[OpenSIPS-Users] Large domain list buffer error?
Greetings all, I have a relatively large domain list and when performing a "opensipsctl domain show" I get the following error on the console and only 75% or so of the domains show up: ERROR:mi_fifo:mi_write_tree: failed to write - EOC does not fit in! Is there a place I should allocate more buffer space for this operation to complete properly? Thanks! -Ryan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] dialog replication
Hello Bogdan,
Yes, Razvan identified issue where $fs wasn't set properly. I fixed
this on script level not sure if this clean solution.
https://paste.fedoraproject.org/paste/lIzEh5Q1vd4XePW6oUgC2g
any insight thank you.
volga629
On Thu, Aug 30, 2018 at 1:20 PM, Bogdan-Andrei Iancu
wrote:
Hi,
I see the INVITE is received and sent out on the same interface :
207.210.246.38:5060, so there is no interface exchange, so no reason
for a double RR.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Bootcamp 2018
http://opensips.org/training/OpenSIPS_Bootcamp_2018/
On 08/28/2018 07:20 PM, volga...@networklab.ca wrote:
Hello Everyone,
What possible cause that dialog is removing Record-Route for second
interface, that cause send call to WAN instead the LAN.
https://paste.fedoraproject.org/paste/XsqlLhO0CYteE3APcVkKpw
route {
if(!has_totag() && is_method("INVITE")) {
create_dialog();
xlog("L_INFO", "Got request on ip addr [$Ri] and call
dir $avp(DLG_dir)\n");
# Wan route
$var(ip_lst) = $shv(vip_wan_lst);
route(SET_SOURCE_SOCKET);
if($avp(DLG_dir)=="topbx") {
switch($(var(req_ip){s.select,3,.})) {
case "38":
set_dlg_sharing_tag("vip1");
xlog("L_INFO", "[$rm] Set dialog tag
vip1 ~> $(var(req_ip){s.select,3,.})\n");
break;
case "39":
set_dlg_sharing_tag("vip2");
xlog("L_INFO", "[$rm] Set dialog tag
vip2 ~> $(var(req_ip){s.select,3,.})\n");
break;
case "40":
set_dlg_sharing_tag("vip3");
xlog("L_INFO", "[$rm] Set dialog tag
vip3 ~> $(var(req_ip){s.select,3,.})\n");
break;
default:
xlog("L_INFO", "[$rm] Unknown last
ocetet ~> $(var(req_ip){s.select,3,.})\n");
}
}
if(!is_method("REGISTER")) {
record_route();
}
}
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Re: [OpenSIPS-Users] Doubt about call center module
Hi Bogdan
Yes, It's the same scenario and same message. The call flow is:
Asterisk Dials(port 5070) -> Opensips (port 5060) forward to Queue -> Calls
local user
I'm using standard Queue scenario:
server1
client1
message
1
1
And SIP message is the same on all calls, just changed Call-id/tags:
U 10.10.10.10:5070 -> 10.10.10.10:5060
INVITE sip:fila-1@10.10.10.10:5060 SIP/2.0.
Via: SIP/2.0/UDP 10.10.10.10:5070;branch=z9hG4bK2abb2acc;rport.
Max-Forwards: 70.
From: ;tag=as6440e239.
To: .
Contact: .
Call-ID: 357cf76348e4e68325d065e85282320a@10.10.10.10:5070.
CSeq: 102 INVITE.
User-Agent: PBX SIPTEK.
Date: Thu, 30 Aug 2018 17:30:30 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE.
Supported: replaces, timer.
P-Asserted-Identity: "5511" .
Content-Type: application/sdp.
Content-Length: 353.
[SDP OMMITED]
I updated to latest 2.4.2 GIT version (commit
8b6830cdd96298682fcc298095ad1b718c54c77d), same problem is happening.
Also you can access the server if you want, it's dedicated to this test.
Thanks
On Thu, Aug 30, 2018 at 1:04 PM Bogdan-Andrei Iancu
wrote:
> Hi Daniel,
>
> Are you sure you configured a proper SIP URI as "message_queue" in the
> flow description ? My impression is you have an empty string there - and
> OpenSIPS is trying to put the call on the queue (as there is no agent), but
> the SIP URI is not valid.
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
> OpenSIPS Bootcamp 2018
> http://opensips.org/training/OpenSIPS_Bootcamp_2018/
>
> On 08/29/2018 10:26 PM, Daniel Zanutti wrote:
>
> Got some more info.
>
> *This is the first call that worked fine:*
> ..
>
> *This is the second call that had the problem:*
> .
> Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
> DBG:call_center:cc_call_state_machine: selecting QUEUE
> Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
> DBG:call_center:cc_queue_push_call: QUEUE - adding call 0x7fd8510524a8
> Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
> DBG:call_center:cc_queue_push_call: adding call on pos 0 (already 1 calls),
> l=(nil) h=(nil)
> Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
> DBG:call_center:w_handle_call: new destination for call(0x7fd8510524a8) is
> (state=2)
> .
>
>
> On Mon, Aug 27, 2018 at 6:15 PM Daniel Zanutti
> wrote:
>
>> Trying to configure the call center modules, but found a problem when
>> there is no agents available.
>>
>> If there is 1 agent available, call is sent to him with no problem:
>>
>> Aug 27 18:11:00 plat5 /sbin/opensips[23575]: Recebida asterisk - Tentando
>> entrar na fila fila-1
>> Aug 27 18:11:00 plat5 /sbin/opensips[23575]: Entrou na fila com sucesso
>> (fila-1)!
>> Aug 27 18:11:01 plat5 /sbin/opensips[23569]: incoming reply
>>
>> But when there is no agent available, opensips refuses:
>> Aug 27 18:11:07 plat5 /sbin/opensips[23569]: Recebida asterisk - Tentando
>> entrar na fila fila-1
>> Aug 27 18:11:07 plat5 /sbin/opensips[23569]:
>> ERROR:b2b_logic:b2b_process_scenario_init: Failed to get the value for the
>> b2b client ruri
>> Aug 27 18:11:07 plat5 /sbin/opensips[23569]:
>> ERROR:call_center:set_call_leg: failed to init new b2bua call (empty ID
>> received)
>> Aug 27 18:11:07 plat5 /sbin/opensips[23569]:
>> ERROR:call_center:w_handle_call: failed to set new destination for call
>> Aug 27 18:11:07 plat5 /sbin/opensips[23569]: errnum: -1
>>
>> Error -1 means flowID is invalid, but I sent the same value on both calls.
>>
>> This is the call:
>>
>> cc_handle_call("$rU")
>>
>> I'm using Opensips 2.4.2 with Debian 8.11.
>>
>> Am I missing something or found a bug?
>>
>> Thanks
>>
>
>
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>
>
>
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Re: [OpenSIPS-Users] Does topology hiding match RFC3261?
Thank you Vasilev ! Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 08/22/2018 11:19 AM, vasilevalex wrote: I created issue and sent possible patch https://github.com/OpenSIPS/opensips/issues/1447 -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html ___ 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] Async Radius support in RPM
Hi Dragomir, You still need to recompile the radius module by hand, after patching (as in 2.2). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 08/27/2018 03:23 PM, Dragomir Haralambiev wrote: Hello Team! In opensips 2.2.x I had to recompile opensips manually in order to apply the asynchronous radius request support. I am now planing on migrating the new 2.4 LTS. Do I still need to patch the code or I can simply use the pre-compiled binaries available trough the opensips repos? Best regards, Dragormir ___ 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] dialog replication
Hi,
I see the INVITE is received and sent out on the same interface :
207.210.246.38:5060, so there is no interface exchange, so no reason for
a double RR.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Bootcamp 2018
http://opensips.org/training/OpenSIPS_Bootcamp_2018/
On 08/28/2018 07:20 PM, volga...@networklab.ca wrote:
Hello Everyone,
What possible cause that dialog is removing Record-Route for second
interface, that cause send call to WAN instead the LAN.
https://paste.fedoraproject.org/paste/XsqlLhO0CYteE3APcVkKpw
route {
if(!has_totag() && is_method("INVITE")) {
create_dialog();
xlog("L_INFO", "Got request on ip addr [$Ri] and call
dir $avp(DLG_dir)\n");
# Wan route
$var(ip_lst) = $shv(vip_wan_lst);
route(SET_SOURCE_SOCKET);
if($avp(DLG_dir)=="topbx") {
switch($(var(req_ip){s.select,3,.})) {
case "38":
set_dlg_sharing_tag("vip1");
xlog("L_INFO", "[$rm] Set dialog tag
vip1 ~> $(var(req_ip){s.select,3,.})\n");
break;
case "39":
set_dlg_sharing_tag("vip2");
xlog("L_INFO", "[$rm] Set dialog tag
vip2 ~> $(var(req_ip){s.select,3,.})\n");
break;
case "40":
set_dlg_sharing_tag("vip3");
xlog("L_INFO", "[$rm] Set dialog tag
vip3 ~> $(var(req_ip){s.select,3,.})\n");
break;
default:
xlog("L_INFO", "[$rm] Unknown last
ocetet ~> $(var(req_ip){s.select,3,.})\n");
}
}
if(!is_method("REGISTER")) {
record_route();
}
}
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Re: [OpenSIPS-Users] Doubt about call center module
Hi Daniel,
Are you sure you configured a proper SIP URI as "message_queue" in the
flow description ? My impression is you have an empty string there - and
OpenSIPS is trying to put the call on the queue (as there is no agent),
but the SIP URI is not valid.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Bootcamp 2018
http://opensips.org/training/OpenSIPS_Bootcamp_2018/
On 08/29/2018 10:26 PM, Daniel Zanutti wrote:
Got some more info.
*This is the first call that worked fine:*
..
*This is the second call that had the problem:*
.
Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
DBG:call_center:cc_call_state_machine: selecting QUEUE
Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
DBG:call_center:cc_queue_push_call: QUEUE - adding call 0x7fd8510524a8
Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
DBG:call_center:cc_queue_push_call: adding call on pos 0 (already 1
calls), l=(nil) h=(nil)
Aug 29 16:04:38 plat5 /sbin/opensips[24890]:
DBG:call_center:w_handle_call: new destination for
call(0x7fd8510524a8) is (state=2)
.
On Mon, Aug 27, 2018 at 6:15 PM Daniel Zanutti
mailto:daniel.zanu...@gmail.com>> wrote:
Trying to configure the call center modules, but found a problem
when there is no agents available.
If there is 1 agent available, call is sent to him with no problem:
Aug 27 18:11:00 plat5 /sbin/opensips[23575]: Recebida asterisk -
Tentando entrar na fila fila-1
Aug 27 18:11:00 plat5 /sbin/opensips[23575]: Entrou na fila com
sucesso (fila-1)!
Aug 27 18:11:01 plat5 /sbin/opensips[23569]: incoming reply
But when there is no agent available, opensips refuses:
Aug 27 18:11:07 plat5 /sbin/opensips[23569]: Recebida asterisk -
Tentando entrar na fila fila-1
Aug 27 18:11:07 plat5 /sbin/opensips[23569]:
ERROR:b2b_logic:b2b_process_scenario_init: Failed to get the value
for the b2b client ruri
Aug 27 18:11:07 plat5 /sbin/opensips[23569]:
ERROR:call_center:set_call_leg: failed to init new b2bua call
(empty ID received)
Aug 27 18:11:07 plat5 /sbin/opensips[23569]:
ERROR:call_center:w_handle_call: failed to set new destination for
call
Aug 27 18:11:07 plat5 /sbin/opensips[23569]: errnum: -1
Error -1 means flowID is invalid, but I sent the same value on
both calls.
This is the call:
cc_handle_call("$rU")
I'm using Opensips 2.4.2 with Debian 8.11.
Am I missing something or found a bug?
Thanks
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Re: [OpenSIPS-Users] routing calls to several asterisks
thank you I'll give that a try
On Thu, Aug 30, 2018, 3:18 AM vasilevalex,
wrote:
> In the same situation I used dialplan and dynamic routing modules like
> this:
>
> # Get ID of destination Asterisk server according to CustomerID
> dp_translate("1", "$var(cust_id)/$var(dst_srv)");
> if ($var(dst_srv)==NULL) {
> exit;
> }
> # Set route for SIP according ID of Asterisk server from Dynamic routing
> gateways table
> route_to_gw("$var(dst_srv)");
> route(relay);
>
> Both modules load information from DB only during start time or via MI
> command. So all these requests are made from memory. I think this is more
> effective.
>
>
>
> --
> Sent from:
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>
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Re: [OpenSIPS-Users] routing calls to several asterisks
In the same situation I used dialplan and dynamic routing modules like this:
# Get ID of destination Asterisk server according to CustomerID
dp_translate("1", "$var(cust_id)/$var(dst_srv)");
if ($var(dst_srv)==NULL) {
exit;
}
# Set route for SIP according ID of Asterisk server from Dynamic routing
gateways table
route_to_gw("$var(dst_srv)");
route(relay);
Both modules load information from DB only during start time or via MI
command. So all these requests are made from memory. I think this is more
effective.
--
Sent from:
http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html
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