too many hops".
I have the same configuration file for the two opsnisps.

Any help?

Here is the configuration file:

opensips.cfg


### Global Parameters #

#debug=3
debug=4
log_stderror=yes
log_facility=LOG_LOCAL0

fork=yes
children=4

/* uncomment the following lines to enable debugging */
#debug=6
#fork=no
#log_stderror=yes

/* uncomment the next line to enable the auto temporary blacklisting of
   not available destinations (default disabled) */
#disable_dns_blacklist=no

/* uncomment the next line to enable IPv6 lookup after IPv4 dns
   lookup failures (default disabled) */
#dns_try_ipv6=yes

/* comment the next line to enable the auto discovery of local aliases
   based on revers DNS on IPs */
auto_aliases=no
mhomed=yes


#listen=udp:127.0.0.1:5060   # CUSTOMIZE ME
listen=udp:10.0.0.5:5060


### Modules Section 


#set module path
mpath="/usr/lib/opensips/modules/"

# DB_MYSQL module
loadmodule "db_mysql.so"
modparam("db_mysql", "exec_query_threshold", 60)
modparam("db_mysql", "timeout_interval", 2)

#  UAC_AUTH module: Authentication module
loadmodule "uac_auth.so"
modparam("uac_auth", "credential", "username:10.0.0.5:password")

## UAC_REGISTRANT module: module of registration on a remote server
loadmodule "uac_registrant.so"
modparam("uac_registrant", "hash_size", 2)
modparam("uac_registrant", "timer_interval", 100)
modparam("uac_registrant", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips")
modparam("uac_registrant", "table_name", "registrant")
modparam("uac_registrant", "registrar_column", "registrar")
modparam("uac_registrant", "proxy_column", "proxy")
modparam("uac_registrant", "aor_column", "aor")
modparam("uac_registrant", "third_party_registrant_column",
"third_party_registrant")
modparam("uac_registrant", "username_column", "username")
modparam("uac_registrant", "password_column", "password")
modparam("uac_registrant", "binding_URI_column", "binding_URI")
modparam("uac_registrant", "binding_params_column", "binding_params")
modparam("uac_registrant", "expiry_column", "expiry")
modparam("uac_registrant", "forced_socket_column", "forced_socket")

 SIGNALING module
loadmodule "signaling.so"

 StateLess module
loadmodule "sl.so"

 Transaction Module
loadmodule "tm.so"
modparam("tm", "fr_timeout", 5)
modparam("tm", "fr_inv_timeout", 30)
modparam("tm", "restart_fr_on_each_reply", 0)
modparam("tm", "onreply_avp_mode", 1)

 Record Route Module
loadmodule "rr.so"
/* do not append from tag to the RR (no need for this script) */
modparam("rr", "append_fromtag", 0)

 MAX ForWarD module
loadmodule "maxfwd.so"

 SIP MSG OPerationS module
loadmodule "sipmsgops.so"

 FIFO Management Interface
loadmodule "mi_fifo.so"
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
modparam("mi_fifo", "fifo_mode", 0666)


## Text Operations Module
loadmodule "textops.so"


 URI module
loadmodule "uri.so"
modparam("uri", "use_uri_table", 0)
modparam("uri", "db_url", "mysql://opensips:opensipsrw@localhost/opensips")

 USeR LOCation module
loadmodule "usrloc.so"
modparam("usrloc", "nat_bflag", "NAT")
modparam("usrloc", "db_mode",   2)
modparam("usrloc", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips")

 REGISTRAR module
loadmodule "registrar.so"
/* uncomment the next line not to allow more than 10 contacts per AOR */
#modparam("registrar", "max_contacts", 10)
modparam("registrar", "tcp_persistent_flag", 7)

 ACCounting module
loadmodule "acc.so"
/* what special events should be accounted ? */
modparam("acc", "early_media", 0)
modparam("acc", "report_cancels", 0)
/* by default we do not adjust the direct of the sequential requests.
   if you enable this parameter, be sure the enable "append_fromtag"
   in "rr" module */
modparam("acc", "detect_direction", 0)
modparam("acc", "failed_transaction_flag", "ACC_FAILED")
/* account triggers (flags) */
modparam("acc", "log_flag", "ACC_DO")
modparam("acc", "log_missed_flag", "ACC_MISSED")


#AUTHentication modules
loadmodule "auth.so"
loadmodule "auth_db.so"
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("auth_db",
"db_url","mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
modparam("auth_db", "load_credentials", "")


### Doamian module
loadmodule "domain.so"
modparam("domain",
"db_url","mysql://opensips:opensipsrw@localhost/opensips")
modparam("domain", "db_mode", 1)   # Use caching
modparam("domain", "domain_table", "domain")
modparam("domain", "domain_col", "domain_name")
modparam("domain", "attrs_col", "attributes")

modparam("auth_db|usrloc|uri", "use_domain", 1)

 DIALOG module
loadmodule "dialog.so"
modparam("dialog", "dlg_match_mode", 1)
modparam("dialog", "default_timeout", 21600)  # 6 

[OpenSIPS-Users] sipsak and sip_trace module

[Francjos]

hello ,

I wish 

I'm using two opensips servers in redundancy mode, i would like one  be able
to check if the other is still aliveand vice versa. I use a sipsak to
perform this.
But when i execute the commande  "sipsak -s sip:opensips@ where xxx in
the ip address of the other server, i get the following:


Server 10.0.0.5 gives:
.


U 2016/05/10 13:57:12.402720 10.0.0.5:47413 -> 10.0.0.4:5060
OPTIONS sip:opensips@10.0.0.4 SIP/2.0.
Via: SIP/2.0/UDP 10.0.0.5:47413;branch=z9hG4bK.3fc50b66;rport;alias.
From: sip:sipsak@10.0.0.5:47413;tag=4c5c78b9.
To: sip:opensips@10.0.0.4.
Call-ID: 1281128633@10.0.0.5.
CSeq: 1 OPTIONS.
Contact: sip:sipsak@10.0.0.5:47413.
Content-Length: 0.
Max-Forwards: 70.
User-Agent: sipsak 0.9.6.
Accept: text/plain.
.


U 2016/05/10 13:57:12.405823 10.0.0.4:5060 -> 10.0.0.5:47413
SIP/2.0 500 Service full.
Via: SIP/2.0/UDP
10.0.0.5:47413;received=10.0.0.5;branch=z9hG4bK.3fc50b66;rport=47413;alias.
From: sip:sipsak@10.0.0.5:47413;tag=4c5c78b9.
To: sip:opensips@10.0.0.4;tag=61890dad1e908c702027bf054a266115.c3bd.
Call-ID: 1281128633@10.0.0.5.
CSeq: 1 OPTIONS.
Server: OpenSIPS (2.1.2 (x86_64/linux)).
Content-Length: 0.
.




Server 10.0.0.4 gives:




root@front-1:/etc/opensips# ngrep -d eth0 -t -W byline "$1" port 5060 -q
interface: eth0 (10.0.0.0/255.255.255.0)
filter: (ip or ip6) and ( port 5060 )

U 2016/05/10 13:57:12.406208 10.0.0.5:47413 -> 10.0.0.4:5060
OPTIONS sip:opensips@10.0.0.4 SIP/2.0.
Via: SIP/2.0/UDP 10.0.0.5:47413;branch=z9hG4bK.3fc50b66;rport;alias.
From: sip:sipsak@10.0.0.5:47413;tag=4c5c78b9.
To: sip:opensips@10.0.0.4.
Call-ID: 1281128633@10.0.0.5.
CSeq: 1 OPTIONS.
Contact: sip:sipsak@10.0.0.5:47413.
Content-Length: 0.
Max-Forwards: 70.
User-Agent: sipsak 0.9.6.
Accept: text/plain.
.


U 2016/05/10 13:57:12.406735 10.0.0.4:5060 -> 10.0.0.5:47413
SIP/2.0 500 Service full.
Via: SIP/2.0/UDP
10.0.0.5:47413;received=10.0.0.5;branch=z9hG4bK.3fc50b66;rport=47413;alias.
From: sip:sipsak@10.0.0.5:47413;tag=4c5c78b9.
To: sip:opensips@10.0.0.4;tag=61890dad1e908c702027bf054a266115.c3bd.
Call-ID: 1281128633@10.0.0.5.
CSeq: 1 OPTIONS.
Server: OpenSIPS (2.1.2 (x86_64/linux)).
Content-Length: 0.



I guess the error "500 Service full" is related to a bloc in the script
where i define the routing logic for the load balancing:





## Freeswitch
route[2] {

xlog("L_NOTICE","[$pr:$fU@$si:$sp]: This is Media-Server Route Use
Load-balancer NOW!!\n");
 if (!load_balance("1","calls")) {
 sl_send_reply("500","Service full");
exit;
}
xlog("L_NOTICE","[$pr:$fU@$si:$sp]: Selected destination
Media-Server : $du\n");
}

In need more explanation how i can fix that.
Is sipsak a tool of checking the status of the servers or are there othor
tools?

My end goal is to put in place high availability with the two servers.
Ideas please!




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[OpenSIPS-Users] Bad udp checksum

[Francjos]

Hello,
I'm using Opensips as loadbalancer to freeswitch. When i dial my DID,
without invoking any script in the dialplan, that works fine. 
I've editded a script in javascript and invoke it in the dialplan. When i
dial my DID number, i have these errors:

480 Temporarily unvailable (in the log) and i see also "bad udp cksum when
Freeswitch tries to reply.

Responding to INVITE with 480 in the fs_cli 

I've asked the question on freeswitch community , but i havan' got correct
answer.
I ask my mind if it is related the configuration of Opensips!!

Any help?
Thank you




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[OpenSIPS-Users] SIPTRAce module

[Francjos]

Hello, 
I,ve conigured two opensips servers and i would like each one to send sipsak
to another and see if it is still working. 
I've installed sipsak on both Opensips. 
In order to see the request and reply messages, i've used the siptrace
module. 
I've loaded it on both Opensips and set the parameters as follows: 

 loadmodule “siptrace.so” 

modparam(“siptrace”, “db_url“,
“mysql://opensips:opensipsrw@localhost/opensips”) 
modparam(“siptrace”, “trace_flag“, “22”)# Here the trace_flag will 22. 
modparam(“siptrace”, “traced_user_avp“, “$avp(traced_user)”) 
modparam(“siptrace”, “trace_on”, 1) 

I also load the avpops module : 

loadmodule “avpops.so” 
modparam(“avpops”,”db_url”,“mysql://opensips:opensipsrw@localhost/opensips”) 
modparam(“avpops”, “avp_table”, “New_usr_preferences”) 

After that, i've  added the avp in the usr_preferences table using this
command: 

opensipsctl avp add –T usr_preferences opensips@myripaddress trace 1 1 

I also add this routing logic at the beginning of the main routing logic: 

if(avp_db_load(“$fu”,”$avp(trace)”)) { 
$avp(traced_user)=$fu; 
setflag(22); 
 sip_trace(); 
 xlog(“L_INFO”,”User $fu being traced”); 
} 

I restarted Opensips and tracing: 

opensipsctl restart 
opensipsctl fifo sip_trace on 

When i send a ping using the sipsak tool like this: 

sipsak -s sip:opensips@myotheripaddress, i hope i can visualize,  the
tracing in the sip_trace table, but when i visualise its content, the table
is empty. 

Am i simming something in order to get corrects results? 

Thanky you 



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[OpenSIPS-Users] Opensips high availability

[Francjos]

I have this scenario: 

Two Opensips are registered at ovh. When icoming calls are made, they land
in one of the two Opensips, the one registered last. When i stop the later,
the other Opensips does not answer those incoming calls immediately, i have
to restart it and re-register it at ovh. 
I understand that i have to write a script that must automate all of the
instructions. 

1) first, each Opensips must know that the other has stopped working 
2) second, after knowing that, this Opensips must restart and register to
ovh immediately in order to be able to receive those incoming calls. 

Any ideas of how the script can be written? 

Thank you. 




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Re: [OpenSIPS-Users] Opensips SIP trunk call handling

[Francjos]

Hello, 
I sent you invitation on skype,nduwayezu Joselyne on skype,  i'm waiting for
your reply.I still have problems with the incoming call throught Opensips.
I rely on your help please!!



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Re: [OpenSIPS-Users] OpenSips as an inbound loadbalancer and outbound proxy - issue with routing BYES

[Francjos]

Hello Younas,
I sent you the invitation on skype,i'm waiting for your reply. I still have
problems with the incoming call throught opensips.
Thank you for your attention.



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Re: [OpenSIPS-Users] Rely forbidden

[Francjos]

Here is the output of :ngrep -d eth0 -t -W byline "$1" port 5060 -q
Can you help me to understand the problem and how to resolve it?

^Croot@front-2:~# ngrep -d eth0 -t -W byline "$1" port 5060 -q
interface: eth0 (10.0.0.0/255.255.255.0)
filter: (ip or ip6) and ( port 5060 )

U 2016/04/15 09:59:46.445723 91.121.129.159:5060 -> 10.0.0.5:5060
INVITE sip:opensips@localhost:5060;transport=udp SIP/2.0.
Call-ID: 02023-ii-1b75f5ca-75e715...@sip3.ovh.fr.
Contact: .
Content-Type: application/sdp.
CSeq: 426731554 INVITE.
From: "003228800555"
;tag=02023-LB-1b75f5cb-77f5c7f17.
Max-Forwards: 27.
Record-Route: .
To: .
Via: SIP/2.0/UDP 91.121.129.159:5060;branch=z9hG4bK-UXTN-2667384e-065369b8.
Allow:
REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK.
User-Agent: Cirpack/v4.70 (gw_sip).
Content-Length: 315.
.
v=0.
o=cp10 146071438652 146071438652 IN IP4 10.7.16.156.
s=SIP Call.
c=IN IP4 91.121.129.144.
t=0 0.
m=audio 31298 RTP/AVP 18 0 8 101.
b=AS:21.
a=rtpmap:18 G729/8000/1.
a=fmtp:18 annexb=no.
a=rtpmap:0 PCMU/8000/1.
a=rtpmap:8 PCMA/8000/1.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:30.
a=sendrecv.


U 2016/04/15 09:59:46.446223 10.0.0.5:5060 -> 91.121.129.159:5060
SIP/2.0 500 Service full.
Call-ID: 02023-ii-1b75f5ca-75e715...@sip3.ovh.fr.
CSeq: 426731554 INVITE.
From: "003228800555"
;tag=02023-LB-1b75f5cb-77f5c7f17.
To:
;tag=c97b4d1cb1f3d0da549e06a8d482ef63.57bb.
Via: SIP/2.0/UDP 91.121.129.159:5060;branch=z9hG4bK-UXTN-2667384e-065369b8.
Server: OpenSIPS (2.1.2 (x86_64/linux)).
Content-Length: 0.
.


U 2016/04/15 09:59:46.459264 91.121.129.159:5060 -> 10.0.0.5:5060
ACK sip:opensips@localhost:5060;transport=udp SIP/2.0.
Call-ID: 02023-ii-1b75f5ca-75e715...@sip3.ovh.fr.
CSeq: 426731554 ACK.
From: "003228800555"
;tag=02023-LB-1b75f5cb-77f5c7f17.
Max-Forwards: 27.
To:
;tag=c97b4d1cb1f3d0da549e06a8d482ef63.57bb.
Via: SIP/2.0/UDP 91.121.129.159:5060;branch=z9hG4bK-UXTN-2667384e-065369b8.
Content-Length: 0.
.


U 2016/04/15 09:59:48.514025 91.121.129.159:5060 -> 10.0.0.5:5060
INVITE sip:opensips@localhost:5060;transport=udp SIP/2.0.
Call-ID: 25873-ax-1b75f716-0d45b7...@sip3.ovh.fr.
Contact: .
Content-Type: application/sdp.
CSeq: 426731844 INVITE.
From: "003228800555"
;tag=25873-BW-1b75f717-40cd751b1.
Max-Forwards: 27.
Record-Route: .
To: .
Via: SIP/2.0/UDP 91.121.129.159:5060;branch=z9hG4bK-ZRXL-26673a19-1004fd6f.
Allow:
REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK.
User-Agent: Cirpack/v4.70 (gw_sip).
Content-Length: 315.
.
v=0.
o=cp10 146071438849 146071438849 IN IP4 10.7.16.156.
s=SIP Call.
c=IN IP4 91.121.129.145.
t=0 0.
m=audio 34728 RTP/AVP 18 0 8 101.
b=AS:21.
a=rtpmap:18 G729/8000/1.
a=fmtp:18 annexb=no.
a=rtpmap:0 PCMU/8000/1.
a=rtpmap:8 PCMA/8000/1.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:30.
a=sendrecv.


U 2016/04/15 09:59:48.514288 10.0.0.5:5060 -> 91.121.129.159:5060
SIP/2.0 500 Service full.
Call-ID: 25873-ax-1b75f716-0d45b7...@sip3.ovh.fr.
CSeq: 426731844 INVITE.
From: "003228800555"
;tag=25873-BW-1b75f717-40cd751b1.
To:
;tag=c97b4d1cb1f3d0da549e06a8d482ef63.9640.
Via: SIP/2.0/UDP 91.121.129.159:5060;branch=z9hG4bK-ZRXL-26673a19-1004fd6f.
Server: OpenSIPS (2.1.2 (x86_64/linux)).
Content-Length: 0.
.


U 2016/04/15 09:59:48.527361 91.121.129.159:5060 -> 10.0.0.5:5060
ACK sip:opensips@localhost:5060;transport=udp SIP/2.0.
Call-ID: 25873-ax-1b75f716-0d45b7...@sip3.ovh.fr.
CSeq: 426731844 ACK.
From: "003228800555"
;tag=25873-BW-1b75f717-40cd751b1.
Max-Forwards: 27.
To:
;tag=c97b4d1cb1f3d0da549e06a8d482ef63.9640.
Via: SIP/2.0/UDP 91.121.129.159:5060;branch=z9hG4bK-ZRXL-26673a19-1004fd6f.
Content-Length: 0.
.


U 2016/04/15 09:59:50.821619 91.121.129.159:5060 -> 10.0.0.5:5060
INVITE sip:opensips@localhost:5060;transport=udp SIP/2.0.
Call-ID: 30481-qo-1b75f83d-7dfab0...@sip3.ovh.fr.
Contact: .
Content-Type: application/sdp.
CSeq: 426732128 INVITE.
From: "003228800555"
;tag=30481-QM-1b75f83e-2775f2177.
Max-Forwards: 27.
Record-Route: .
To: .
Via: SIP/2.0/UDP 91.121.129.159:5060;branch=z9hG4bK-BGVQ-26673bc9-1067a939.
Allow:
REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK.
User-Agent: Cirpack/v4.70 (gw_sip).
Content-Length: 445.
.
v=0.
o=cp10 146071439056 146071439056 IN IP4 10.7.1.129.
s=SIP Call.
c=IN IP4 91.121.129.144.
t=0 0.
m=audio 30962 RTP/AVP 18 4 0 8 125 

Re: [OpenSIPS-Users] Rely forbidden

[Francjos]

I add alias="10.7.1.68" in opensips.cfg but no change. And when i try to use
domain module, but it is not loaded when i restart opensips. These are the
errors:
*error during check of domain table version
*failed to initialize module domain



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Re: [OpenSIPS-Users] Rely forbidden

[Francjos]

When i analyse the output of the command sngrep, there is something i don't
understand, 
the sip to = 0972550428@10.7.1.68
I recognize the 0972550428, but don't recognize the other part. What that
mean?
Since yesterday i'm searching why the INVITE is rejected in Opensips.
Can somebody help please?



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Re: [OpenSIPS-Users] Rely forbidden

[Francjos]

Hello,

I setup a real provider and register opensips at the provider using username
and password i got from there.

Here is the call trace; anonymous@anonymous.inval is the number i used to
call my DID

root@front-2:/etc/opensips# sngrep -d eth0 -O save.pcap port 5060 and udp

 
sngrep - SIP messages flow viewer
  Current Mode: Online Dialogs: 40
  Display Filter:
  Idx  SIP From  SIP ToMsgs  Source 
   
DestinationMethod  Call State
  [ ] 1anonymous@anonymous.inval 0972550428@10.7.1.68  3
91.121.129.159:506010.0.0.5:5060  INVITE  REJECTED
  [ ] 2anonymous@anonymous.inval 0972550428@10.7.1.68  3
91.121.129.159:506010.0.0.5:5060  INVITE  REJECTED
  [ ] 3pinger@10.0.0.5:5060  10.0.0.6:5060 8
10.0.0.5:5060  10.0.0.6:5060  OPTIONS
  [ ] 4pinger@10.0.0.5:5060  10.0.0.8:5060 8
10.0.0.5:5060  10.0.0.8:5060  OPTIONS
  [ ] 5anonymous@anonymous.inval 0972550428@10.7.1.68  3
91.121.129.159:506010.0.0.5:5060  INVITE  REJECTED
  [ ] 60033972550...@sip3.ovh.fr 0033972550...@sip3.ovh.fr 6
10.0.0.5:5060  91.121.129.159:5060REGISTER
  [ ] 7anonymous@anonymous.inval 0972550428@10.7.1.68  3
91.121.129.159:506010.0.0.5:5060  INVITE  REJECTED
  [ ] 8anonymous@anonymous.inval 0972550428@10.7.1.68  3
91.121.129.159:506010.0.0.5:5060  INVITE  REJECTED
  [ ] 9pinger@10.0.0.5:5060  10.0.0.6:5060 2
10.0.0.5:5060  10.0.0.6:5060  OPTIONS
  [ ] 10   pinger@10.0.0.5:5060  10.0.0.8:5060 2
10.0.0.5:5060  10.0.0.8:5060  OPTIONS
  [ ] 11   anonymous@anonymous.inval 0972550428@10.7.1.68  3
91.121.129.159:506010.0.0.5:5060  INVITE  REJECTED
  [ ] 12   anonymous@anonymous.inval 0972550428@10.7.1.68  3
91.121.129.159:506010.0.0.5:5060  INVITE  REJECTED
  [ ] 13   anonymous@anonymous.inval 0972550428@10.7.1.68  3
91.121.129.159:506010.0.0.5:5060  INVITE  REJECTED
  [ ] 14   anonymous@anonymous.inval 0972550428@10.7.1.68  3
91.121.129.159:506010.0.0.5:5060  INVITE  REJECTED
  [ ] 15   anonymous@anonymous.inval 0972550428@10.7.1.68  3
91.121.129.159:506010.0.0.5:5060  INVITE  REJECTED
  [ ] 16   pinger@10.0.0.5:5060  10.0.0.6:5060 8
10.0.0.5:5060  10.0.0.6:5060  OPTIONS
  [ ] 17   pinger@10.0.0.5:5060  10.0.0.8:5060 8
10.0.0.5:5060  10.0.0.8:5060  OPTIONS
  [ ] 18   pinger@10.0.0.5:5060  10.0.0.6:5060 2
10.0.0.5:5060  10.0.0.6:5060  OPTIONS
  [ ] 19   pinger@10.0.0.5:5060  10.0.0.8:5060 2
10.0.0.5:5060  10.0.0.8:5060  OPTIONS
  [ ] 20   keepalive@91.121.129.159: 0033972550...@sip3.ovh.fr 2
91.121.129.159:506010.0.0.5:5060  OPTIONS
  [ ] 21   pinger@10.0.0.5:5060  10.0.0.6:5060 8
10.0.0.5:5060  10.0.0.6:5060  OPTIONS
  [ ] 22   pinger@10.0.0.5:5060  10.0.0.8:5060 8
10.0.0.5:5060  10.0.0.8:5060  OPTIONS
  [ ] 23   pinger@10.0.0.5:5060  10.0.0.6:5060 2
10.0.0.5:5060  10.0.0.6:5060  OPTIONS
  [ ] 24   pinger@10.0.0.5:5060  10.0.0.8:5060 2
10.0.0.5:5060  10.0.0.8:5060  OPTIONS
  [ ] 25   0033972550...@sip3.ovh.fr 0033972550...@sip3.ovh.fr 9
10.0.0.5:5060  91.121.129.159:5060REGISTER
  [ ] 26   pinger@10.0.0.5:5060  10.0.0.6:5060 8
10.0.0.5:5060  10.0.0.6:5060  OPTIONS
  [ ] 27   pinger@10.0.0.5:5060  10.0.0.8:5060 8
10.0.0.5:5060  10.0.0.8:5060  OPTIONS
  [ ] 28   pinger@10.0.0.5:5060  10.0.0.6:5060 2
10.0.0.5:5060  10.0.0.6:5060  OPTIONS
  [ ] 29   pinger@10.0.0.5:5060  10.0.0.8:5060 2
10.0.0.5:5060  10.0.0.8:5060  OPTIONS
  [ ] 30   keepalive@91.121.129.159: 0033972550...@sip3.ovh.fr 2
91.121.129.159:506010.0.0.5:5060  OPTIONS
  [ ] 31   pinger@10.0.0.5:5060  10.0.0.6:5060 2
10.0.0.5:5060  10.0.0.6:5060  OPTIONS
  [ ] 32   pinger@10.0.0.5:5060  10.0.0.8:5060 2
10.0.0.5:5060  10.0.0.8:5060  OPTIONS
  [ ] 33   pinger@10.0.0.5:5060  10.0.0.6:5060 8
10.0.0.5:5060  10.0.0.6:5060  OPTIONS
  [ ] 34   pinger@10.0.0.5:5060  10.0.0.8:5060 8
10.0.0.5:5060  10.0.0.8:5060  OPTIONS
  [ ] 35   pinger@10.0.0.5:5060  

[OpenSIPS-Users] Rely forbidden

[Francjos]

I've configured Opensips to register to a sip provider and acts as
loadbalancer.
When i call my number (that i got from my provider),i have the following(i
try to capture packets using tshark):
Capturing on 'eth0'
-40.640830 91.X.X.X -> 10.Y.Y.Y  SIP/SDP 1096 Requests: INVITE
sip:opensips@localhost:5060;transport=udp
- 50.641021   10.Y.Y.Y  ->  91.X.X.X   SIP/SDP 452 Status : 403 Rely
forbidden
Where "Rely forbidden" is related to this piece of code in opensips.cfg

if(!uri==myself){
send_relpy("403", "Relay forbidden");
exit;
}

Can you help me to understand this and how i can correct this error?
Thank you.



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Re: [OpenSIPS-Users] OpenSips as an inbound loadbalancer and outbound proxy - issue with routing BYES

[Francjos]

Thank you Younas
I sent an invitation on skype,  but i had two persons with same ID, one Aqs
Younas and another aqs younas. Let's me know when you get the invitation.



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Re: [OpenSIPS-Users] OpenSips as an inbound loadbalancer and outbound proxy - issue with routing BYES

[Francjos]

Sorry, i'm not going to answer your question, but, since i'm also using
Opensips as an inbound loadbalancer and outbound proxy, i would like to ask
how did you configure Opensips and freeswitch, because, me, when i try to
call my number (that i got from my privider), the number is unreachable or
busy.
can you tell me the steps you ging through in order incoming calls to be
working? 
Thank you



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Re: [OpenSIPS-Users] Opensips SIP trunk call handling

[Francjos]

Thank you for your reply



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Re: [OpenSIPS-Users] Opensips SIP trunk call handling

[Francjos]

To tell Freeswitch that calls come from Opensips proxy, do i have to create a
new external profile in sip_profiles directory or add an extension in
dialplan/public.xml or both of two?

Second question, in this file
:/usr/local/freeswitch/conf/autoload_configs/acl.conf.xml , i read that i
have to specify the CIDR, is the ip address the one of Opensips?

Thank you



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Re: [OpenSIPS-Users] Opensips SIP trunk call handling

[Francjos]

First, I'm gonna  test the simple case without load balancing , i'll do
changes after i success the simple routing.

Thanks



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Re: [OpenSIPS-Users] Opensips SIP trunk call handling

[Francjos]

Thank you for redirecting me to the tutorial. I picked up a part of what i
need,i think i can adjust it to my needs but the problem i still have is
where to include the logic of load balancing in opensips.cfg. The logic is
the following:

 if (is_method("INVITE")) {
if (!load_balance("1","pstn","1")) {
send_reply("503","Service Unavailable");
exit;
}
}
else if (is_method("REGISTER")) {
if (!ds_select_dst("1", "0")) {
send_reply("503","Service Unavailable");
exit;
}
}
else {
send_reply("405","Method Not Allowed");
exit;
}

So that Opensips can routes calls to the write Freeswitch.

Thanks again.



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Re: [OpenSIPS-Users] Opensips SIP trunk call handling

[Francjos]

Thanks .Can you please give me more details on how i have to proceed.
Thanks again



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[OpenSIPS-Users] Opensips SIP trunk call handling

[Francjos]

Hello,

I wish to configure Opensips that have to act as load balancer to freeswitch
boxes.
When i directly connect freeswitch to the trunk, incoming calls are managed
in /usr/local/freeswitch/conf/dialplan/public/ directory. What about if i
use load balancing? Where do i have to manage incoming (and out ) calls? IN
Freeswitch or in Opensips?

Thanks



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[OpenSIPS-Users] Multiples registrations

[Francjos]

hello,

Sorry i'm gonna ask a question that is not about Opensips, but Freeswitch, i
dont know if i have right to ask it. My question is this:


I'trying te register two freeswitch servers at ovh. The first is registered,
but when i try to register the second server, i have the following error: 
Invalid profile [external] 


It is possible to do multiples registrations on the same account at ovh with
freeswitch? 
If yes, how can i do this?

Let's me know. 



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[OpenSIPS-Users] Openisp and freeswitch

[Francjos]

Hello,
I've configured Opensips to register at OVH via a sip trunk. The
registration is done.
I would like to use freeswitch behind opensips server.
-How can i tell Opensips that it must route calls to freeswitch?
-How can i tell freeswitch that it handles calls that come from Opensips?

The situation is that there are no sip phones that will be connected to
opensips. So to test my senario, i'll only use calls that come  through OVH.
I dont see exactyl how i can do to achieeve this? 
Can someone explain me ho to procced?

Thank you



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Re: [OpenSIPS-Users] dialplan module

[Francjos]

I read the tutorial but it seems not to be approrpiate to my environnement.
In the enterprise where i do my training, they wish to put in palce two
Opensips servers and two Freeswitch servers. The Opensips serves must
register to their sip provider via a sip trunk (the registration is already
done). The opensips servers act as sip proxy and route sip signaling to
FreeSwitch servers, Freeswitch servers handle media.
There is no sip phones that  will be connected to the servers, when a call
comes through their sip trunk, the call is handled according to an IVR menu,
there is no person to answer the incoming call.

So i need clarification on the way i have to proced , step by step.
- For example, How and where the number (telephone line) corresponding to
the sip trunk is configured? In opensips or in freeswitch server?

Thanks a lot




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Re: [OpenSIPS-Users] dialplan module

[Francjos]

I do not install opensips from source, i've done a deposit from opensips
repository to /etc/apt/sources.list.
So i dont see how i can recompile via "make menuconfig" because there is no
Makefile.



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[OpenSIPS-Users] dialplan module

[Francjos]

Hello,

I'm using Opensips 2.1.2 but the dialplan module is not present in the
directory 
/usr/lib/opensips/modules. Because of this, the module can't be  loaded.
Is there a way to include the module?

Thanks



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Re: [OpenSIPS-Users] Opensips IVR

[Francjos]

I would like to use FreeSwithc as media server. Whiwh configurations do i
have  to put in place in both Opensips and Freeswitch?



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[OpenSIPS-Users] openisps as a proxy to freeswitch

[Francjos]

Hello again,

I've configured opensips to register to a sip provider via a sip trunk. That
works fine.
Now, i would like opensips acts as a proxy server and freeswitch as media
server. Opensips handles only sip signaling whereas freeswitch must handle
calls.
Which configurations  have i to put in place and how?
Thanks




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Re: [OpenSIPS-Users] Opensips IVR

[Francjos]

Thaks for your reply. I 'll use Freeswitch and opensips in front of
freeswitch



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[OpenSIPS-Users] Opensips IVR

[Francjos]

Hello every body,

Is it possible to implement an IVR (Interact Voice Response) with opensips?
If yes, how is it implemented?

Thanks



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Re: [OpenSIPS-Users] uac_registrant module

[Francjos]

Hello,

I solved the problem, it was a problem with privileges on the user
'opnesips'.
To resolve this, i 've connected to the data base using my credentials and
execute the following commands:
mysq > grant all privileges on opensips.* to opensips@localhost identified
by 'opensipsrw';
and  flush privileges;

After, i restart opensips and the problem was solved.

I' so happy!



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Re: [OpenSIPS-Users] uac_registrant module

[Francjos]

I defined the parameter db_url and all the parametrs but i still have errors:

ERROR:db_mysql:db_mysql_new_connection: initial connect failed
ERROR:core:db_do_init: could not add connection to the pool
ERROR:uac_registrant:init_reg_db: unable to connect to the database
ERROR:uac_registrant:mod_init: failed to initialise the DB support
ERROR:core:init_mod: failed to initialise module uac_registrant
ERROR:core:main: error while initialising modules




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Re: [OpenSIPS-Users] uac_registrant module

[Francjos]

These are the errors 

ERROR:uac_registrant:mod_init: DB URL is not defined
ERROR:core:init_mod: fail to initialize module uac_registrant
ERROR:core:main: error while initializing modules

Thanks



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[OpenSIPS-Users] uac_registrant module

[Francjos]

Hello, 
I've successed to laod and use uac_registrant module on one of my two
opensips sip servers.I can register one of the two opensips sip server.
But now, the problem i have on the other opensips, when i load
uac_registrant module, opensips does not want to start.
Is there something you can suggest?
Thank you.



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[OpenSIPS-Users] opensips database creation

[Francjos]

Hello ,

After i solve the problem on my first opensips, i'm trying to configure a
second opensips server.
I followed the same steps as for the configuration of the first server, but
now  i have a problem creating "opensips database"
When i execute the commad "opensipsdbctl create", it  just test the server
charset, , trying to create database opensips , but it does not create it,
it seems as it loops or waits for something on the command line.
Any idea?
Thanks



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[OpenSIPS-Users] Opensips registration

[Francjos]

Hello,

I'm getting Opensips to register to a remote server(my provider). I' ve
loaded uac_auth and uac_registrant modules. Every think works fine, i cane
see the content of the registrant table.
So,how can i do the registration to take place?is there a logic i have to
perform?

Thanks



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Re: [OpenSIPS-Users] Opensips registration

[Francjos]

I 've got the solution, the problem was the IP address and the port on which
opensips listerns. In place of opensips being bound to the localhost, i bind
it to my ubuntu server via his private IP address.

Thank you very much for your advises.



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Re: [OpenSIPS-Users] Opensips registration

[Francjos]

I edit the log file and see the following errors:

ERROR:tm:msg_send: send() for proto 1 failed
ERROR:tm:t_uac: attempt to send to 'sip:sip3.ovh.fr' failed
ERROR:core:proto_udp_send:sendto(sock, 0x7f85e7496cc0, 372, 0,
0x7f85e74955e8, 16): invalid argument(22)
CRITICAL:core:proto_udp_send: invalid sendtoparameters#012one posssible
reason is the server is bound to localhost and#012attempts to send to the
net

I write the errors as they appear on the scren



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Re: [OpenSIPS-Users] Opensips registration

[Francjos]

I set the timer interval at 120 and the expiry field at 600.
When i start opensips, and execute the command "opensipsctl registrant dump,
the state is "NOT_REGISTERED _STATE"; after when i re-excute the command,
the state is "REGISTERING_STATE",but after the state becomes
"REGISTERED_TIMEOUT_STATE", so the registration never takes place, i dont
understand why.
Is there something to do with the registrar module?




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Re: [OpenSIPS-Users] registrant table

[Francjos]

Tkank you very much to make me understand the difference.
So, given these parameters:

Username: 
 Password: yyy
 Authorisation username: zzz
 My telefon number : a
 Domain/registar: sip3.ovh.fr

I can fill the registrtant table's columns as follows:
registar_column : sip3.ovh.fr
proxy_column : NULL
third_party_registrant_column : NULL
username_column : 
password_column : yyy
binding_params_column: NULL

What about aor_column and binding_column ??

 Thank you aganin



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Re: [OpenSIPS-Users] registrant table

[Francjos]

Okey. And i have an other questrion:
I have connexion parameters provides by provider:
Username: 
Password: yyy
Authorisation username: zzz
My telefon number : a
Domain/registar: sip3.ovh.fr

1) when i create entry in the registrant  table, may i going tu use these
parameters to fill the registrant table's columns?

2) I create new subscriber using this command: opensipsctl add
xxx...@opensips.org 

In which order i have to perform these two points, 1) and 2)






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[OpenSIPS-Users] registrant table

[Francjos]

Hello,

I've loaded the uac_auth and the uac_registrant modules .But i dont see what
is the relationship between the registrant table and my database.
How can i do to create entry in the registrant table? is the entry  the same
as one in the database?
What is the difference between an opensips user and an subsciber?
thanks



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[OpenSIPS-Users] Two opensips servers and one sip trunk

[Francjos]

I have to configure my two opensips servers so that one and only one can
register at a provider via a sip trunk. The provider allows one registration
pre account, so the two opensips can't register at the same time.
I know that in order opensips server to register at a proviver, i must use
the uac_registrant module.

My questions are these:

Which mechanism can i use so that (and how to configure it) , if one of the
two servers fails, the other is aware of the event?

Once the other is aware ot the event, how can this last directly register to
the provider via the same and UNIQUE sip trunk?

I realy realy need help.

Thanks




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Re: [OpenSIPS-Users] opensips sip trunk

[Francjos]

I forgot to say that the standby server is going to be running all the time
in order  the mechanism thas is configured on it and that controls if the
master has fails , be able to do his job.
Secondary, the provider allows only one registration per account.



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Re: [OpenSIPS-Users] opensips sip trunk

[Francjos]

Thank you John, i'll try what you advise me,if i have problem i'll again post
my question maybe some else will help.

Thank you



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Re: [OpenSIPS-Users] opensips sip trunk

[Francjos]

Thank you a lot John. and which exported parametres i have  to choose because
there are many.
The second question is this: since i wish to use two opensips in redundant
enrironnement (master and slave), how can i do so that, if the master goes
down, the slave registers directly itself to the sip trunk provider?

Thanks



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[OpenSIPS-Users] opensips sip trunk

[Francjos]

How to configure a SIP trunk in opensips?I'm using two opensips servers as
load balancer and the scenario is that one and only one  of the two opensips
will connect to the sip server of the provider via a sip trunk. How can i
configure opensips for that purpose?

Thank you



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Re: [OpenSIPS-Users] Sip trunking and heartbeat

[Francjos]

And how do you that, i mean what should you configure?how do you use the IP
adress?



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[OpenSIPS-Users] Sip trunking and heartbeat

[Francjos]

Hello everyone,

I'm using heartbeat on two opensips servers , one active and the other
passive. I would like to d this:
When the one which active does not work, the other which is passive cane
replace it and continue receiving calls.
The problem i have is the following:
how  can i do in order one opensips is connected to the sip trunk, and when
it geos down, the other opensips have to connect to the sip trunk and
continue to receive the calls?

Thanks



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[OpenSIPS-Users] Opensips installing

[Francjos]

After taking the software from the repository using the commandes that
follow:

apt-key adv --keyserver keyserver.ubuntu.com --recv-keys 5F2FBB7C
echo "deb http://apt.opensips.org trusty 2.1-releases"
>>/etc/apt/sources.list

I've just installed the packages and i woul like to ask how i can install
the software itsel and in which fine i can perform the installation

Thanks

NDUWAYEZU Joselyne

-- 
Haute École de Bruxelles




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[OpenSIPS-Users] Installing openSIPS

[Francjos]

Hi,
Sorry for my english.I'm instaling opensips, i've used the following
commands to copy the software from the repository:
"apt-key adv --keyserver keyserver.ubuntu.com --recv-keys 5F2FBB7C
echo "deb http://apt.opensips.org trusty 2.1-releases"
>>/etc/apt/sources.list "

I've just installed the packages and i would like to ask what i'm gonna do
know in order to install the software itself and in which file i can prform
the installation.

Thakns



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