As I said, in the cc_flows, you have no value for the "message_queue"
column - this is a must, it has to be an URL to provide playback for the
call queuing.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Bootcamp 2018
http://opensips.org/training/OpenSIPS_Bootcamp_2018/
On 08/31/2018 04:06 PM, Daniel Zanutti wrote:
Hi Bogdan
Here it is table cc_flows:
id flowid priority skill prependcid message_welcome
message_queue
------ ------ -------- ------- ---------- ---------------
---------------
1 fila-1 256 suporte fila-1
Also table agents:
id agentid location logstate skills
last_call_end
------ ---------------------- -------------------------------
-------- ------- ---------------
1 [email protected] <mailto:[email protected]>
sip:[email protected]:5060 <http://sip:[email protected]:5060>
1 suporte 1535650312
Thanks
On Fri, Aug 31, 2018 at 5:02 AM Bogdan-Andrei Iancu
<[email protected] <mailto:[email protected]>> wrote:
Hi Daniel,
It is not about the B2B scenario, but about how you provisioned
the flow in DB. Could you simply dump the output of "select * from
cc_flows" ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Bootcamp 2018
http://opensips.org/training/OpenSIPS_Bootcamp_2018/
On 08/30/2018 08:34 PM, Daniel Zanutti wrote:
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:
<?xml version="1.0"?>
<scenario id="call center" name="Call center" param="1"
type="script">
<init>
<bridge>
<server>
<id>server1</id>
</server>
<client>
<id>client1</id>
<type>message</type>
<destination>
<value type="param">1</value>
</destination>
</client>
</bridge>
<state>1</state>
</init>
</scenario>
And SIP message is the same on all calls, just changed Call-id/tags:
U 10.10.10.10:5070 <http://10.10.10.10:5070> -> 10.10.10.10:5060
<http://10.10.10.10:5060>
INVITE sip:[email protected]:5060
<http://sip:[email protected]:5060> SIP/2.0.
Via: SIP/2.0/UDP 10.10.10.10:5070;branch=z9hG4bK2abb2acc;rport.
Max-Forwards: 70.
From: <sip:[email protected]:5070
<http://sip:[email protected]:5070>>;tag=as6440e239.
To: <sip:[email protected]:5060
<http://sip:[email protected]:5060>>.
Contact: <sip:[email protected]:5070
<http://sip:[email protected]:5070>>.
Call-ID: [email protected]:5070
<http://[email protected]: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: "551122223333" <sip:[email protected]
<mailto:sip%[email protected]>>.
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
<[email protected] <mailto:[email protected]>> 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
<[email protected] <mailto:[email protected]>>
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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