[OpenSIPS-Users] Jitsi sent REGISTER, Opensips received nothing
I have a test with one Jitsi using Opensips on the Internet Wireshark showed me that Jitsi sent several REGISTER packages, in the same time I used the command tcpdump to listen on the Opensips Server , but got nothing. Anyway, on the server, opensips, rtpproxy, media-dispatcher, media-relay are running! what happened to opensips server? why it won't response to distanced request? I appreciate your opinion, thanks a lot! Best regards -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Jitsi-sent-REGISTER-Opensips-received-nothing-tp7596019.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Jitsi sent REGISTER, Opensips received nothing
I would take a look at the network in between. If Jitsi is sending out packets, and opensips is not receiving, then look at routers, switches and firewall in between. Are you able to ping the opensips device from the same network that Jitsi is on? This should verify basic routing, then start looking at any potential firewall rules. If you are not getting ping requests, then you will want to check your routing with opensips. Make sure can ping outside of your network that Jitsi is on as well. Just my thoughts, Gordon Sent from my iPhone6 On Mar 19, 2015, at 1:23 AM, jacky 542590...@qq.com wrote: I have a test with one Jitsi using Opensips on the Internet Wireshark showed me that Jitsi sent several REGISTER packages, in the same time I used the command tcpdump to listen on the Opensips Server , but got nothing. Anyway, on the server, opensips, rtpproxy, media-dispatcher, media-relay are running! what happened to opensips server? why it won't response to distanced request? I appreciate your opinion, thanks a lot! Best regards -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Jitsi-sent-REGISTER-Opensips-received-nothing-tp7596019.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users - This email was processed through Nexepe Spam filter to filter junk messages. If you feel this message has been tagged incorrectly, you can change its category by clicking the link below. Click here http://10.0.9.41:5272/FrontController?operation=mbeuf=1_-940_20150319_183500.emlchkBayesian=1pr=1mt=1ma=s to mark email as junk. - Think before you Print. This e-mail may contain confidential and privileged material for the sole use of the intended recipient. Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender by reply e-mail and delete all copies of this message. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Which port router open
Hello Everyone, I'm trying to setup my personal sip server. In this moment it works only in my network, but now I want to open the router port for external access. I have open the port 5060 but maybe it's not enough. Thank you, Mattia ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] [Release] OpenSIPS 2.1 Release Candidate is out !
On 03/18/2015 08:26 PM, Bogdan-Andrei Iancu wrote: Hi all, I'm really excited to announce that after almost one year of hard work, the OpenSIPS 2.1 RC is now available for download. http://opensips.org/pub/opensips/latest/src/ The 2.1 version is a major step in OpenSIPS evolution, encapsulating major redesign (protocols, timers, reactors, async support) and valuable additions (websockets, Fraud Detection, SIP Compression, Async DB queries, Topology Hiding and more). http://www.opensips.org/About/Version-2-1-0-Notes OpenSIPS 2.1 is the first in line benefiting of the New Design (NG) brainstorming and work - where many radical and innovative concepts have been introduced, to put OpenSIPS on the top of the SIP server engines - in terms of efficiency, scalability, features set and flexibility. OpenSIPS 2.1 is the result of the whole community effort - there were many people involved in the design, in implementation, testing and reporting - and we want to thank you all for making this release possible: https://github.com/OpenSIPS/opensips/blob/2.1/CREDITS Version 2.1 is a release candidate - the stable GA version is to be release in the next 1.5 month, after more in depth testing . In the mean while, enjoy the fantastic ride with OpenSIPS 2.1 :) For a better experience with the system provisioning and user management, we recommend you to use the new OpenSIPS Control Panel 6.1[1], which is now fully compatible with OpenSIPS 2.1. Many greetings from the OpenSIPS team [2]! [1] https://sourceforge.net/projects/opensips-cp [2] http://opensips.org/images/team/opensips-team.jpg Best regards, Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Which port router open
Default SIP port 5060 UDP also you need media port call RTP -- Sent from my iPhone On Mar 19, 2015, at 7:29 AM, Mattia Adducchio m.adducc...@progel.net wrote: Hello Everyone, I'm trying to setup my personal sip server. In this moment it works only in my network, but now I want to open the router port for external access. I have open the port 5060 but maybe it's not enough. Thank you, Mattia ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] 500 Server error in REGISTER message
There's a mismatched curly-brace issue in your configuration.
Brace at line 2 matches with brace at line 18. Nothing matches with the
closing curly-brace at line 19. I think we are missing a curly-brace at
line 15.
My two-cents to the OpenSIPS team would be consider verbose curly-braces
for the configuration script. Python/C like tab-based indenting might
seem to improve readability and keep the code concise, but it introduces
these unintended errors.
```
1 if (is_method(REGISTER))
2 {
3 # authenticate the REGISTER requests (uncomment to enable auth)
4 ##if (!www_authorize(, subscriber))
5 ##{
6 ## www_challenge(, 0);
7 ## exit;
8 ##}
9 ##
10 ##if (!db_check_to())
11 ##{
12 ## sl_send_reply(403,Forbidden auth ID);
13 ## exit;
14 ##}
15 if (!save(location))
16 sl_reply_error();
17 exit;
18 }
19 }
20
```
On 3/17/15 10:52 AM, Satish Patel wrote:
I got those code from Book Building Telephony System with OpenSIPS 1.6
Here is the code from book
if (is_method(REGISTER))
{
# authenticate the REGISTER requests (uncomment to enable auth)
##if (!www_authorize(, subscriber))
##{
## www_challenge(, 0);
## exit;
##}
##
##if (!db_check_to())
##{
## sl_send_reply(403,Forbidden auth ID);
## exit;
##}
if (!save(location))
sl_reply_error();
exit;
}
}
On Tue, Mar 17, 2015 at 1:48 PM, Satish Patel satish@gmail.com
mailto:satish@gmail.com wrote:
Eric,
I found what was the issue, I sent you REGISTER method snippet
before, if you look at it, If remove/comment out sl_reply_error();
line in following code, it stopped sending 500 Error. Very
interesting.. Do you think i need to put that in curly braces { } ?
if (!save(location))
xlog(L_ERR, Saving contact failed - M=$rm
RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n);
sl_reply_error();
exit;
}
On Tue, Mar 17, 2015 at 1:27 PM, Satish Patel satish@gmail.com
mailto:satish@gmail.com wrote:
Even after disabled siptrace it is happening. no luck :(
On Tue, Mar 17, 2015 at 1:20 PM, Eric Tamme e...@uphreak.com
mailto:e...@uphreak.com wrote:
Turn of your sip tracing and see if the issue occurs. Its
running some sl_callbacks (which i assume are realated to
siptrace).
On 03/17/2015 11:19 AM, Satish Patel wrote:
I haven't done anything related stateless. also in my
config, i haven't manually specify that 500 error anywhere
where i can doubt. I don't know from where it is coming.
must be internally from opensips.
On Tue, Mar 17, 2015 at 1:14 PM, Eric Tamme
e...@uphreak.com mailto:e...@uphreak.com wrote:
Ah - nm, i see it in an sl callback
Mar 17 22:19:01 sip2
/usr/local/opensips-2-head/sbin/opensips[31285]: DBG:sl:sl_reply_error:
error text is Server error occurred (1/SL)
... so are you doing anything statless in your config? This
looks like it might be siptrace related.
On 03/17/2015 11:11 AM, Eric Tamme wrote:
I do not see the 500 from opensips in this log.
On 03/17/2015 11:07 AM, Satish Patel wrote:
Here is the debug 4 logs http://pastebin.com/CdPxFrNp
173.48.111.111 - UA
188.79.242.164 - OpenSIPs
On Tue, Mar 17, 2015 at 12:45 PM, Eric Tamme
e...@uphreak.com mailto:e...@uphreak.com wrote:
This is a ladder diagram, not a sip trace. A
ladder diagram is not useful in this case.
Turn your debug up to 4, capture the log of the
register/500 happening and submit a link to the
pastebin. DO NOT paste the contents into an email.
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[OpenSIPS-Users] SIps as SBC
Can SIPS can be used as an SBC? Thanks Abdul ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIps as SBC
Who? T ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIps as SBC
Checkout Blox.org Thanks Varadhan M On Thu, Mar 19, 2015 at 7:33 PM, malik sherif asheri...@hotmail.com wrote: Can SIPS can be used as an SBC? Thanks Abdul ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Presence Error messages
I am attempting to troubleshoot what I think is a presence/b2b_sca issue. I keep getting an error message from presence as follows: ERROR:presence:update_presentity: No E_Tag match [ff5ad69c9be06cffaa136492f4fb3b50] At some point during the day, I will see this error message: ERROR:presence:handle_subscribe: in event specific subscription handling At this point, an outbound call from a line appearance provisioned to the b2b_sca module fails. The outbound call is being attempted from a Cisco SPA525G2, and the message on the phone screen shows no line-despite happening at a time when it is known there are no calls on the system that I can see. The observed behavior of the phone is to show No line, and the phone itself gets a What do these error messages mean? What is the system trying to tell me? I am trying to get my brain around what the error messages are saying so I can figure out where to look next in my troubleshooting. The proxy is functioning well in all other respects. Cordially, Peter Nayland Kust Director of Technologies BusinesSuites 24624 Interstate 45 North, Suite 200 Houston, TX 77386 Tel: 281.378.8051 Fax: 855.287.6961 peter.k...@businessuites.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] WebSocket Support in OpenSIPS 2.1
Great news, Are there any media engines that can be used in conjunction with opensips that would allow the interop between webrtc sip clients and standard sip? I am aware that freeswitch will currently do this. Thanks, Tito On Wed, Mar 18, 2015 at 3:36 AM, Saúl Ibarra Corretgé s...@ag-projects.com wrote: Congrats to the team! Great work! On 17 Mar 2015, at 20:00, Răzvan Crainea raz...@opensips.org wrote: Hello, all! Aaand, we're finally making it official: OpenSIPS 2.1 will have *WebSockets* support! Are you planning to use (or perhaps you're already using) WebRTC based SIP clients, but you are having hard time setting up the platform? Starting from now, it has never been simpler - based on your needs and feedback[1] we decided to implement a WebSocket server directly in OpenSIPS. And we're doing it now! Starting with the new OpenSIPS 2.1 release you will be able to plug your web-based SIP clients directly in your OpenSIPS server using the new WebSocket transport protocol[2]. We've also setup a short tutorial[3] for you to integrate this feature in your platform easier. Many thanks to Eric Tamme (lirakis) for all his help with the tutorial as well as for the intensive tests. [1] http://www.opensips.org/Community/IRCmeeting20141029 [2] http://www.opensips.org/html/docs/modules/2.1.x/proto_ws [3] http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1 Best regards, -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Saúl Ibarra Corretgé AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ERROR:dialog:dlg_validate_dialog:
Hello,
Just to recap, you are saying that the Contact the user agent is sending
is broken and you are happy that OpenSIPS is properly fixing the
message, but you want to get rid of the ERRORs in the log ? If this is
the case, you can use setdebug [1] for this.
Try something like
setdebug(-3)
if ($DLG_status!=NULL !validate_dialog() ) {
xlog( in-dialog bogus request \n);
fix_route_dialog();
}
setdebug()
http://www.opensips.org/Documentation/Script-CoreFunctions-2-1#toc48
Best Regards,
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com
On 18.03.2015 22:47, Satish Patel wrote:
I know you guys are super busy in OpenSIPS 2.1 release, but any
suggestion on above issue?
On Wed, Mar 18, 2015 at 12:17 AM, Satish Patel satish@gmail.com
mailto:satish@gmail.com wrote:
I am getting following error in log, I can understand my contact:
and Route: values mismatching here. why it is happening? is there
a way to get rid on this error?
Following is scenario. Only getting error in BYE message.
[UA][OpenSIP]---[Freeswitch]-[Opensip]-[SIP
Provide]
ERROR:dialog:dlg_validate_dialog: failed to validate remote
contact: dlg=[sip:16463737221
tel:16463737221@188.178.235.222:5061;transport=udp] ,
req=[sip:188.178.235.222;lr;ftag=840e2e35;did=1f4.ca6a6956]
I am using fix_route_dialog() in loose_route()
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route() || match_dialog()) {
if ($DLG_status!=NULL
!validate_dialog() ) {
xlog( in-dialog bogus request \n);
fix_route_dialog();
}
xlog(L_INFO, Loose route failed on
$hdr(route)\n);
if (is_method(BYE)) {
#setflag(ACC_DO); # do accounting ...
#setflag(ACC_FAILED); # ... even
if the transaction fails
} else if (is_method(INVITE)) {
# even if in most of the cases is
useless, do RR for
# re-INVITEs alos, as some buggy
clients do change route set
# during the dialog.
record_route();
}
if (check_route_param(nat=yes))
setflag(NAT);
# route it out to whatever destination was
set by loose_route()
# in $du (destination URI).
route(relay);
} else {
if ( is_method(ACK) ) {
if ( t_check_trans() ) {
# non loose-route, but
stateful ACK; must be an ACK after
# a 487 or e.g. 404 from
upstream server
xlog(non loose-route
section\n);
t_relay();
exit;
} else {
# ACK without matching
transaction -
# ignore and discard
xlog(ACK without matching
transaction\n);
exit;
}
}
xlog(L_INFO, destination uri after
loose_route: $du\n);
sl_send_reply(404,Not here);
}
exit;
}
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Re: [OpenSIPS-Users] error 483
Hello, 483 usually means 'Too Many Hops'. If you do a SIP trace on the server, do you see OpenSIPS looping the request to itself ? Maybe the SIP phone sends the IP of the server instead of the domain that you have configured, and your script is configured to route out such requests. Best Regards, Vlad Paiu OpenSIPS Developer http://www.opensips-solutions.com On 19.03.2015 04:15, Carlos Cruz wrote: Hi; Can someone tell me why or where I may find the info I need; I'm able to register external remote phones (hard phones), but the internal phone (soft phones) within the same network as the OpenSIPS test server report error 483. Thanks!! Carlos ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ERROR:dialog:dlg_validate_dialog:
Great! will give it a shot!
Just surprised why it is not matching both dlg and req? does
fix_route_dialog();
has any impact on system when you have very high CPS etc?
It would be good if fix issue from root, instead of external resources
which eat CPU ticks :)
dlg=[sip:16463737221@188.178.235.222:5061;transport=udp] ,
req=[sip:188.178.235.222;lr;ftag=840e2e35;did=1f4.ca6a6956]
On Thu, Mar 19, 2015 at 12:24 PM, Vlad Paiu vladp...@opensips.org wrote:
Hello,
Just to recap, you are saying that the Contact the user agent is sending
is broken and you are happy that OpenSIPS is properly fixing the message,
but you want to get rid of the ERRORs in the log ? If this is the case, you
can use setdebug [1] for this.
Try something like
setdebug(-3)
if ($DLG_status!=NULL !validate_dialog() ) {
xlog( in-dialog bogus request \n);
fix_route_dialog();
}
setdebug()
http://www.opensips.org/Documentation/Script-CoreFunctions-2-1#toc48
Best Regards,
Vlad Paiu
OpenSIPS Developerhttp://www.opensips-solutions.com
On 18.03.2015 22:47, Satish Patel wrote:
I know you guys are super busy in OpenSIPS 2.1 release, but any suggestion
on above issue?
On Wed, Mar 18, 2015 at 12:17 AM, Satish Patel satish@gmail.com
wrote:
I am getting following error in log, I can understand my contact: and
Route: values mismatching here. why it is happening? is there a way to get
rid on this error?
Following is scenario. Only getting error in BYE message.
[UA][OpenSIP]---[Freeswitch]-[Opensip]-[SIP
Provide]
ERROR:dialog:dlg_validate_dialog: failed to validate remote contact:
dlg=[sip:16463737221@188.178.235.222:5061;transport=udp] ,
req=[sip:188.178.235.222;lr;ftag=840e2e35;did=1f4.ca6a6956]
I am using fix_route_dialog() in loose_route()
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route() || match_dialog()) {
if ($DLG_status!=NULL !validate_dialog() ) {
xlog( in-dialog bogus request \n);
fix_route_dialog();
}
xlog(L_INFO, Loose route failed on
$hdr(route)\n);
if (is_method(BYE)) {
#setflag(ACC_DO); # do accounting ...
#setflag(ACC_FAILED); # ... even if the
transaction fails
} else if (is_method(INVITE)) {
# even if in most of the cases is
useless, do RR for
# re-INVITEs alos, as some buggy clients
do change route set
# during the dialog.
record_route();
}
if (check_route_param(nat=yes))
setflag(NAT);
# route it out to whatever destination was set by
loose_route()
# in $du (destination URI).
route(relay);
} else {
if ( is_method(ACK) ) {
if ( t_check_trans() ) {
# non loose-route, but stateful
ACK; must be an ACK after
# a 487 or e.g. 404 from upstream
server
xlog(non loose-route section\n);
t_relay();
exit;
} else {
# ACK without matching
transaction -
# ignore and discard
xlog(ACK without matching
transaction\n);
exit;
}
}
xlog(L_INFO, destination uri after
loose_route: $du\n);
sl_send_reply(404,Not here);
}
exit;
}
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Re: [OpenSIPS-Users] Dispatcher user specific route question - 2.1
Hello, If you want to do dispatching between multiple setids, ds_select_dst() allows that. See the docs at [1] , you can provide a comma separated list of setids - so your $avp(zone) can contain '1,2' and OpenSIPS will try to first send to the servers in setid 1, and then, if those fail, to the servers in setid 2. [1] http://www.opensips.org/html/docs/modules/1.11.x/dispatcher#id294368 Best Regards, Vlad Paiu OpenSIPS Developer http://www.opensips-solutions.com On 19.03.2015 06:17, Satish Patel wrote: I have add extra zone column in subscriber table, +--+-+ | username | zone | +--+-+ |1001 |1| |1002 |2| +--+-+ In dispatcher table I have following two Freeswitch in two groups. +---+-++ | setid | destination | description| +---+--+---+ | 1 | sip:fs1.example.com http://fs1.example.com | Freeswitch-1 | | 2 | sip:fs2.example.com http://fs2.example.com | Freeswitch-2 | +---+--+---+ in opensips.cfg script i am query subscriber table base on incoming username and storing zone in avp(zone) variable, and calling same variable in following code if ( !ds_select_dst($avp(zone), 4, FM10)) Question: now either user belongs to zone 1 or 2, so it is *NOT* going to do load-balancing between two. But if I want to allow some user to do load-balancing then how it will be possible in above scenario? Can i set setid on fly so i can pass request along with user request and set same group for both switch and user call load-balance on both switch? Any other idea? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Jitsi sent REGISTER, Opensips received nothing
Hello, Well, if you did a tcpdump on the OpenSIPS box and saw nothing, then it means the packages aren't actually reaching the box. Please check that there are no firewalls in between the client and OpenSIPS that are blocking the traffic. Best Regards, Vlad Paiu OpenSIPS Developer http://www.opensips-solutions.com On 19.03.2015 08:23, jacky wrote: I have a test with one Jitsi using Opensips on the Internet Wireshark showed me that Jitsi sent several REGISTER packages, in the same time I used the command tcpdump to listen on the Opensips Server , but got nothing. Anyway, on the server, opensips, rtpproxy, media-dispatcher, media-relay are running! what happened to opensips server? why it won't response to distanced request? I appreciate your opinion, thanks a lot! Best regards -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Jitsi-sent-REGISTER-Opensips-received-nothing-tp7596019.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Again BLF and Presence with Snom 7xx phones and OpenSips
You need to handle the in-dialog SUBSCRIBE requests. eg:
if has_totag() {
...
if (loose_route()) {
...
} else {
...
if (is_method(SUBSCRIBE)) {
route(2);
exit;
}
...
}
...
}
From: Bogdan-Andrei Iancu
Sent: Thursday, February 26, 2015 7:56 AM
To: OpenSIPS users mailling list ; mailto:michele.pina...@unisi.it
Subject: Re: [OpenSIPS-Users] Again BLF and Presence with Snom 7xx phones and
OpenSips
Hi Michele,
The problem in your script is that you do not handle the sequential (in-dialog)
SUBSCRIBE requests (as you have the second one in your trace, ending with 404
and terminating the subscription).
In the if ( has_totag() ) block, you have:
} else {
if (is_method(SUBSCRIBE) $rd == 127.0.0.1:5060) { # CUSTOMIZE ME
The $rd detection does not cover all your cases, as you configure the presence
module to advertise as SIP contact sip:prese...@voip.unisi.it:5060. So, the
test fails.
You can adapt the test like:
if (is_method(SUBSCRIBE) $rd == voip.unisi.it) { # CUSTOMIZE ME
Or set the contact in presence with the real IP:
modparam(presence, server_address,
mailto:sip:presence@127.0.0.1:5060)
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.comOn 24.02.2015 12:04, Michele Pinassi wrote:
Hi all,
I'm still stuck on this issue: BLF not working. For example, on my SNOM 760
(ext 5002) i activated BLF for some ext, like 5020. Using SIPGREP i saw:
SUBSCRIBE sip:5...@voip.unisi.it;user=phone SIP/2.0.
Via: SIP/2.0/UDP 172.20.1.10:57286;branch=z9hG4bK-nprg3gvnk4q1;rport.
From: mailto:sip:5...@voip.unisi.it:5060;tag=nyux2omhly.
To: mailto:sip:5...@voip.unisi.it;user=phone.
Call-ID: 3944ec54dc20-pfzjpjhrpm6p.
CSeq: 2 SUBSCRIBE.
Max-Forwards: 70.
Contact: mailto:sip:5002@172.20.1.10:57286;reg-id=1.
Event: dialog.
Accept: application/dialog-info+xml.
User-Agent: snom760/8.7.3.25.9.
Proxy-Authorization: Digest
Expires: 3600.
Content-Length: 0.
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
172.20.1.10:57286;received=172.20.1.10;branch=z9hG4bK-nprg3gvnk4q1;rport=57286.
From: mailto:sip:5...@voip.unisi.it:5060;tag=nyux2omhly.
To:
mailto:sip:5...@voip.unisi.it;user=phone;tag=f315b2d58ae8829149b784764c5a40e3-163d.
Call-ID: 3944ec54dc20-pfzjpjhrpm6p.
CSeq: 2 SUBSCRIBE.
Expires: 3600.
Contact: mailto:sip:prese...@voip.unisi.it:5060.
Server: OpenSIPS (1.11.3-tls (i386/linux)).
Content-Length: 0.
NOTIFY sip:5002@172.20.1.10:57286 SIP/2.0.
Via: SIP/2.0/UDP 172.20.1.2:5060;branch=z9hG4bKdb02.83d58916.0.
To: mailto:sip:5...@voip.unisi.it;tag=nyux2omhly.
From: mailto:sip:5...@voip.unisi.it;tag=f315b2d58ae8829149b784764c5a40e3-163d.
CSeq: 1 NOTIFY.
Call-ID: 3944ec54dc20-pfzjpjhrpm6p.
Max-Forwards: 70.
Content-Length: 147.
User-Agent: OpenSIPS (1.11.3-tls (i386/linux)).
Event: dialog.
Contact: mailto:sip:prese...@voip.unisi.it:5060.
Subscription-State: active;expires=3600.
Content-Type: application/dialog-info+xml.
.
?xml version=1.0?
dialog-info xmlns=urn:ietf:params:xml:ns:dialog-info version=0
state=full entity=mailto:sip:5...@voip.unisi.it/
SIP/2.0 200 Ok.
Via: SIP/2.0/UDP 172.20.1.2:5060;branch=z9hG4bKdb02.83d58916.0.
From: mailto:sip:5...@voip.unisi.it;tag=f315b2d58ae8829149b784764c5a40e3-163d.
To: mailto:sip:5...@voip.unisi.it;tag=nyux2omhly.
Call-ID: 3944ec54dc20-pfzjpjhrpm6p.
CSeq: 1 NOTIFY.
Content-Length: 0.
SUBSCRIBE sip:prese...@voip.unisi.it:5060 SIP/2.0.
Via: SIP/2.0/UDP 172.20.1.25:32768;branch=z9hG4bK-lbgnea3kuorq;rport.
From: mailto:sip:5...@voip.unisi.it:5060;tag=w8vp9q5iyn.
To:
mailto:sip:5...@voip.unisi.it;user=phone;tag=f315b2d58ae8829149b784764c5a40e3-29cc.
Call-ID: 54ec3a578c9e-klgn0s3i32zo.
CSeq: 75 SUBSCRIBE.
Max-Forwards: 70.
Contact: mailto:sip:5007@172.20.1.25:32768;reg-id=1.
Event: dialog.
Accept: application/dialog-info+xml.
User-Agent: snom710/8.7.3.25.9.
Expires: 3600.
Content-Length: 0.
SIP/2.0 404 Not here.
Via: SIP/2.0/UDP
172.20.1.25:32768;received=172.20.1.25;branch=z9hG4bK-lbgnea3kuorq;rport=32768.
From: mailto:sip:5...@voip.unisi.it:5060;tag=w8vp9q5iyn.
To:
mailto:sip:5...@voip.unisi.it;user=phone;tag=f315b2d58ae8829149b784764c5a40e3-29cc.
Call-ID: 54ec3a578c9e-klgn0s3i32zo.
CSeq: 75 SUBSCRIBE.
Server: OpenSIPS (1.11.3-tls (i386/linux)).
Content-Length: 0.
NOTIFY sip:5002@172.20.1.10:57286 SIP/2.0.
Via: SIP/2.0/UDP 172.20.1.2:5060;branch=z9hG4bKdbe9.7966c706.0.
To: mailto:sip:5...@voip.unisi.it;tag=iklb1qjh1v.
From: mailto:sip:5...@voip.unisi.it;tag=f315b2d58ae8829149b784764c5a40e3-b571.
CSeq: 2 NOTIFY.
Call-ID: ee35ec54a72b-draf1nwo4qn7.
Max-Forwards: 70.
Content-Length: 0.
User-Agent: OpenSIPS (1.11.3-tls (i386/linux)).
Event: dialog.
Contact: mailto:sip:prese...@voip.unisi.it:5060.
Subscription-State: terminated;reason=timeout.
SIP/2.0 200 Ok.
Via: SIP/2.0/UDP
Re: [OpenSIPS-Users] Dispatcher user specific route question - 2.1
Thanks Vlad, Superb! so it will do round-robin? or fail-over? On Thu, Mar 19, 2015 at 12:30 PM, Vlad Paiu vladp...@opensips.org wrote: Hello, If you want to do dispatching between multiple setids, ds_select_dst() allows that. See the docs at [1] , you can provide a comma separated list of setids - so your $avp(zone) can contain '1,2' and OpenSIPS will try to first send to the servers in setid 1, and then, if those fail, to the servers in setid 2. [1] http://www.opensips.org/html/docs/modules/1.11.x/dispatcher#id294368 Best Regards, Vlad Paiu OpenSIPS Developerhttp://www.opensips-solutions.com On 19.03.2015 06:17, Satish Patel wrote: I have add extra zone column in subscriber table, +--+-+ | username | zone | +--+-+ |1001 |1| |1002 |2| +--+-+ In dispatcher table I have following two Freeswitch in two groups. +---+-++ | setid | destination | description| +---+--+---+ | 1 | sip:fs1.example.com | Freeswitch-1 | | 2 | sip:fs2.example.com | Freeswitch-2 | +---+--+---+ in opensips.cfg script i am query subscriber table base on incoming username and storing zone in avp(zone) variable, and calling same variable in following code if ( !ds_select_dst($avp(zone), 4, FM10)) Question: now either user belongs to zone 1 or 2, so it is *NOT* going to do load-balancing between two. But if I want to allow some user to do load-balancing then how it will be possible in above scenario? Can i set setid on fly so i can pass request along with user request and set same group for both switch and user call load-balance on both switch? Any other idea? ___ Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Dispatcher user specific route question - 2.1
Hello, It will do fail-over. Best Regards, Vlad Paiu OpenSIPS Developer http://www.opensips-solutions.com On 19.03.2015 18:39, Satish Patel wrote: Thanks Vlad, Superb! so it will do round-robin? or fail-over? On Thu, Mar 19, 2015 at 12:30 PM, Vlad Paiu vladp...@opensips.org mailto:vladp...@opensips.org wrote: Hello, If you want to do dispatching between multiple setids, ds_select_dst() allows that. See the docs at [1] , you can provide a comma separated list of setids - so your $avp(zone) can contain '1,2' and OpenSIPS will try to first send to the servers in setid 1, and then, if those fail, to the servers in setid 2. [1] http://www.opensips.org/html/docs/modules/1.11.x/dispatcher#id294368 Best Regards, Vlad Paiu OpenSIPS Developer http://www.opensips-solutions.com On 19.03.2015 06:17, Satish Patel wrote: I have add extra zone column in subscriber table, +--+-+ | username | zone | +--+-+ |1001 |1| |1002 |2| +--+-+ In dispatcher table I have following two Freeswitch in two groups. +---+-++ | setid | destination | description| +---+--+---+ | 1 | sip:fs1.example.com http://fs1.example.com | Freeswitch-1 | | 2 | sip:fs2.example.com http://fs2.example.com | Freeswitch-2 | +---+--+---+ in opensips.cfg script i am query subscriber table base on incoming username and storing zone in avp(zone) variable, and calling same variable in following code if ( !ds_select_dst($avp(zone), 4, FM10)) Question: now either user belongs to zone 1 or 2, so it is *NOT* going to do load-balancing between two. But if I want to allow some user to do load-balancing then how it will be possible in above scenario? Can i set setid on fly so i can pass request along with user request and set same group for both switch and user call load-balance on both switch? Any other idea? ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Dispatcher user specific route question - 2.1
Thanks! for quick answer!! On Thu, Mar 19, 2015 at 12:41 PM, Vlad Paiu vladp...@opensips.org wrote: Hello, It will do fail-over. Best Regards, Vlad Paiu OpenSIPS Developerhttp://www.opensips-solutions.com On 19.03.2015 18:39, Satish Patel wrote: Thanks Vlad, Superb! so it will do round-robin? or fail-over? On Thu, Mar 19, 2015 at 12:30 PM, Vlad Paiu vladp...@opensips.org wrote: Hello, If you want to do dispatching between multiple setids, ds_select_dst() allows that. See the docs at [1] , you can provide a comma separated list of setids - so your $avp(zone) can contain '1,2' and OpenSIPS will try to first send to the servers in setid 1, and then, if those fail, to the servers in setid 2. [1] http://www.opensips.org/html/docs/modules/1.11.x/dispatcher#id294368 Best Regards, Vlad Paiu OpenSIPS Developerhttp://www.opensips-solutions.com On 19.03.2015 06:17, Satish Patel wrote: I have add extra zone column in subscriber table, +--+-+ | username | zone | +--+-+ |1001 |1| |1002 |2| +--+-+ In dispatcher table I have following two Freeswitch in two groups. +---+-++ | setid | destination | description| +---+--+---+ | 1 | sip:fs1.example.com | Freeswitch-1 | | 2 | sip:fs2.example.com | Freeswitch-2 | +---+--+---+ in opensips.cfg script i am query subscriber table base on incoming username and storing zone in avp(zone) variable, and calling same variable in following code if ( !ds_select_dst($avp(zone), 4, FM10)) Question: now either user belongs to zone 1 or 2, so it is *NOT* going to do load-balancing between two. But if I want to allow some user to do load-balancing then how it will be possible in above scenario? Can i set setid on fly so i can pass request along with user request and set same group for both switch and user call load-balance on both switch? Any other idea? ___ Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIP/2.0 477 Send failed (477/TM) - Route
Hi Leo, If you look in your logs, you should see some errors where OpenSIPS complains about not being able to open some TCP connection. Basically OpenSIPS tried to forward the call by TCP but failed for some reasons (TCP related). Check the logs. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 17.03.2015 18:37, leo wrote: Hello, I'm receiving the following message when a try to place a call: SIP/2.0 477 Send failed (477/TM) This is desired issue because the callee UA is not online and in userloc it is not expired yet. My question is, which would be the process or the route this event (477 Send failed) is processed? I've tried to log on failure_route, onreply_route and even on branch_route but it was unsuccessful. Thanks a lot, Leo -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/SIP-2-0-477-Send-failed-477-TM-Route-tp7595929.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
