Re: [OpenSIPS-Users] opensips 2.1 call_center queue position
Hi Bogdan, Happy new year, I hope you are well? Did you see my reply below,? The count works now but its not routing the call. Many thanks Jon From: Jonathan Hunter <hunter...@hotmail.com> Sent: 23 December 2016 15:52 To: Bogdan-Andrei Iancu; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position Hi Bogdan, That seems to be better, certainly from the count incrementing, however a couple of issues, I am now seeing this error, and when first call hits opensips it doesnt now route to the Agent who is logged in free and available; : ERROR:b2b_entities:b2b_send_request: Can not send request [INVITE] for entity type [0] for dlg[0x7fe3f348a0f8]->[B2B.273.29] in terminated state Dec 23 15:49:22 HPBXProxy1-beta /usr/local/sbin/opensips[12140]: ERROR:b2b_logic:b2bl_bridge: Failed to send INVITE request Dec 23 15:49:22 HPBXProxy1-beta /usr/local/sbin/opensips[12140]: ERROR:call_center:set_call_leg: bridging failed Dec 23 15:49:22 HPBXProxy1-beta /usr/local/sbin/opensips[12140]: ERROR:call_center:b2bl_callback_customer: failed to set new destination for call Dec 23 15:49:22 HPBXProxy1-beta /usr/local/sbin/opensips[12140]: ERROR:b2b_logic:b2b_logic_notify_request: The callback function was unsuccessful Also the first member in the queue is given value 0, and the second is then given value 1, can it start at 1 or is there a reason behind it? Thanks Jon From: Bogdan-Andrei Iancu <bog...@opensips.org> Sent: 23 December 2016 14:14 To: Jonathan Hunter; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position Hi Jonathan, A Merry Christmas to you too ! I found a small mistake in the original patch I sent you. Please revert that one and use this new patch (see attachment). Let me know if it does the trick. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com [http://www.opensips-solutions.com/imgs/slideshow/slide3.jpg]<http://www.opensips-solutions.com/> Home — OpenSIPS Solutions<http://www.opensips-solutions.com/> www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 22.12.2016 13:58, Jonathan Hunter wrote: Hi Bogdan, Merry Christmas to you and the mailing list! I am testing the real queue position, and with one call in progress, and then 2 subsequent calls, both of which hit the hold music loop, I unfortunately dont see the position value increment, it remains at cc_pos=0 for all the calls, this is with just one queue defined to keep things very simple. Please let me know what debug or information you require, here are some outputs; opensipsctl fifo cc_list_calls Call:: 255.0 Ref=2 State=queued Call Time=2 Flow=Cust1 Call:: 884.0 Ref=2 State=queued Call Time=51 Flow=Cust1 Call:: 344.0 Ref=1 State=toagent Call Time=70 Flow=Cust1 Agent=2000 opensipsctl fifo cc_list_agents Agent:: 2000 Ref=1 Loged in=YES State=incall opensipsctl fifo cc_list_flows Flow:: Cust1 Avg Call Duration=199 Processed Calls=3 Logged Agents=1 Ongoing Calls=3 Ref=3 opensipsctl fifo cc_list_queue Call:: 0 Waiting for=114 ETW=0 Flow:: Cust1 Priority=256 Skill=custcare Call:: 1 Waiting for=65 ETW=199 Flow:: Cust1 Priority=256 Skill=custcare Many thanks Jon From: Bogdan-Andrei Iancu <bog...@opensips.org><mailto:bog...@opensips.org> Sent: 07 November 2016 20:41 To: Jonathan Hunter; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position Hi Jonathan, I was not able to test it with a real queue, so let me know if it really does the job (in terms of reporting the real position in the queue). Some questions: 1) currently I add that value only when sending to the queue / MOH - do you foresee any need to be added for other announcements like for welcome ? 2) will it be useful to add the ETW (estimate time to wait) ? is it useful ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and 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] opensips 2.1 call_center queue position
Hi Bogdan, That seems to be better, certainly from the count incrementing, however a couple of issues, I am now seeing this error, and when first call hits opensips it doesnt now route to the Agent who is logged in free and available; : ERROR:b2b_entities:b2b_send_request: Can not send request [INVITE] for entity type [0] for dlg[0x7fe3f348a0f8]->[B2B.273.29] in terminated state Dec 23 15:49:22 HPBXProxy1-beta /usr/local/sbin/opensips[12140]: ERROR:b2b_logic:b2bl_bridge: Failed to send INVITE request Dec 23 15:49:22 HPBXProxy1-beta /usr/local/sbin/opensips[12140]: ERROR:call_center:set_call_leg: bridging failed Dec 23 15:49:22 HPBXProxy1-beta /usr/local/sbin/opensips[12140]: ERROR:call_center:b2bl_callback_customer: failed to set new destination for call Dec 23 15:49:22 HPBXProxy1-beta /usr/local/sbin/opensips[12140]: ERROR:b2b_logic:b2b_logic_notify_request: The callback function was unsuccessful Also the first member in the queue is given value 0, and the second is then given value 1, can it start at 1 or is there a reason behind it? Thanks Jon From: Bogdan-Andrei Iancu <bog...@opensips.org> Sent: 23 December 2016 14:14 To: Jonathan Hunter; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position Hi Jonathan, A Merry Christmas to you too ! I found a small mistake in the original patch I sent you. Please revert that one and use this new patch (see attachment). Let me know if it does the trick. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com [http://www.opensips-solutions.com/imgs/slideshow/slide3.jpg]<http://www.opensips-solutions.com/> Home - OpenSIPS Solutions<http://www.opensips-solutions.com/> www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 22.12.2016 13:58, Jonathan Hunter wrote: Hi Bogdan, Merry Christmas to you and the mailing list! I am testing the real queue position, and with one call in progress, and then 2 subsequent calls, both of which hit the hold music loop, I unfortunately dont see the position value increment, it remains at cc_pos=0 for all the calls, this is with just one queue defined to keep things very simple. Please let me know what debug or information you require, here are some outputs; opensipsctl fifo cc_list_calls Call:: 255.0 Ref=2 State=queued Call Time=2 Flow=Cust1 Call:: 884.0 Ref=2 State=queued Call Time=51 Flow=Cust1 Call:: 344.0 Ref=1 State=toagent Call Time=70 Flow=Cust1 Agent=2000 opensipsctl fifo cc_list_agents Agent:: 2000 Ref=1 Loged in=YES State=incall opensipsctl fifo cc_list_flows Flow:: Cust1 Avg Call Duration=199 Processed Calls=3 Logged Agents=1 Ongoing Calls=3 Ref=3 opensipsctl fifo cc_list_queue Call:: 0 Waiting for=114 ETW=0 Flow:: Cust1 Priority=256 Skill=custcare Call:: 1 Waiting for=65 ETW=199 Flow:: Cust1 Priority=256 Skill=custcare Many thanks Jon From: Bogdan-Andrei Iancu <bog...@opensips.org><mailto:bog...@opensips.org> Sent: 07 November 2016 20:41 To: Jonathan Hunter; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position Hi Jonathan, I was not able to test it with a real queue, so let me know if it really does the job (in terms of reporting the real position in the queue). Some questions: 1) currently I add that value only when sending to the queue / MOH - do you foresee any need to be added for other announcements like for welcome ? 2) will it be useful to add the ETW (estimate time to wait) ? is it useful ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and 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] opensips 2.1 call_center queue position
Hi Jonathan,
A Merry Christmas to you too !
I found a small mistake in the original patch I sent you. Please revert
that one and use this new patch (see attachment). Let me know if it does
the trick.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 22.12.2016 13:58, Jonathan Hunter wrote:
Hi Bogdan,
Merry Christmas to you and the mailing list!
I am testing the real queue position, and with one call in progress,
and then 2 subsequent calls, both of which hit the hold music loop, I
unfortunately dont see the position value increment, it remains at
cc_pos=0 for all the calls, this is with just one queue defined to
keep things very simple.
Please let me know what debug or information you require, here are
some outputs;
opensipsctl fifo cc_list_calls
Call:: 255.0 Ref=2 State=queued Call Time=2 Flow=Cust1
Call:: 884.0 Ref=2 State=queued Call Time=51 Flow=Cust1
Call:: 344.0 Ref=1 State=toagent Call Time=70 Flow=Cust1 Agent=2000
opensipsctl fifo cc_list_agents
Agent:: 2000 Ref=1 Loged in=YES State=incall
opensipsctl fifo cc_list_flows
Flow:: Cust1 Avg Call Duration=199 Processed Calls=3 Logged Agents=1
Ongoing Calls=3 Ref=3
opensipsctl fifo cc_list_queue
Call:: 0 Waiting for=114 ETW=0
Flow:: Cust1 Priority=256 Skill=custcare
Call:: 1 Waiting for=65 ETW=199
Flow:: Cust1 Priority=256 Skill=custcare
Many thanks
Jon
*From:* Bogdan-Andrei Iancu <bog...@opensips.org>
*Sent:* 07 November 2016 20:41
*To:* Jonathan Hunter; OpenSIPS users mailling list
*Subject:* Re: [OpenSIPS-Users] opensips 2.1 call_center queue position
Hi Jonathan,
I was not able to test it with a real queue, so let me know if it
really does the job (in terms of reporting the real position in the
queue).
Some questions:
1) currently I add that value only when sending to the queue / MOH -
do you foresee any need to be added for other announcements like for
welcome ?
2) will it be useful to add the ETW (estimate time to wait) ? is it
useful ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
diff --git a/modules/call_center/cc_data.c b/modules/call_center/cc_data.c
index 45112e8..55af6ec 100755
--- a/modules/call_center/cc_data.c
+++ b/modules/call_center/cc_data.c
@@ -915,7 +915,7 @@ void print_queue(struct cc_data *data)
}
-void cc_queue_push_call(struct cc_data *data, struct cc_call *call, int top)
+int cc_queue_push_call(struct cc_data *data, struct cc_call *call, int top)
{
struct cc_call *call_it;
int n = 0;
@@ -971,6 +971,8 @@ void cc_queue_push_call(struct cc_data *data, struct cc_call *call, int top)
if (call->queue_start==0)
call->queue_start = get_ticks();
+
+ return data->queue.calls_no-1-n;
}
diff --git a/modules/call_center/cc_data.h b/modules/call_center/cc_data.h
index e2cf62c..14ebf37 100755
--- a/modules/call_center/cc_data.h
+++ b/modules/call_center/cc_data.h
@@ -227,7 +227,7 @@ void clean_cc_old_data(struct cc_data *data);
void clean_cc_unref_data(struct cc_data *data);
-void cc_queue_push_call(struct cc_data *data, struct cc_call *call, int top);
+int cc_queue_push_call(struct cc_data *data, struct cc_call *call, int top);
struct cc_call *cc_queue_pop_call_for_agent(struct cc_data *data,
struct cc_agent *agent);
diff --git a/modules/call_center/cc_queue.c b/modules/call_center/cc_queue.c
index 757bcf3..3956eb4 100644
--- a/modules/call_center/cc_queue.c
+++ b/modules/call_center/cc_queue.c
@@ -24,6 +24,7 @@
* 2014-03-17 initial version (bogdan)
*/
+#include "../../ut.h"
#include "cc_queue.h"
extern stat_var *stg_terminated_calls;
@@ -40,6 +41,7 @@ int cc_call_state_machine(struct cc_data *data, struct cc_call *call,
struct cc_agent *agent;
str *out = NULL;
int state =0;
+ int pos = -1;
switch (call->state) {
case CC_CALL_NONE:
@@ -81,7 +83,7 @@ int cc_call_state_machine(struct cc_data *data, struct cc_call *call,
break;
}
/* add it to queue */
-cc_queue_push_call( data, call, 0);
+pos = cc_queue_push_call( data, call, 0);
}
break;
case CC_CALL_TOAGENT:
@@ -94,10 +96,20 @@ int cc_call_state_machine(struct cc_data *data, struct cc_call *call,
}
if (out) {
- leg->s = (char*)pkg_malloc( out->len );
+ int l=0;
+ char *s;
+ if (pos>=0)
+ s=int2str((unsigned long)pos, );
+ leg->s = (char*)pkg_malloc( out->len + ((pos<0)?0:(8+l)) );
if (leg->s) {
leg->len = out->len;
memcpy(leg->s,out->s,out->len);
+ if (pos>=0) {
+memcpy(leg->s+leg->len, ";cc_pos=",8);
+leg->len += 8;
+memcpy(leg->s+leg->len, s, l);
+leg->len += l;
+ }
call->prev_state = call->state;
call->state = state;
return 0;
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips 2.1 call_center queue position
Hi Bogdan, Merry Christmas to you and the mailing list! I am testing the real queue position, and with one call in progress, and then 2 subsequent calls, both of which hit the hold music loop, I unfortunately dont see the position value increment, it remains at cc_pos=0 for all the calls, this is with just one queue defined to keep things very simple. Please let me know what debug or information you require, here are some outputs; opensipsctl fifo cc_list_calls Call:: 255.0 Ref=2 State=queued Call Time=2 Flow=Cust1 Call:: 884.0 Ref=2 State=queued Call Time=51 Flow=Cust1 Call:: 344.0 Ref=1 State=toagent Call Time=70 Flow=Cust1 Agent=2000 opensipsctl fifo cc_list_agents Agent:: 2000 Ref=1 Loged in=YES State=incall opensipsctl fifo cc_list_flows Flow:: Cust1 Avg Call Duration=199 Processed Calls=3 Logged Agents=1 Ongoing Calls=3 Ref=3 opensipsctl fifo cc_list_queue Call:: 0 Waiting for=114 ETW=0 Flow:: Cust1 Priority=256 Skill=custcare Call:: 1 Waiting for=65 ETW=199 Flow:: Cust1 Priority=256 Skill=custcare Many thanks Jon From: Bogdan-Andrei Iancu <bog...@opensips.org> Sent: 07 November 2016 20:41 To: Jonathan Hunter; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position Hi Jonathan, I was not able to test it with a real queue, so let me know if it really does the job (in terms of reporting the real position in the queue). Some questions: 1) currently I add that value only when sending to the queue / MOH - do you foresee any need to be added for other announcements like for welcome ? 2) will it be useful to add the ETW (estimate time to wait) ? is it useful ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and 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] opensips 2.1 call_center queue position
Hi Bogdan, Yes I will test with a real queue and let you know, hopefully I will have a test scenario up and running this week. In terms of your questions; For 1) For my needs I think I only require it for the MOH/queue announcement, however if its simple to add, I can test it out for sure, I think the more information available the better, certainly when looking to replace older systems such as a cisco which provide quite alot of information to the caller. And for 2), adding ETW would be useful for sure, as the more information options available the better when looking to implement a queue system like this for me. I will keep you posted with the testing. Thanks again for the patch! Jon From: Bogdan-Andrei Iancu <bog...@opensips.org> Sent: 07 November 2016 20:41 To: Jonathan Hunter; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position Hi Jonathan, I was not able to test it with a real queue, so let me know if it really does the job (in terms of reporting the real position in the queue). Some questions: 1) currently I add that value only when sending to the queue / MOH - do you foresee any need to be added for other announcements like for welcome ? 2) will it be useful to add the ETW (estimate time to wait) ? is it useful ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com Home — OpenSIPS Solutions<http://www.opensips-solutions.com/> www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 07.11.2016 22:35, Jonathan Hunter wrote: Hi Bogdan, Hope you are well. Yes that patch works, it stops the crash, and adds the parameter; Request-Line: INVITE sip:@1.2.3.4:5080;cc_pos=0<mailto:sip:@1.2.3.4:5080;cc_pos=0> SIP/2.0 Which is great! I will now get to work on the solution, thanks again. Jon From: Bogdan-Andrei Iancu <bog...@opensips.org><mailto:bog...@opensips.org> Sent: 07 November 2016 15:15 To: Jonathan Hunter; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position Hi Jonathan, Please revert the prev patch and try this new one - hopefully it will fix the crash. Thanks and regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com Home — OpenSIPS Solutions<http://www.opensips-solutions.com/> www.opensips-solutions.com<http://www.opensips-solutions.com> OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 06.11.2016 18:50, Jonathan Hunter wrote: Hi Bogdan, Sorry for the delay. I installed directly via make install, not via packages. Jon From: Bogdan-Andrei Iancu <bog...@opensips.org><mailto:bog...@opensips.org> Sent: 03 November 2016 10:39 To: Jonathan Hunter; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position Hi Jonathan, Have you installed OpenSIPS via packages ? or directly via "make install" ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com Home — OpenSIPS Solutions<http://www.opensips-solutions.com/> www.opensips-solutions.com<http://www.opensips-solutions.com> OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 02.11.2016 11:33, Jonathan Hunter wrote: Hi Bogdan, I am getting the core dumps, but containing no symbol tables, so I presume I need to recompile with debug flags enabled? Core was generated by `/usr/local/sbin/opensips -P /var/run/opensips.pid'. Program terminated with signal 11, Segmentation fault. #0 0x004ed7fb in ?? () "/core.24882" is a core file. Please specify an executable to debug. (gdb) bt full #0 0x004ed7fb in ?? () No symbol table info available. #1 0x7f6af7604468 in ?? () No symbol table info available. #2 0x001a in ?? () No symbol table info available. #3 0x in ?? () No symbol table info available. I installed 2.1 from sources, so whats the best way to do this? thanks Jon From: Bogdan-Andrei Iancu <bog...@opensips.org><mailto:bog...@opensips.org> Sent: 02 November 2016 08:09 To: Jonathan Hunter; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position For sure it is a patch issue. if you have a backtrace, it will useful. Thanks, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com Home — OpenSIPS Solutions<http://www.opensips-solutions.com/> www.opensips-solution
Re: [OpenSIPS-Users] opensips 2.1 call_center queue position
Hi Jonathan, Please revert the prev patch and try this new one - hopefully it will fix the crash. Thanks and regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 06.11.2016 18:50, Jonathan Hunter wrote: Hi Bogdan, Sorry for the delay. I installed directly via make install, not via packages. Jon *From:* Bogdan-Andrei Iancu <bog...@opensips.org> *Sent:* 03 November 2016 10:39 *To:* Jonathan Hunter; OpenSIPS users mailling list *Subject:* Re: [OpenSIPS-Users] opensips 2.1 call_center queue position Hi Jonathan, Have you installed OpenSIPS via packages ? or directly via "make install" ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com Home — OpenSIPS Solutions <http://www.opensips-solutions.com/> www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 02.11.2016 11:33, Jonathan Hunter wrote: Hi Bogdan, I am getting the core dumps, but containing no symbol tables, so I presume I need to recompile with debug flags enabled? Core was generated by `/usr/local/sbin/opensips -P /var/run/opensips.pid'. Program terminated with signal 11, Segmentation fault. #0 0x004ed7fb in ?? () "/core.24882" is a core file. Please specify an executable to debug. (gdb) bt full #0 0x004ed7fb in ?? () No symbol table info available. #1 0x7f6af7604468 in ?? () No symbol table info available. #2 0x001a in ?? () No symbol table info available. #3 0x in ?? () No symbol table info available. I installed 2.1 from sources, so whats the best way to do this? thanks Jon *From:* Bogdan-Andrei Iancu <bog...@opensips.org> *Sent:* 02 November 2016 08:09 *To:* Jonathan Hunter; OpenSIPS users mailling list *Subject:* Re: [OpenSIPS-Users] opensips 2.1 call_center queue position For sure it is a patch issue. if you have a backtrace, it will useful. Thanks, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com Home — OpenSIPS Solutions <http://www.opensips-solutions.com/> www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 02.11.2016 09:56, Jonathan Hunter wrote: Hi Bogdan, Thanks very much for this. I have just applied patch (installed from sources so when to call_center module directory and ran patch < call_center_pos.patch) then did a recompile. However when I now route to the call center (cc_handle_call) it generates a core and kills opensips; user 2000 has Callqueue set so send to Call Queue Route Nov 2 07:53:42 HPBXProxy1-beta /sbin/opensips[21141]: NOTICE:core:io_wait_loop_epoll: EPOLLIN(read) event: epollwait() set event EPOLLHUP - connection closed by the remote peer! Nov 2 07:53:42 HPBXProxy1-beta /sbin/opensips[21141]: CRITICAL:core:receive_fd: EOF on 19 Nov 2 07:53:42 HPBXProxy1-beta /sbin/opensips[21112]: INFO:core:handle_sigs: child process 21119 exited by a signal 11 Nov 2 07:53:42 HPBXProxy1-beta /sbin/opensips[21112]: INFO:core:handle_sigs: core was generated Nov 2 07:53:42 HPBXProxy1-beta /sbin/opensips[21112]: INFO:core:handle_sigs: terminating due to SIGCHLD Do you need me to backtrace/debug through to get the issue? Or is problem how I applied patch? Many thanks Jon *From:* Bogdan-Andrei Iancu <bog...@opensips.org> *Sent:* 01 November 2016 21:44 *To:* Jonathan Hunter; OpenSIPS users mailling list *Subject:* Re: [OpenSIPS-Users] opensips 2.1 call_center queue position Hi Jonathan, Please give it a try to this patch - it is not really tested, but when the call is sent the Queue announcement, it should have a ";cc_pos=xxx" parameter giving the position is the queue (0 being the first to be dispatched to agents). Let me know if it works. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com Home — OpenSIPS Solutions <http://www.opensips-solutions.com/> www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 28.10.2016 15:59, Jonathan Hunter wrote: Hi Bogdan, Great news, really do appreciate that. Many thanks Jon *From:* Bogdan-Andrei Iancu <bog...@opensips.org> *Sent:* 28 October 2016 12:48 *To:* Jonathan Hunter; OpenSIPS users mailling list *Subject:*
Re: [OpenSIPS-Users] opensips 2.1 call_center queue position
Hi Bogdan, Sorry for the delay. I installed directly via make install, not via packages. Jon From: Bogdan-Andrei Iancu <bog...@opensips.org> Sent: 03 November 2016 10:39 To: Jonathan Hunter; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position Hi Jonathan, Have you installed OpenSIPS via packages ? or directly via "make install" ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com Home — OpenSIPS Solutions<http://www.opensips-solutions.com/> www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 02.11.2016 11:33, Jonathan Hunter wrote: Hi Bogdan, I am getting the core dumps, but containing no symbol tables, so I presume I need to recompile with debug flags enabled? Core was generated by `/usr/local/sbin/opensips -P /var/run/opensips.pid'. Program terminated with signal 11, Segmentation fault. #0 0x004ed7fb in ?? () "/core.24882" is a core file. Please specify an executable to debug. (gdb) bt full #0 0x004ed7fb in ?? () No symbol table info available. #1 0x7f6af7604468 in ?? () No symbol table info available. #2 0x001a in ?? () No symbol table info available. #3 0x in ?? () No symbol table info available. I installed 2.1 from sources, so whats the best way to do this? thanks Jon From: Bogdan-Andrei Iancu <bog...@opensips.org><mailto:bog...@opensips.org> Sent: 02 November 2016 08:09 To: Jonathan Hunter; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position For sure it is a patch issue. if you have a backtrace, it will useful. Thanks, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com Home — OpenSIPS Solutions<http://www.opensips-solutions.com/> www.opensips-solutions.com<http://www.opensips-solutions.com> OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 02.11.2016 09:56, Jonathan Hunter wrote: Hi Bogdan, Thanks very much for this. I have just applied patch (installed from sources so when to call_center module directory and ran patch < call_center_pos.patch) then did a recompile. However when I now route to the call center (cc_handle_call) it generates a core and kills opensips; user 2000 has Callqueue set so send to Call Queue Route Nov 2 07:53:42 HPBXProxy1-beta /sbin/opensips[21141]: NOTICE:core:io_wait_loop_epoll: EPOLLIN(read) event: epollwait() set event EPOLLHUP - connection closed by the remote peer! Nov 2 07:53:42 HPBXProxy1-beta /sbin/opensips[21141]: CRITICAL:core:receive_fd: EOF on 19 Nov 2 07:53:42 HPBXProxy1-beta /sbin/opensips[21112]: INFO:core:handle_sigs: child process 21119 exited by a signal 11 Nov 2 07:53:42 HPBXProxy1-beta /sbin/opensips[21112]: INFO:core:handle_sigs: core was generated Nov 2 07:53:42 HPBXProxy1-beta /sbin/opensips[21112]: INFO:core:handle_sigs: terminating due to SIGCHLD Do you need me to backtrace/debug through to get the issue? Or is problem how I applied patch? Many thanks Jon From: Bogdan-Andrei Iancu <bog...@opensips.org><mailto:bog...@opensips.org> Sent: 01 November 2016 21:44 To: Jonathan Hunter; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position Hi Jonathan, Please give it a try to this patch - it is not really tested, but when the call is sent the Queue announcement, it should have a ";cc_pos=xxx" parameter giving the position is the queue (0 being the first to be dispatched to agents). Let me know if it works. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com Home — OpenSIPS Solutions<http://www.opensips-solutions.com/> www.opensips-solutions.com<http://www.opensips-solutions.com> OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 28.10.2016 15:59, Jonathan Hunter wrote: Hi Bogdan, Great news, really do appreciate that. Many thanks Jon From: Bogdan-Andrei Iancu <bog...@opensips.org><mailto:bog...@opensips.org> Sent: 28 October 2016 12:48 To: Jonathan Hunter; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position Hi Jonathan, No, it is no yet available. Give me couple of days and I will make a patch for it. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com Home — OpenSIPS Solutions&l
Re: [OpenSIPS-Users] opensips 2.1 call_center queue position (Jonathan Hunter)
Hello I recently started to explore call_center module and I think that it might not be patch related. here is my experience. I downloaded opensips branch 2.2 commit c8a605b. when agent is registered and there are no clients in line/queue the caller gets connected to the agent almost instantly and everything works fine. If the agent is in wrapup_time and I call again the same flow/queue I get send to the media server and moh starts streaming. Now when the wrapup_time finishes and the caller needs to be connected to the agent or somewhere earlier before that opensips crashes and caller is stuck in moh stream. but if there is no call during wrapup_time period and I call again everything works fine. tested with 1 agent in 2 flows, 1 user and asterisk as media server to stream moh sorry if I high jacket this thread but I think it's relevant to the crash Jonathan Hunter is experiencing. == last few log lines from the crash Nov 2 08:01:12 opensips /sbin/opensips[12152]: b2b_reply BYE : [F=sip:use...@opensips.test R= D=UA= IP=(192.168.116.125:5060 192.168.116.125:5060) ID=B2B.323.7879765.1478070026] Nov 2 08:01:20 opensips /sbin/opensips[12151]: CALLCENTER MODULE: [F=sip:use...@opensips.test R=sip:engl...@opensips.test D= M=INVITE IP=(192.168.69.114:39336 192.168.116.125:5060) ID=zSWAmB.rM89XJB-wKMDABguDJ21SKN3Q] Nov 2 08:01:20 opensips /sbin/opensips[12151]: INFO:b2b_logic:b2bl_add_client: adding entity [0xb33c9c4c]->[B2B.464.2920496.1478070079] to tuple [0xb33c9d40]->[13.0] Nov 2 08:01:20 opensips /sbin/opensips[12152]: INFO:b2b_logic:b2b_add_dlginfo: Dialog pair: [zSWAmB.rM89XJB-wKMDABguDJ21SKN3Q] - [B2B.464.2920496.1478070079] Nov 2 08:01:20 opensips /sbin/opensips[12152]: b2b_reply INVITE : [F=sip:use...@opensips.test R= D= UA= IP=(192.168.117.100:5065 192.168.116.125:5060) ID=B2B.464.2920496.1478070079] Nov 2 08:01:21 opensips /sbin/opensips[12152]: b2b_request ACK : [F=sip:use...@opensips.test R=sip:192.168.116.125:5060 D= UA= IP=(192.168.69.114:39336 192.168.116.125:5060) ID=zSWAmB.rM89XJB-wKMDABguDJ21SKN3Q] Nov 2 08:01:25 opensips /sbin/opensips[12152]: b2b_request INVITE : [F=sip:use...@opensips.test R=sip:192.168.116.125:5060 D= UA= IP=(192.168.69.114:39336 192.168.116.125:5060) ID=zSWAmB.rM89XJB-wKMDABguDJ21SKN3Q] Nov 2 08:01:26 opensips /sbin/opensips[12124]: INFO:core:handle_sigs: child process 12152 exited by a signal 11 Nov 2 08:01:26 opensips /sbin/opensips[12181]: CRITICAL:core:receive_fd: EOF on 34 Nov 2 08:01:26 opensips /sbin/opensips[12124]: INFO:core:handle_sigs: core was generated Nov 2 08:01:26 opensips /sbin/opensips[12124]: INFO:core:handle_sigs: terminating due to SIGCHLD Nov 2 08:01:26 opensips /sbin/opensips[12181]: INFO:core:sig_usr: signal 15 received Nov 2 08:01:26 opensips /sbin/opensips[12173]: INFO:core:sig_usr: signal 15 received On 11/02/2016 10:33 AM, users-requ...@lists.opensips.org wrote: Re: opensips 2.1 call_center queue position (Jonathan Hunter) ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips 2.1 call_center queue position
Hi Jonathan, Please give it a try to this patch - it is not really tested, but when the call is sent the Queue announcement, it should have a ";cc_pos=xxx" parameter giving the position is the queue (0 being the first to be dispatched to agents). Let me know if it works. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 28.10.2016 15:59, Jonathan Hunter wrote: Hi Bogdan, Great news, really do appreciate that. Many thanks Jon *From:* Bogdan-Andrei Iancu <bog...@opensips.org> *Sent:* 28 October 2016 12:48 *To:* Jonathan Hunter; OpenSIPS users mailling list *Subject:* Re: [OpenSIPS-Users] opensips 2.1 call_center queue position Hi Jonathan, No, it is no yet available. Give me couple of days and I will make a patch for it. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com Home — OpenSIPS Solutions <http://www.opensips-solutions.com/> www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 25.10.2016 19:22, Jonathan Hunter wrote: Hi Bogdan, Sorry cant recall If I replied to this. Is cc_pos available now to extract from the module? Thats the only thing I need then I can implement call center which I think will be much more scale-able than the other approach I am using with FreeSWITCH, I would use that just for announcements. Any response/help appreciated. Jon *From:* Bogdan-Andrei Iancu <bog...@opensips.org> *Sent:* 13 October 2016 10:59 *To:* Jonathan Hunter; OpenSIPS users mailling list *Subject:* Re: [OpenSIPS-Users] opensips 2.1 call_center queue position Hi Jonathan, No, currently this is not possible. I was trying to envision a solution for your need. But, checking the code, it is really difficult to add the headers to the INVITEs originated by OpenSIPS (via the B2BUA), as we need some flexibility (different headers to different INVITEs belonging to the same B2B scenario , and even more, we need to traverse couple of internal APIs - to propagate the hdrs from Call center module all the way to TM). So, a simpler approach may be to add such extra info as URI params to the RURI. Like if you have the RURI "sip:queue@192.168.1.10:5060" for the queue/waiting playback, the RURI in the INVITE to the media server will look like : sip:queue@192.168.1.10:5060;cc_eta=40;cc_pos=10 - cc_eta being the estimated time to wait in seconds and cc_pos the position in the queue. What do you think of this ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12.10.2016 17:21, Jonathan Hunter wrote: Hi Bogdan, Yes being able to grab the queue position would be perfect. Is that possible? Thanks Jon -------- Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position To: hunter...@hotmail.com; users@lists.opensips.org From: bog...@opensips.org Date: Wed, 12 Oct 2016 15:42:43 +0300 Hi Jonathan, When a call is mapped to a flow / queue (before playing the welcome message), we know the ETA (estimated time to wait) and when is placed in the queue (before playing the queuing) we internally know the position in the queue. Would it help to have the position in the queue placed into a custome SIP header, when sending the INVITE to the message_queue URL ? or to the welcome message ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12.10.2016 12:06, Jonathan Hunter wrote: Hello Bogdan, Thanks for the response. In terms of my question, with a number of queuing platforms, they have the capability to tell the caller, what position they are in , and when they are likely to be answered. I just wondered if this logic was already within the module, or if I would need to use an external code/script to facilitate this function? As I presume call_center tracks the number of calls currently in a queue ? I would then want to be able to extract that information, and if a caller was for example in 3rd place in a queue, I could inject the relevant audio from freeswitch to tell them their current position? Does that make sense? :) Just wanted to know if its something this module can do? Thanks Jon -------- Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position To: users@lists.opensips.org <mailto:users@lists.opensips.org>; hunter...@hotmail.com <mailto:hunter...@hotmail.com> From: bog...@opensips.org <mailto:bog
Re: [OpenSIPS-Users] opensips 2.1 call_center queue position
Hi Bogdan, Great news, really do appreciate that. Many thanks Jon From: Bogdan-Andrei Iancu <bog...@opensips.org> Sent: 28 October 2016 12:48 To: Jonathan Hunter; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position Hi Jonathan, No, it is no yet available. Give me couple of days and I will make a patch for it. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com Home — OpenSIPS Solutions<http://www.opensips-solutions.com/> www.opensips-solutions.com OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. On 25.10.2016 19:22, Jonathan Hunter wrote: Hi Bogdan, Sorry cant recall If I replied to this. Is cc_pos available now to extract from the module? Thats the only thing I need then I can implement call center which I think will be much more scale-able than the other approach I am using with FreeSWITCH, I would use that just for announcements. Any response/help appreciated. Jon From: Bogdan-Andrei Iancu <bog...@opensips.org><mailto:bog...@opensips.org> Sent: 13 October 2016 10:59 To: Jonathan Hunter; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position Hi Jonathan, No, currently this is not possible. I was trying to envision a solution for your need. But, checking the code, it is really difficult to add the headers to the INVITEs originated by OpenSIPS (via the B2BUA), as we need some flexibility (different headers to different INVITEs belonging to the same B2B scenario , and even more, we need to traverse couple of internal APIs - to propagate the hdrs from Call center module all the way to TM). So, a simpler approach may be to add such extra info as URI params to the RURI. Like if you have the RURI "sip:queue@192.168.1.10:5060"<mailto:sip:queue@192.168.1.10:5060> for the queue/waiting playback, the RURI in the INVITE to the media server will look like : sip:queue@192.168.1.10:5060;cc_eta=40;cc_pos=10<mailto:sip:queue@192.168.1.10:5060;cc_eta=40;cc_pos=10> - cc_eta being the estimated time to wait in seconds and cc_pos the position in the queue. What do you think of this ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12.10.2016 17:21, Jonathan Hunter wrote: Hi Bogdan, Yes being able to grab the queue position would be perfect. Is that possible? Thanks Jon ____________ Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position To: hunter...@hotmail.com<mailto:hunter...@hotmail.com>; users@lists.opensips.org<mailto:users@lists.opensips.org> From: bog...@opensips.org<mailto:bog...@opensips.org> Date: Wed, 12 Oct 2016 15:42:43 +0300 Hi Jonathan, When a call is mapped to a flow / queue (before playing the welcome message), we know the ETA (estimated time to wait) and when is placed in the queue (before playing the queuing) we internally know the position in the queue. Would it help to have the position in the queue placed into a custome SIP header, when sending the INVITE to the message_queue URL ? or to the welcome message ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12.10.2016 12:06, Jonathan Hunter wrote: Hello Bogdan, Thanks for the response. In terms of my question, with a number of queuing platforms, they have the capability to tell the caller, what position they are in , and when they are likely to be answered. I just wondered if this logic was already within the module, or if I would need to use an external code/script to facilitate this function? As I presume call_center tracks the number of calls currently in a queue ? I would then want to be able to extract that information, and if a caller was for example in 3rd place in a queue, I could inject the relevant audio from freeswitch to tell them their current position? Does that make sense? :) Just wanted to know if its something this module can do? Thanks Jon ________________ Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position To: users@lists.opensips.org<mailto:users@lists.opensips.org>; hunter...@hotmail.com<mailto:hunter...@hotmail.com> From: bog...@opensips.org<mailto:bog...@opensips.org> Date: Wed, 12 Oct 2016 11:23:45 +0300 Hello Jon, The message_queue is a SIP URI pointing to an audio announcement to play to roll of the waiting/in-queue playback. This needs to be an announcements that never ends (from the perspective of the media server); only the the OpenSIPS Queue may terminate the playback, when it decides to take out the call from waiting and to deliver it to an agent. As for your question,
Re: [OpenSIPS-Users] opensips 2.1 call_center queue position
Hi Jonathan,
No, it is no yet available. Give me couple of days and I will make a
patch for it.
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 25.10.2016 19:22, Jonathan Hunter wrote:
Hi Bogdan,
Sorry cant recall If I replied to this.
Is cc_pos available now to extract from the module?
Thats the only thing I need then I can implement call center which I
think will be much more scale-able than the other approach I am using
with FreeSWITCH, I would use that just for announcements.
Any response/help appreciated.
Jon
*From:* Bogdan-Andrei Iancu <bog...@opensips.org>
*Sent:* 13 October 2016 10:59
*To:* Jonathan Hunter; OpenSIPS users mailling list
*Subject:* Re: [OpenSIPS-Users] opensips 2.1 call_center queue position
Hi Jonathan,
No, currently this is not possible. I was trying to envision a
solution for your need.
But, checking the code, it is really difficult to add the headers to
the INVITEs originated by OpenSIPS (via the B2BUA), as we need some
flexibility (different headers to different INVITEs belonging to the
same B2B scenario , and even more, we need to traverse couple of
internal APIs - to propagate the hdrs from Call center module all the
way to TM).
So, a simpler approach may be to add such extra info as URI params to
the RURI. Like if you have the RURI "sip:queue@192.168.1.10:5060" for
the queue/waiting playback, the RURI in the INVITE to the media server
will look like : sip:queue@192.168.1.10:5060;cc_eta=40;cc_pos=10 -
cc_eta being the estimated time to wait in seconds and cc_pos the
position in the queue.
What do you think of this ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 12.10.2016 17:21, Jonathan Hunter wrote:
Hi Bogdan,
Yes being able to grab the queue position would be perfect.
Is that possible?
Thanks
Jon
----
Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position
To: hunter...@hotmail.com; users@lists.opensips.org
From: bog...@opensips.org
Date: Wed, 12 Oct 2016 15:42:43 +0300
Hi Jonathan,
When a call is mapped to a flow / queue (before playing the welcome
message), we know the ETA (estimated time to wait) and when is placed
in the queue (before playing the queuing) we internally know the
position in the queue.
Would it help to have the position in the queue placed into a custome
SIP header, when sending the INVITE to the message_queue URL ? or to
the welcome message ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 12.10.2016 12:06, Jonathan Hunter wrote:
Hello Bogdan,
Thanks for the response.
In terms of my question, with a number of queuing platforms, they
have the capability to tell the caller, what position they are in
, and when they are likely to be answered.
I just wondered if this logic was already within the module, or
if I would need to use an external code/script to facilitate this
function?
As I presume call_center tracks the number of calls currently in
a queue ? I would then want to be able to extract that
information, and if a caller was for example in 3rd place in a
queue, I could inject the relevant audio from freeswitch to tell
them their current position?
Does that make sense? :) Just wanted to know if its something
this module can do?
Thanks
Jon
Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position
To: users@lists.opensips.org <mailto:users@lists.opensips.org>;
hunter...@hotmail.com <mailto:hunter...@hotmail.com>
From: bog...@opensips.org <mailto:bog...@opensips.org>
Date: Wed, 12 Oct 2016 11:23:45 +0300
Hello Jon,
The message_queue is a SIP URI pointing to an audio announcement
to play to roll of the waiting/in-queue playback. This needs to
be an announcements that never ends (from the perspective of the
media server); only the the OpenSIPS Queue may terminate the
playback, when it decides to take out the call from waiting and
to deliver it to an agent.
As for your question, I'm not sure I understand what you mean by
"inject a message with queue position for the caller in question"
- could you detail please ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 11.10.2016 13:36, Jonathan Hunter wrote:
Hi guys,
I have implemented an opensips/freeswitch environment, and I
wish to add call queues to it, and I like the look of
call_center, so just checking this out in comparison to
mod_callcenter in FS world.
Re: [OpenSIPS-Users] opensips 2.1 call_center queue position
Hi Bogdan, Sorry cant recall If I replied to this. Is cc_pos available now to extract from the module? Thats the only thing I need then I can implement call center which I think will be much more scale-able than the other approach I am using with FreeSWITCH, I would use that just for announcements. Any response/help appreciated. Jon From: Bogdan-Andrei Iancu <bog...@opensips.org> Sent: 13 October 2016 10:59 To: Jonathan Hunter; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position Hi Jonathan, No, currently this is not possible. I was trying to envision a solution for your need. But, checking the code, it is really difficult to add the headers to the INVITEs originated by OpenSIPS (via the B2BUA), as we need some flexibility (different headers to different INVITEs belonging to the same B2B scenario , and even more, we need to traverse couple of internal APIs - to propagate the hdrs from Call center module all the way to TM). So, a simpler approach may be to add such extra info as URI params to the RURI. Like if you have the RURI "sip:queue@192.168.1.10:5060"<mailto:sip:queue@192.168.1.10:5060> for the queue/waiting playback, the RURI in the INVITE to the media server will look like : sip:queue@192.168.1.10:5060;cc_eta=40;cc_pos=10<mailto:sip:queue@192.168.1.10:5060;cc_eta=40;cc_pos=10> - cc_eta being the estimated time to wait in seconds and cc_pos the position in the queue. What do you think of this ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12.10.2016 17:21, Jonathan Hunter wrote: Hi Bogdan, Yes being able to grab the queue position would be perfect. Is that possible? Thanks Jon ____ Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position To: hunter...@hotmail.com<mailto:hunter...@hotmail.com>; users@lists.opensips.org<mailto:users@lists.opensips.org> From: bog...@opensips.org<mailto:bog...@opensips.org> Date: Wed, 12 Oct 2016 15:42:43 +0300 Hi Jonathan, When a call is mapped to a flow / queue (before playing the welcome message), we know the ETA (estimated time to wait) and when is placed in the queue (before playing the queuing) we internally know the position in the queue. Would it help to have the position in the queue placed into a custome SIP header, when sending the INVITE to the message_queue URL ? or to the welcome message ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12.10.2016 12:06, Jonathan Hunter wrote: Hello Bogdan, Thanks for the response. In terms of my question, with a number of queuing platforms, they have the capability to tell the caller, what position they are in , and when they are likely to be answered. I just wondered if this logic was already within the module, or if I would need to use an external code/script to facilitate this function? As I presume call_center tracks the number of calls currently in a queue ? I would then want to be able to extract that information, and if a caller was for example in 3rd place in a queue, I could inject the relevant audio from freeswitch to tell them their current position? Does that make sense? :) Just wanted to know if its something this module can do? Thanks Jon ____________ Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position To: users@lists.opensips.org<mailto:users@lists.opensips.org>; hunter...@hotmail.com<mailto:hunter...@hotmail.com> From: bog...@opensips.org<mailto:bog...@opensips.org> Date: Wed, 12 Oct 2016 11:23:45 +0300 Hello Jon, The message_queue is a SIP URI pointing to an audio announcement to play to roll of the waiting/in-queue playback. This needs to be an announcements that never ends (from the perspective of the media server); only the the OpenSIPS Queue may terminate the playback, when it decides to take out the call from waiting and to deliver it to an agent. As for your question, I'm not sure I understand what you mean by "inject a message with queue position for the caller in question" - could you detail please ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11.10.2016 13:36, Jonathan Hunter wrote: Hi guys, I have implemented an opensips/freeswitch environment, and I wish to add call queues to it, and I like the look of call_center, so just checking this out in comparison to mod_callcenter in FS world. My main question is if using the call_center module if you can inject a message with queue position for the caller in question, as I cant see that in documentation, I only see message_queue which I assume could be used to report the callers position, but just wondered if anyone has done this and if they could give me some tips as to if possible?
Re: [OpenSIPS-Users] opensips 2.1 call_center queue position
Hi Bogdan,
Thanks for the response, and yes I understand you are envisioning a solution,
which I appreciate as usual :)
Yes the addition of cc_pos would work for me, as I can extract any number of
variables form the ruri, this would do the job.
Thanks
Jon
Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position
To: hunter...@hotmail.com; users@lists.opensips.org
From: bog...@opensips.org
Date: Thu, 13 Oct 2016 13:59:40 +0300
Hi Jonathan,
No, currently this is not possible. I was trying to envision a solution
for your need.
But, checking the code, it is really difficult to add the headers
to the INVITEs originated by OpenSIPS (via the B2BUA), as we need
some flexibility (different headers to different INVITEs belonging
to the same B2B scenario , and even more, we need to traverse
couple of internal APIs - to propagate the hdrs from Call center
module all the way to TM).
So, a simpler approach may be to add such extra info as URI params
to the RURI. Like if you have the RURI
"sip:queue@192.168.1.10:5060" for the queue/waiting playback, the
RURI in the INVITE to the media server will look like :
sip:queue@192.168.1.10:5060;cc_eta=40;cc_pos=10
- cc_eta being the estimated time to wait in seconds and cc_pos
the position in the queue.
What do you think of this ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 12.10.2016 17:21, Jonathan Hunter
wrote:
Hi Bogdan,
Yes being able to grab the queue position would be perfect.
Is that possible?
Thanks
Jon
Subject: Re: [OpenSIPS-Users] opensips
2.1 call_center queue position
To: hunter...@hotmail.com; users@lists.opensips.org
From: bog...@opensips.org
Date: Wed, 12 Oct 2016 15:42:43 +0300
Hi Jonathan,
When a call is mapped to a flow / queue (before playing
the welcome message), we know the ETA (estimated time to
wait) and when is placed in the queue (before playing the
queuing) we internally know the position in the queue.
Would it help to have the position in the queue placed
into a custome SIP header, when sending the INVITE to the
message_queue URL ? or to the welcome message ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 12.10.2016 12:06,
Jonathan Hunter wrote:
Hello Bogdan,
Thanks for the response.
In terms of my question, with a number of queuing
platforms, they have the capability to tell the
caller, what position they are in , and when they are
likely to be answered.
I just wondered if this logic was already within
the module, or if I would need to use an external
code/script to facilitate this function?
As I presume call_center tracks the number of calls
currently in a queue ? I would then want to be able to
extract that information, and if a caller was for
example in 3rd place in a queue, I could inject the
relevant audio from freeswitch to tell them their
current position?
Does that make sense? :) Just wanted to know if
its something this module can do?
Thanks
Jon
Subject: Re:
[OpenSIPS-Users] opensips 2.1 call_center queue
position
To: users@lists.opensips.org;
hunter...@hotmail.com
From: bog...@opensips.org
Date: Wed, 12 Oct 2016 11:23:45 +0300
Hello Jon,
The message_queue is a SIP URI pointing to an
audio announcement to play to roll of the
waiting/in-queue playback. This needs to be an
announcements that never ends (from the
perspective of
Re: [OpenSIPS-Users] opensips 2.1 call_center queue position
Hi Jonathan, No, currently this is not possible. I was trying to envision a solution for your need. But, checking the code, it is really difficult to add the headers to the INVITEs originated by OpenSIPS (via the B2BUA), as we need some flexibility (different headers to different INVITEs belonging to the same B2B scenario , and even more, we need to traverse couple of internal APIs - to propagate the hdrs from Call center module all the way to TM). So, a simpler approach may be to add such extra info as URI params to the RURI. Like if you have the RURI "sip:queue@192.168.1.10:5060" for the queue/waiting playback, the RURI in the INVITE to the media server will look like : sip:queue@192.168.1.10:5060;cc_eta=40;cc_pos=10 - cc_eta being the estimated time to wait in seconds and cc_pos the position in the queue. What do you think of this ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12.10.2016 17:21, Jonathan Hunter wrote: Hi Bogdan, Yes being able to grab the queue position would be perfect. Is that possible? Thanks Jon Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position To: hunter...@hotmail.com; users@lists.opensips.org From: bog...@opensips.org Date: Wed, 12 Oct 2016 15:42:43 +0300 Hi Jonathan, When a call is mapped to a flow / queue (before playing the welcome message), we know the ETA (estimated time to wait) and when is placed in the queue (before playing the queuing) we internally know the position in the queue. Would it help to have the position in the queue placed into a custome SIP header, when sending the INVITE to the message_queue URL ? or to the welcome message ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12.10.2016 12:06, Jonathan Hunter wrote: Hello Bogdan, Thanks for the response. In terms of my question, with a number of queuing platforms, they have the capability to tell the caller, what position they are in , and when they are likely to be answered. I just wondered if this logic was already within the module, or if I would need to use an external code/script to facilitate this function? As I presume call_center tracks the number of calls currently in a queue ? I would then want to be able to extract that information, and if a caller was for example in 3rd place in a queue, I could inject the relevant audio from freeswitch to tell them their current position? Does that make sense? :) Just wanted to know if its something this module can do? Thanks Jon Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position To: users@lists.opensips.org <mailto:users@lists.opensips.org>; hunter...@hotmail.com <mailto:hunter...@hotmail.com> From: bog...@opensips.org <mailto:bog...@opensips.org> Date: Wed, 12 Oct 2016 11:23:45 +0300 Hello Jon, The message_queue is a SIP URI pointing to an audio announcement to play to roll of the waiting/in-queue playback. This needs to be an announcements that never ends (from the perspective of the media server); only the the OpenSIPS Queue may terminate the playback, when it decides to take out the call from waiting and to deliver it to an agent. As for your question, I'm not sure I understand what you mean by "inject a message with queue position for the caller in question" - could you detail please ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11.10.2016 13:36, Jonathan Hunter wrote: Hi guys, I have implemented an opensips/freeswitch environment, and I wish to add call queues to it, and I like the look of call_center, so just checking this out in comparison to mod_callcenter in FS world. My main question is if using the call_center module if you can inject a message with queue position for the caller in question, as I cant see that in documentation, I only see message_queue which I assume could be used to report the callers position, but just wondered if anyone has done this and if they could give me some tips as to if possible? Many thanks Jon ___ 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
Re: [OpenSIPS-Users] opensips 2.1 call_center queue position
Hi Bogdan, Yes being able to grab the queue position would be perfect. Is that possible? Thanks Jon Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position To: hunter...@hotmail.com; users@lists.opensips.org From: bog...@opensips.org Date: Wed, 12 Oct 2016 15:42:43 +0300 Hi Jonathan, When a call is mapped to a flow / queue (before playing the welcome message), we know the ETA (estimated time to wait) and when is placed in the queue (before playing the queuing) we internally know the position in the queue. Would it help to have the position in the queue placed into a custome SIP header, when sending the INVITE to the message_queue URL ? or to the welcome message ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12.10.2016 12:06, Jonathan Hunter wrote: Hello Bogdan, Thanks for the response. In terms of my question, with a number of queuing platforms, they have the capability to tell the caller, what position they are in , and when they are likely to be answered. I just wondered if this logic was already within the module, or if I would need to use an external code/script to facilitate this function? As I presume call_center tracks the number of calls currently in a queue ? I would then want to be able to extract that information, and if a caller was for example in 3rd place in a queue, I could inject the relevant audio from freeswitch to tell them their current position? Does that make sense? :) Just wanted to know if its something this module can do? Thanks Jon Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position To: users@lists.opensips.org; hunter...@hotmail.com From: bog...@opensips.org Date: Wed, 12 Oct 2016 11:23:45 +0300 Hello Jon, The message_queue is a SIP URI pointing to an audio announcement to play to roll of the waiting/in-queue playback. This needs to be an announcements that never ends (from the perspective of the media server); only the the OpenSIPS Queue may terminate the playback, when it decides to take out the call from waiting and to deliver it to an agent. As for your question, I'm not sure I understand what you mean by "inject a message with queue position for the caller in question" - could you detail please ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11.10.2016 13:36, Jonathan Hunter wrote: Hi guys, I have implemented an opensips/freeswitch environment, and I wish to add call queues to it, and I like the look of call_center, so just checking this out in comparison to mod_callcenter in FS world. My main question is if using the call_center module if you can inject a message with queue position for the caller in question, as I cant see that in documentation, I only see message_queue which I assume could be used to report the callers position, but just wondered if anyone has done this and if they could give me some tips as to if possible? Many thanks Jon ___ 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] opensips 2.1 call_center queue position
Hi Jonathan, When a call is mapped to a flow / queue (before playing the welcome message), we know the ETA (estimated time to wait) and when is placed in the queue (before playing the queuing) we internally know the position in the queue. Would it help to have the position in the queue placed into a custome SIP header, when sending the INVITE to the message_queue URL ? or to the welcome message ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12.10.2016 12:06, Jonathan Hunter wrote: Hello Bogdan, Thanks for the response. In terms of my question, with a number of queuing platforms, they have the capability to tell the caller, what position they are in , and when they are likely to be answered. I just wondered if this logic was already within the module, or if I would need to use an external code/script to facilitate this function? As I presume call_center tracks the number of calls currently in a queue ? I would then want to be able to extract that information, and if a caller was for example in 3rd place in a queue, I could inject the relevant audio from freeswitch to tell them their current position? Does that make sense? :) Just wanted to know if its something this module can do? Thanks Jon Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position To: users@lists.opensips.org; hunter...@hotmail.com From: bog...@opensips.org Date: Wed, 12 Oct 2016 11:23:45 +0300 Hello Jon, The message_queue is a SIP URI pointing to an audio announcement to play to roll of the waiting/in-queue playback. This needs to be an announcements that never ends (from the perspective of the media server); only the the OpenSIPS Queue may terminate the playback, when it decides to take out the call from waiting and to deliver it to an agent. As for your question, I'm not sure I understand what you mean by "inject a message with queue position for the caller in question" - could you detail please ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11.10.2016 13:36, Jonathan Hunter wrote: Hi guys, I have implemented an opensips/freeswitch environment, and I wish to add call queues to it, and I like the look of call_center, so just checking this out in comparison to mod_callcenter in FS world. My main question is if using the call_center module if you can inject a message with queue position for the caller in question, as I cant see that in documentation, I only see message_queue which I assume could be used to report the callers position, but just wondered if anyone has done this and if they could give me some tips as to if possible? Many thanks Jon ___ 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
Re: [OpenSIPS-Users] opensips 2.1 call_center queue position
Hello Bogdan, Thanks for the response. In terms of my question, with a number of queuing platforms, they have the capability to tell the caller, what position they are in , and when they are likely to be answered. I just wondered if this logic was already within the module, or if I would need to use an external code/script to facilitate this function? As I presume call_center tracks the number of calls currently in a queue ? I would then want to be able to extract that information, and if a caller was for example in 3rd place in a queue, I could inject the relevant audio from freeswitch to tell them their current position? Does that make sense? :) Just wanted to know if its something this module can do? Thanks Jon Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position To: users@lists.opensips.org; hunter...@hotmail.com From: bog...@opensips.org Date: Wed, 12 Oct 2016 11:23:45 +0300 Hello Jon, The message_queue is a SIP URI pointing to an audio announcement to play to roll of the waiting/in-queue playback. This needs to be an announcements that never ends (from the perspective of the media server); only the the OpenSIPS Queue may terminate the playback, when it decides to take out the call from waiting and to deliver it to an agent. As for your question, I'm not sure I understand what you mean by "inject a message with queue position for the caller in question" - could you detail please ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11.10.2016 13:36, Jonathan Hunter wrote: Hi guys, I have implemented an opensips/freeswitch environment, and I wish to add call queues to it, and I like the look of call_center, so just checking this out in comparison to mod_callcenter in FS world. My main question is if using the call_center module if you can inject a message with queue position for the caller in question, as I cant see that in documentation, I only see message_queue which I assume could be used to report the callers position, but just wondered if anyone has done this and if they could give me some tips as to if possible? Many thanks Jon ___ 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] opensips 2.1 call_center queue position
Hello Jon, The message_queue is a SIP URI pointing to an audio announcement to play to roll of the waiting/in-queue playback. This needs to be an announcements that never ends (from the perspective of the media server); only the the OpenSIPS Queue may terminate the playback, when it decides to take out the call from waiting and to deliver it to an agent. As for your question, I'm not sure I understand what you mean by "inject a message with queue position for the caller in question" - could you detail please ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 11.10.2016 13:36, Jonathan Hunter wrote: Hi guys, I have implemented an opensips/freeswitch environment, and I wish to add call queues to it, and I like the look of call_center, so just checking this out in comparison to mod_callcenter in FS world. My main question is if using the call_center module if you can inject a message with queue position for the caller in question, as I cant see that in documentation, I only see message_queue which I assume could be used to report the callers position, but just wondered if anyone has done this and if they could give me some tips as to if possible? Many thanks Jon ___ 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] opensips 2.1 call_center queue position
Hi guys, I have implemented an opensips/freeswitch environment, and I wish to add call queues to it, and I like the look of call_center, so just checking this out in comparison to mod_callcenter in FS world. My main question is if using the call_center module if you can inject a message with queue position for the caller in question, as I cant see that in documentation, I only see message_queue which I assume could be used to report the callers position, but just wondered if anyone has done this and if they could give me some tips as to if possible? Many thanks Jon ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
