subject:"\[cisco\-voip\] Native Call Queuing on UCM 12.5 with SIP Trunks"

Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

2019-12-05 Thread Dana Tong

Sorry. Yes that parameter is enabled.

Cheers
Dana



On 6 Dec 2019, at 3:50 am, Anthony Holloway  
wrote:


IMO Native Call queuing is a disappointment for anyone I turn it on for, and we 
end up turning it off.  Some reasons:

1) Time limit for queue is too low, and configuring a loop just makes callers 
shuffle their position in queue
2) Phone can either be logged in or not as a whole, but for all hunt groups DN 
is a member of, so cannot login to a queue temporarily to help out (like a temp 
reskilling in UCCX)
3) The greeting cannot play unconditionally, caller has to be queued first.
4) Caller cannot be queued if all phones are logged out, rather, it requires 1 
specific scenario of at least 1 logged in phone and all logged in phones must 
be on an active call
5) Reporting

PS I don't think you responded to Adam's comment about duplex Streaming in CUCM 
service params.  Do you have that enabled?

On Wed, Dec 4, 2019 at 10:20 PM Dana Tong 
mailto:dana.t...@yellit.com.au>> wrote:
Hi all,

Not getting too far with the TAC at the moment unfortunately.

However how are you guys doing basic hunt groups? Native Queueing or not?

Plus this customer wants calls to just keep queueing if they have no one logged 
in or they are all on the phone. The only option here is to disconnect the call 
or route to another destination. How’s the best way to try to achieve what he 
wants?

Cheers
Dana



From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Dana Tong
Sent: Tuesday, 3 December 2019 7:11 AM
To: Ryan Huff mailto:ryanh...@outlook.com>>; Mark H. 
Turpin mailto:mtur...@covene.com>>; 
cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

Hi all,

No the fault is still present when selecting MTP Required on the trunk, Early 
offer support mandatory on the SIP profile.
I’ve also checked the option “Connect Inbound Call before Playing Queuing 
Announcement” on the SIP profile.

Again, MOH works fine for a normal call. Just not for the queued call.




 RUNNING CONFIG ##
!
!
!
no voice hunt unassigned-number
no voice hunt invalid-number
voice call send-alert
voice call disc-pi-off
voice call convert-discpi-to-prog
voice rtp send-recv
!
voice service voip
ip address trusted list
  ipv4 x.x.x.x/x
rtp-port range 16384 32766
address-hiding
mode border-element
 allow-connections sip to sip
fax protocol none
 sip
  asserted-id pai
  outbound-proxy dns:sbc-blah
  asymmetric payload full
  options-ping 60
  early-offer forced
  midcall-signaling passthru
  privacy-policy passthru
  sip-profiles 100
  no call service stop
!
voice class codec 10
codec preference 1 g722-64
codec preference 2 g711alaw
codec preference 3 g711ulaw
codec preference 4 g729r8
!
!
voice class sip-profiles 100
request REINVITE sdp-header Audio-Attribute modify "inactive" "sendrecv"
 request ACK sdp-header Audio-Attribute modify "sendonly" "sendrecv"
 request REINVITE sdp-header Audio-Attribute modify "inactive" "sendrecv"
 request ACK sdp-header Audio-Attribute modify "sendonly" "sendrecv"
!
voice class sip-profiles 1
request CANCEL sip-header Max-Forwards modify ".*" "User-Agent: CUBE"
 response ANY sip-header Remote-Party-ID modify 
"(.*@).*>(.*<mailto:.*@).*%3e(.*>)" 
"\1company.com.au<http://1company.com.au>>\2"
 request ANY sip-header Remote-Party-ID modify 
"(.*@).*>(.*<mailto:.*@).*%3e(.*>)" 
"\1company.com.au<http://1company.com.au>>\2"
 request INVITE sip-header P-Asserted-Identity modify "(.*)" 
"P-Asserted-Identity: "
 request REINVITE sdp-header Audio-Attribute modify "inactive" "sendrecv"
 request ACK sdp-header Audio-Attribute modify "sendonly" "sendrecv"
!
!
voice class server-group 1
ipv4 a.b.c.d
ipv4 a.b.c.d
description CUCM Servers
!
!
!
!
voice translation-rule 1
!
voice translation-rule 2
rule 1 /^2/ /02/
rule 2 /^3/ /03/
rule 3 /^4/ /04/
rule 5 /^7/ /07/
rule 6 /^8/ /08/
!
voice translation-rule 100
rule 1 /^0\(.*\)/ /\1/
!
voice translation-rule 200
rule 1 /^0\(.*\)/ /\1/
rule 2 /^30\(..\)/ /7301030\1/
rule 3 /^80\(..\)/ /7308580\1/
rule 4 /^95\(..\)/ /7323195\1/
rule 5 /^96\(..\)/ /7323196\1/
rule 6 /^71\(..\)/ /3886671\1/

!
!
voice translation-profile Incoming-Called
translate calling 2
!
voice translation-profile strip_leading_zero
translate calling 200
translate called 100
translate redirect-called 200
!
!

!
sccp local Port-channel1
sccp ccm 10.100.99.21 identifier 1 priority 2 version 7.0
sccp ccm 10.100.99.22 identifier 2 priority 1 version 7.0
sccp
!
sccp ccm group 1
associate ccm 2 priority 1
associate ccm 1 priority 2
associate profile 2 register Xcode-isr433101
associate profile 1 register CFB-isr433

Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

2019-12-05 Thread Anthony Holloway

IMO Native Call queuing is a disappointment for anyone I turn it on for,
and we end up turning it off.  Some reasons:

1) Time limit for queue is too low, and configuring a loop just makes
callers shuffle their position in queue
2) Phone can either be logged in or not as a whole, but for all hunt groups
DN is a member of, so cannot login to a queue temporarily to help out (like
a temp reskilling in UCCX)
3) The greeting cannot play unconditionally, caller has to be queued first.
4) Caller cannot be queued if all phones are logged out, rather, it
requires 1 specific scenario of at least 1 logged in phone and all logged
in phones must be on an active call
5) Reporting

PS I don't think you responded to Adam's comment about duplex Streaming in
CUCM service params.  Do you have that enabled?

On Wed, Dec 4, 2019 at 10:20 PM Dana Tong  wrote:

> Hi all,
>
>
>
> Not getting too far with the TAC at the moment unfortunately.
>
>
>
> However how are you guys doing basic hunt groups? Native Queueing or not?
>
>
>
> Plus this customer wants calls to just keep queueing if they have no one
> logged in or they are all on the phone. The only option here is to
> disconnect the call or route to another destination. How’s the best way to
> try to achieve what he wants?
>
>
>
> Cheers
>
> Dana
>
>
>
>
>
>
>
> *From:* cisco-voip  *On Behalf Of *Dana
> Tong
> *Sent:* Tuesday, 3 December 2019 7:11 AM
> *To:* Ryan Huff ; Mark H. Turpin ;
> cisco-voip@puck.nether.net
> *Subject:* Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP
> Trunks
>
>
>
> Hi all,
>
>
>
> No the fault is still present when selecting MTP Required on the trunk,
> Early offer support mandatory on the SIP profile.
>
> I’ve also checked the option “Connect Inbound Call before Playing Queuing
> Announcement” on the SIP profile.
>
>
>
> Again, MOH works fine for a normal call. Just not for the queued call.
>
>
>
>
>
>
>
>
>
>  RUNNING CONFIG ##
>
> !
>
> !
>
> !
>
> no voice hunt unassigned-number
>
> no voice hunt invalid-number
>
> voice call send-alert
>
> voice call disc-pi-off
>
> voice call convert-discpi-to-prog
>
> voice rtp send-recv
>
> !
>
> voice service voip
>
> ip address trusted list
>
>   ipv4 x.x.x.x/x
>
> rtp-port range 16384 32766
>
> address-hiding
>
> mode border-element
>
>  allow-connections sip to sip
>
> fax protocol none
>
>  sip
>
>   asserted-id pai
>
>   outbound-proxy dns:sbc-blah
>
>   asymmetric payload full
>
>   options-ping 60
>
>   early-offer forced
>
>   midcall-signaling passthru
>
>   privacy-policy passthru
>
>   sip-profiles 100
>
>   no call service stop
>
> !
>
> voice class codec 10
>
> codec preference 1 g722-64
>
> codec preference 2 g711alaw
>
> codec preference 3 g711ulaw
>
> codec preference 4 g729r8
>
> !
>
> !
>
> voice class sip-profiles 100
>
> request REINVITE sdp-header Audio-Attribute modify "inactive" "sendrecv"
>
>  request ACK sdp-header Audio-Attribute modify "sendonly" "sendrecv"
>
>  request REINVITE sdp-header Audio-Attribute modify "inactive" "sendrecv"
>
>  request ACK sdp-header Audio-Attribute modify "sendonly" "sendrecv"
>
> !
>
> voice class sip-profiles 1
>
> request CANCEL sip-header Max-Forwards modify ".*" "User-Agent: CUBE"
>
>  response ANY sip-header Remote-Party-ID modify "(.*@).*>(.*)" "\
> 1company.com.au>\2"
>
>  request ANY sip-header Remote-Party-ID modify "(.*@).*>(.*)" "\
> 1company.com.au>\2"
>
>  request INVITE sip-header P-Asserted-Identity modify "(.*)"
> "P-Asserted-Identity: "
>
>  request REINVITE sdp-header Audio-Attribute modify "inactive" "sendrecv"
>
>  request ACK sdp-header Audio-Attribute modify "sendonly" "sendrecv"
>
> !
>
> !
>
> voice class server-group 1
>
> ipv4 a.b.c.d
>
> ipv4 a.b.c.d
>
> description CUCM Servers
>
> !
>
> !
>
> !
>
> !
>
> voice translation-rule 1
>
> !
>
> voice translation-rule 2
>
> rule 1 /^2/ /02/
>
> rule 2 /^3/ /03/
>
> rule 3 /^4/ /04/
>
> rule 5 /^7/ /07/
>
> rule 6 /^8/ /08/
>
> !
>
> voice translation-rule 100
>
> rule 1 /^0\(.*\)/ /\1/
>
> !
>
> voice translation-rule 200
>
> rule 1 /^0\(.*\)/ /\1/
>
> rule 2 /^30\(..\)/ /7301030\1/
&g

Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

2019-12-04 Thread Dana Tong

Hi all,

Not getting too far with the TAC at the moment unfortunately.

However how are you guys doing basic hunt groups? Native Queueing or not?

Plus this customer wants calls to just keep queueing if they have no one logged 
in or they are all on the phone. The only option here is to disconnect the call 
or route to another destination. How's the best way to try to achieve what he 
wants?

Cheers
Dana



From: cisco-voip  On Behalf Of Dana Tong
Sent: Tuesday, 3 December 2019 7:11 AM
To: Ryan Huff ; Mark H. Turpin ; 
cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

Hi all,

No the fault is still present when selecting MTP Required on the trunk, Early 
offer support mandatory on the SIP profile.
I've also checked the option "Connect Inbound Call before Playing Queuing 
Announcement" on the SIP profile.

Again, MOH works fine for a normal call. Just not for the queued call.




 RUNNING CONFIG ##
!
!
!
no voice hunt unassigned-number
no voice hunt invalid-number
voice call send-alert
voice call disc-pi-off
voice call convert-discpi-to-prog
voice rtp send-recv
!
voice service voip
ip address trusted list
  ipv4 x.x.x.x/x
rtp-port range 16384 32766
address-hiding
mode border-element
 allow-connections sip to sip
fax protocol none
 sip
  asserted-id pai
  outbound-proxy dns:sbc-blah
  asymmetric payload full
  options-ping 60
  early-offer forced
  midcall-signaling passthru
  privacy-policy passthru
  sip-profiles 100
  no call service stop
!
voice class codec 10
codec preference 1 g722-64
codec preference 2 g711alaw
codec preference 3 g711ulaw
codec preference 4 g729r8
!
!
voice class sip-profiles 100
request REINVITE sdp-header Audio-Attribute modify "inactive" "sendrecv"
 request ACK sdp-header Audio-Attribute modify "sendonly" "sendrecv"
 request REINVITE sdp-header Audio-Attribute modify "inactive" "sendrecv"
 request ACK sdp-header Audio-Attribute modify "sendonly" "sendrecv"
!
voice class sip-profiles 1
request CANCEL sip-header Max-Forwards modify ".*" "User-Agent: CUBE"
 response ANY sip-header Remote-Party-ID modify 
"(.*@).*>(.*<mailto:.*@).*%3e(.*>)" "\1company.com.au>\2"
 request ANY sip-header Remote-Party-ID modify 
"(.*@).*>(.*<mailto:.*@).*%3e(.*>)" "\1company.com.au>\2"
 request INVITE sip-header P-Asserted-Identity modify "(.*)" 
"P-Asserted-Identity: "
 request REINVITE sdp-header Audio-Attribute modify "inactive" "sendrecv"
 request ACK sdp-header Audio-Attribute modify "sendonly" "sendrecv"
!
!
voice class server-group 1
ipv4 a.b.c.d
ipv4 a.b.c.d
description CUCM Servers
!
!
!
!
voice translation-rule 1
!
voice translation-rule 2
rule 1 /^2/ /02/
rule 2 /^3/ /03/
rule 3 /^4/ /04/
rule 5 /^7/ /07/
rule 6 /^8/ /08/
!
voice translation-rule 100
rule 1 /^0\(.*\)/ /\1/
!
voice translation-rule 200
rule 1 /^0\(.*\)/ /\1/
rule 2 /^30\(..\)/ /7301030\1/
rule 3 /^80\(..\)/ /7308580\1/
rule 4 /^95\(..\)/ /7323195\1/
rule 5 /^96\(..\)/ /7323196\1/
rule 6 /^71\(..\)/ /3886671\1/

!
!
voice translation-profile Incoming-Called
translate calling 2
!
voice translation-profile strip_leading_zero
translate calling 200
translate called 100
translate redirect-called 200
!
!

!
sccp local Port-channel1
sccp ccm 10.100.99.21 identifier 1 priority 2 version 7.0
sccp ccm 10.100.99.22 identifier 2 priority 1 version 7.0
sccp
!
sccp ccm group 1
associate ccm 2 priority 1
associate ccm 1 priority 2
associate profile 2 register Xcode-isr433101
associate profile 1 register CFB-isr4331-01
!
ccm-manager music-on-hold
!
!
dspfarm profile 2 transcode
 codec g729abr8
codec g729ar8
codec g711alaw
codec g711ulaw
codec g722-64
codec g729br8
codec g729r8
maximum sessions 3
associate application SCCP
!
dspfarm profile 1 conference
 codec g729br8
codec g729r8
codec g729abr8
codec g729ar8
codec g711alaw
codec g711ulaw
codec g722-64
maximum sessions 4
associate application SCCP
!
dial-peer voice 101 voip
description ## outgoing SIP Connect ##
translation-profile outgoing strip_leading_zero
destination-pattern 0T
session protocol sipv2
session target sip-server
session transport udp
voice-class codec 10
 voice-class sip profiles 1
voice-class sip options-keepalive
dtmf-relay rtp-nte
fax protocol none
ip qos dscp cs3 signaling
!
dial-peer voice 201 voip
description ## incoming SIP Connect ##
translation-profile incoming Incoming-Called
rtp payload-type nse 99
session protocol sipv2
session target sip-server
session transport udp
incoming called-number .
voice-class codec 10
 voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
fax protocol none
!
dial-peer voice 401 voip
description ## outgoing cucm ##
preference 1
destination-pattern [2378]$
session protocol sipv2
session server-

Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

2019-12-02 Thread Pawlowski, Adam

We went from 15.4(3)M9 to 15.7(3)M4b

Noted also in this version that we previously allowed video/media to flow to 
the CUBE just in case we were ever to use it for routing media calls, and this 
version (15.7) drops calls from Jabber, etc when we place them on hold.

TAC said it was something to do with the number of options in the SDP changing 
and it being a protocol violation. The IEC error and Q code did sort of 
indicate this, but, as to why Cisco's systems don't want to play nice protocol 
wise with each other I don't know.



From: Ryan Huff 
Sent: Monday, December 2, 2019 10:25 AM
To: Pawlowski, Adam ; cisco-voip@puck.nether.net
Subject: Re: Native Call Queuing on UCM 12.5 with SIP Trunks

Adam,

I've not encountered the streaming issue with a SIP trunk, just H.323.. 
interesting.

Out of curiosity, what code level in the ISR G2 did you go to when you 
encountered this? The behavior of simplex streaming is to send a null/fake IP 
address in the logic channel multimedia control message, which for anything 
required to check incoming RTP packets against the IP/port would fail (creating 
the silent MoH condition) which I was under the impression was only the G3 ISRs.

Thanks,

Ryan

From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
on behalf of Pawlowski, Adam mailto:aj...@buffalo.edu>>
Sent: Monday, December 2, 2019 8:50 AM
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks


After an upgrade on our CUBE to a later IOS (ISR G2) we needed to have the 
duplex streaming parameter enabled for MOH/Ringback to work properly.



I believe there is a note about that being needed for the 4400 series gateways 
as well, but, I know we didn't have it before but needed it at some point to 
get media to be heard properly by the far end in a network hold kind of 
situation.



Adam







From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of Mark H. Turpin
Sent: Monday, December 2, 2019 8:20 AM
To: Dana Tong mailto:dana.t...@yellit.com.au>>; 
cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks



Does the problem go away if you force an MTP?



Can you provide a debug ccsip messages of an external call?



Can you share your sanitized CUBE config?



Curious if your call/media is changing after the initial announcement and your 
ITSP doesn't care for the way you're attempting to change it.





From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
on behalf of Dana Tong mailto:dana.t...@yellit.com.au>>
Sent: Monday, December 2, 2019 12:36 AM
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> 
mailto:cisco-voip@puck.nether.net>>
Subject: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks



*** EXTERNAL EMAIL - DO NOT CLICK LINKS ***

Hi all,



This being back on the tools is doing my head in. I think I've been away too 
long.



So I have configured native call queuing on UCM 12.5.x

Internal calls queue fine. The initial announcement plays. Period announcements 
work and the MOH is fine in between.



External calls hear the initial announcement.

There is no MOH after the announcement and the external user has ring-back tone.

There is no periodic announcement.



Any thoughts on why internal is okay and external is not working? Is it 
relating to SIP supplementary services?



Cheers

Dana


___
cisco-voip mailing list
cisco-voip@puck.nether.net
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Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

2019-12-02 Thread Ryan Huff

Adam,

I've not encountered the streaming issue with a SIP trunk, just H.323.. 
interesting.

Out of curiosity, what code level in the ISR G2 did you go to when you 
encountered this? The behavior of simplex streaming is to send a null/fake IP 
address in the logic channel multimedia control message, which for anything 
required to check incoming RTP packets against the IP/port would fail (creating 
the silent MoH condition) which I was under the impression was only the G3 ISRs.

Thanks,

Ryan

From: cisco-voip  on behalf of Pawlowski, 
Adam 
Sent: Monday, December 2, 2019 8:50 AM
To: cisco-voip@puck.nether.net 
Subject: Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks


After an upgrade on our CUBE to a later IOS (ISR G2) we needed to have the 
duplex streaming parameter enabled for MOH/Ringback to work properly.



I believe there is a note about that being needed for the 4400 series gateways 
as well, but, I know we didn’t have it before but needed it at some point to 
get media to be heard properly by the far end in a network hold kind of 
situation.



Adam







From: cisco-voip  On Behalf Of Mark H. 
Turpin
Sent: Monday, December 2, 2019 8:20 AM
To: Dana Tong ; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks



Does the problem go away if you force an MTP?



Can you provide a debug ccsip messages of an external call?



Can you share your sanitized CUBE config?



Curious if your call/media is changing after the initial announcement and your 
ITSP doesn't care for the way you're attempting to change it.





From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
on behalf of Dana Tong mailto:dana.t...@yellit.com.au>>
Sent: Monday, December 2, 2019 12:36 AM
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> 
mailto:cisco-voip@puck.nether.net>>
Subject: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks



*** EXTERNAL EMAIL - DO NOT CLICK LINKS ***

Hi all,



This being back on the tools is doing my head in. I think I’ve been away too 
long.



So I have configured native call queuing on UCM 12.5.x

Internal calls queue fine. The initial announcement plays. Period announcements 
work and the MOH is fine in between.



External calls hear the initial announcement.

There is no MOH after the announcement and the external user has ring-back tone.

There is no periodic announcement.



Any thoughts on why internal is okay and external is not working? Is it 
relating to SIP supplementary services?



Cheers

Dana


___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

2019-12-02 Thread Ryan Huff

Mark,

That is a good thought.. there are a few scenarios where a CUCM/CUC.. etc mid 
call change will advertise a "no media change" reINVITE out to the PSTN. While 
I don't think that would be the case here (I would expect the peer to drop the 
call because they couldn't negotiate the null/0.0.0.0 media address), SIP 
traces would certainly show if it were. If it were the case, likely under voice 
service voip >> sip the syntax midcall-signaling passthru media-change would 
prevent mid call signaling from egressing the CUBE unless it actually had 
legitimate media changes.

-Ryan


From: cisco-voip  on behalf of Mark H. 
Turpin 
Sent: Monday, December 2, 2019 8:20 AM
To: Dana Tong ; cisco-voip@puck.nether.net 

Subject: Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

Does the problem go away if you force an MTP?

Can you provide a debug ccsip messages of an external call?

Can you share your sanitized CUBE config?

Curious if your call/media is changing after the initial announcement and your 
ITSP doesn't care for the way you're attempting to change it.


From: cisco-voip  on behalf of Dana Tong 

Sent: Monday, December 2, 2019 12:36 AM
To: cisco-voip@puck.nether.net 
Subject: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

*** EXTERNAL EMAIL - DO NOT CLICK LINKS ***


Hi all,



This being back on the tools is doing my head in. I think I’ve been away too 
long.



So I have configured native call queuing on UCM 12.5.x

Internal calls queue fine. The initial announcement plays. Period announcements 
work and the MOH is fine in between.



External calls hear the initial announcement.

There is no MOH after the announcement and the external user has ring-back tone.

There is no periodic announcement.



Any thoughts on why internal is okay and external is not working? Is it 
relating to SIP supplementary services?



Cheers

Dana


___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

2019-12-02 Thread Pawlowski, Adam

After an upgrade on our CUBE to a later IOS (ISR G2) we needed to have the 
duplex streaming parameter enabled for MOH/Ringback to work properly.

I believe there is a note about that being needed for the 4400 series gateways 
as well, but, I know we didn't have it before but needed it at some point to 
get media to be heard properly by the far end in a network hold kind of 
situation.

Adam



From: cisco-voip  On Behalf Of Mark H. 
Turpin
Sent: Monday, December 2, 2019 8:20 AM
To: Dana Tong ; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

Does the problem go away if you force an MTP?

Can you provide a debug ccsip messages of an external call?

Can you share your sanitized CUBE config?

Curious if your call/media is changing after the initial announcement and your 
ITSP doesn't care for the way you're attempting to change it.


From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
on behalf of Dana Tong mailto:dana.t...@yellit.com.au>>
Sent: Monday, December 2, 2019 12:36 AM
To: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> 
mailto:cisco-voip@puck.nether.net>>
Subject: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

*** EXTERNAL EMAIL - DO NOT CLICK LINKS ***

Hi all,



This being back on the tools is doing my head in. I think I've been away too 
long.



So I have configured native call queuing on UCM 12.5.x

Internal calls queue fine. The initial announcement plays. Period announcements 
work and the MOH is fine in between.



External calls hear the initial announcement.

There is no MOH after the announcement and the external user has ring-back tone.

There is no periodic announcement.



Any thoughts on why internal is okay and external is not working? Is it 
relating to SIP supplementary services?



Cheers

Dana


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Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

2019-12-02 Thread Mark H. Turpin

Does the problem go away if you force an MTP?

Can you provide a debug ccsip messages of an external call?

Can you share your sanitized CUBE config?

Curious if your call/media is changing after the initial announcement and your 
ITSP doesn't care for the way you're attempting to change it.


From: cisco-voip  on behalf of Dana Tong 

Sent: Monday, December 2, 2019 12:36 AM
To: cisco-voip@puck.nether.net 
Subject: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

*** EXTERNAL EMAIL - DO NOT CLICK LINKS ***


Hi all,



This being back on the tools is doing my head in. I think I’ve been away too 
long.



So I have configured native call queuing on UCM 12.5.x

Internal calls queue fine. The initial announcement plays. Period announcements 
work and the MOH is fine in between.



External calls hear the initial announcement.

There is no MOH after the announcement and the external user has ring-back tone.

There is no periodic announcement.



Any thoughts on why internal is okay and external is not working? Is it 
relating to SIP supplementary services?



Cheers

Dana


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Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

2019-12-02 Thread Ryan Huff

Additionally, If it’s a custom MoH (not the built in Cisco tune), make sure the 
file is uploaded to each CUCM server individually; the working phone 2 phone 
scenario could be happening on nodes where the moh file is, and the trunk 
registered to a node where it is not.

Thanks,

Ryan

On Dec 2, 2019, at 02:36, daniele visaggio  wrote:


Hi,

Try to check if your trunk has access to moh resources.

This means your trunk needs to be placed inside a device pool whose mrgl 
contains at least one mrg containing at least one moh server.

Sip trunk > device pool > mrgl > mrg > moh server.

Additional test: call some external party through your sip trunk and try to put 
called party on hold. Does the called party hear moh?

Regards

On Mon, Dec 2, 2019, 07:54 Dana Tong 
mailto:dana.t...@yellit.com.au>> wrote:
Sorry forgot to mention that Supplementary services such as call forward all, 
transfer, hold/retrieve, and conferencing work fine.



From: Dana Tong
Sent: Monday, 2 December 2019 4:37 PM
To: cisco-voip@puck.nether.net
Subject: Native Call Queuing on UCM 12.5 with SIP Trunks

Hi all,

This being back on the tools is doing my head in. I think I’ve been away too 
long.

So I have configured native call queuing on UCM 12.5.x
Internal calls queue fine. The initial announcement plays. Period announcements 
work and the MOH is fine in between.

External calls hear the initial announcement.
There is no MOH after the announcement and the external user has ring-back tone.
There is no periodic announcement.

Any thoughts on why internal is okay and external is not working? Is it 
relating to SIP supplementary services?

Cheers
Dana

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Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

2019-12-01 Thread daniele visaggio

Hi,

Try to check if your trunk has access to moh resources.

This means your trunk needs to be placed inside a device pool whose mrgl
contains at least one mrg containing at least one moh server.

Sip trunk > device pool > mrgl > mrg > moh server.

Additional test: call some external party through your sip trunk and try to
put called party on hold. Does the called party hear moh?

Regards

On Mon, Dec 2, 2019, 07:54 Dana Tong  wrote:

> Sorry forgot to mention that Supplementary services such as call forward
> all, transfer, hold/retrieve, and conferencing work fine.
>
>
>
>
>
>
>
> *From:* Dana Tong
> *Sent:* Monday, 2 December 2019 4:37 PM
> *To:* cisco-voip@puck.nether.net
> *Subject:* Native Call Queuing on UCM 12.5 with SIP Trunks
>
>
>
> Hi all,
>
>
>
> This being back on the tools is doing my head in. I think I’ve been away
> too long.
>
>
>
> So I have configured native call queuing on UCM 12.5.x
>
> Internal calls queue fine. The initial announcement plays. Period
> announcements work and the MOH is fine in between.
>
>
>
> External calls hear the initial announcement.
>
> There is no MOH after the announcement and the external user has ring-back
> tone.
>
> There is no periodic announcement.
>
>
>
> Any thoughts on why internal is okay and external is not working? Is it
> relating to SIP supplementary services?
>
>
>
> Cheers
>
> Dana
>
>
> ___
> cisco-voip mailing list
> cisco-voip@puck.nether.net
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>
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Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

2019-12-01 Thread Dana Tong

Sorry forgot to mention that Supplementary services such as call forward all, 
transfer, hold/retrieve, and conferencing work fine.

From: Dana Tong
Sent: Monday, 2 December 2019 4:37 PM
To: cisco-voip@puck.nether.net
Subject: Native Call Queuing on UCM 12.5 with SIP Trunks

Hi all,

This being back on the tools is doing my head in. I think I've been away too 
long.

So I have configured native call queuing on UCM 12.5.x
Internal calls queue fine. The initial announcement plays. Period announcements 
work and the MOH is fine in between.

External calls hear the initial announcement.
There is no MOH after the announcement and the external user has ring-back tone.
There is no periodic announcement.

Any thoughts on why internal is okay and external is not working? Is it 
relating to SIP supplementary services?

Cheers
Dana

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[cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

2019-12-01 Thread Dana Tong

Hi all,

This being back on the tools is doing my head in. I think I've been away too 
long.

So I have configured native call queuing on UCM 12.5.x
Internal calls queue fine. The initial announcement plays. Period announcements 
work and the MOH is fine in between.

External calls hear the initial announcement.
There is no MOH after the announcement and the external user has ring-back tone.
There is no periodic announcement.

Any thoughts on why internal is okay and external is not working? Is it 
relating to SIP supplementary services?

Cheers
Dana

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Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

Re: [cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

[cisco-voip] Native Call Queuing on UCM 12.5 with SIP Trunks

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