date:20190325

Re: [cisco-voip] video based auto-attendant

2019-03-25 Thread Lelio Fulgenzi

Those look like really neat services. I’ll be honest, I didn’t expect the video 
IVR to be so expensive. ☹ Wish I had the ability to do it with the on-prem 
equipment and software I have.

---
Lelio Fulgenzi, B.A. | Senior Analyst
Computing and Communications Services | University of Guelph
Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 2W1
519-824-4120 Ext. 56354 | le...@uoguelph.ca

www.uoguelph.ca/ccs | @UofGCCS on Instagram, 
Twitter and Facebook

[University of Guelph Cornerstone with Improve Life tagline]

From: Bernhard Albler 
Sent: Monday, March 25, 2019 1:57 AM
To: Lelio Fulgenzi 
Cc: cisco-voip voyp list 
Subject: Re: [cisco-voip] video based auto-attendant

Cloud wise selfie.vc / rec.vc and 
ivr.vc (https://www.ivr.vc) will fit the bill

cheers
bernhard

On Sat, Mar 23, 2019 at 11:05 PM Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:

I’m thinking about setting something up for our video conference rooms... 
something like video based help videos.

But with MediaSense being EOL’ed, video greetings on Connection seems like a 
lost cause.

What other options are out there?

It would have to be a unicast type of solution, so each access would play the 
file from the beginning.

Are there cloud based solutions?




-sent from mobile device-

Lelio Fulgenzi, B.A. | Senior Analyst
Computing and Communications Services | University of Guelph
Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON | N1G 2W1
519-824-4120 Ext. 56354 | 
le...@uoguelph.ca

www.uoguelph.ca/ccs | @UofGCCS on Instagram, 
Twitter and Facebook

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--
Bernhard Albler, +4369917207384
--
"Was Nachwelt! Wie komm' ich dazu was für die Nachwelt zu tun? Was hat denn die 
Nachwelt für mich getan?"
--Carl Friedrich Zelter
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[cisco-voip] SIp Trunk call failing after PBX upgrade

2019-03-25 Thread ROZA, Ariel

Hi, guys and gals.

I have a customer with a CUCM 9.0(2) cluster.
It is connected to a SIP PBX via a direct SIP TRUNK (No SBC, CUBE or 
otherwise). The PBX has four different nodes, all configured in the SIP TRUNK

They claim it was working fine until last Thursday, where they did an upgrade 
to one of the nodes of the PBX. After that, calls going from PBX to CUCM fail 
with a 488 Media Not Acceptable error.
They also have tried making calls from one of the not upgraded nodes, with the 
same error.
I have been looking into the SIP traces, and I see nothing really telling of a 
problem there.

We reseted the SIP trunk with no success.
I have looked at the región configuration, and all regions are set to the 
System Default (G722, G711)
I also tried changing the preferred codec in the SIP trunk, with no success.

Following this, I am pasting the SIP messages of a failed call from PBX -> CUCM 
and a successfull call in the reverse, from CUCM -> PBX.

Can you see if anything is wrong or odd?

Regards,

Ariel.

Failed Call from PBX


INVITE sip:3366@10.4.128.27 SIP/2.0
Via: SIP/2.0/UDP 
172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm
From: " " ;tag=2792862
To: 
Call-ID: 501227892-15@172.27.0.15
CSeq: 1 INVITE
Contact: 
Max-Forwards: 70
User-Agent: MitE1x v4.4.5.1062
Expires: 300
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO
P-Early-Media: Supported
P-Asserted-Identity: " " 
P-Mitrol-idLlamada: 190322160050689_MIT_07437
P-Mitrol-LoginID: 
P-Mitrol-PerfilRuteo: 100
Content-Length: 233
Content-Type: application/sdp
v=0
o=86329 -835641967 1 IN IP4 172.27.0.15
s=MitE1x Call
c=IN IP4 172.27.0.15
t=0 0
m=audio 36112 RTP/AVP 0 8 101
a=sendrecv
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15


Reply from CUCM
---

SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/UDP 
172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm
From: "Gabriel Querol" ;tag=2792862
To: ;tag=573234994
Date: Fri, 22 Mar 2019 19:00:23 GMT
Call-ID: 501227892-15@172.27.0.15
CSeq: 1 INVITE
Allow-Events: presence
Warning: 304 10.4.128.27 "Media Type(s) Unavailable"
Reason: Q.850;cause=65
Content-Length: 0




SUCESSFULL CALL FROM CUCM
-
INVITE sip:*86329@172.27.0.12:5060 SIP/2.0
Via: SIP/2.0/UDP 10.4.128.27:5060;branch=z9hG4bK668c2eabb0b8
From: "  (3307)" 
;tag=429005~b085ab57-efd9-4eb3-9a97-adbf28ac4c95-50893220
To: 
Date: Mon, 25 Mar 2019 10:40:36 GMT
Call-ID: 6b366f80-c981b024-4f13-1b80040a@10.4.128.27
Supported: timer,resource-priority,replaces
Min-SE:  1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, 
SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Cisco-Guid: 1798729600-065536-010811-0461374474
Session-Expires:  1800
P-Asserted-Identity: "  (3307)" 
Remote-Party-ID: "  (3307)" 
;party=calling;screen=yes;privacy=off
Contact: 
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 212
v=0
o=CiscoSystemsCCM-SIP 429005 1 IN IP4 10.4.128.27
s=SIP Call
c=IN IP4 10.4.128.12
t=0 0
m=audio 30530 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15


Answer from the PBX
--

SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.4.128.27:5060;branch=z9hG4bK668c2eabb0b8
From: "Gabriel Querol (3307)" 
;tag=429005~b085ab57-efd9-4eb3-9a97-adbf28ac4c95-50893220
To: ;tag=43743456
Call-ID: 6b366f80-c981b024-4f13-1b80040a@10.4.128.27
CSeq: 101 INVITE
Server: MitE1x v4.4.5.1062
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO
P-Mitrol-idLlamada: 190325074112281_MIT_02447
Content-Length: 217
Content-Type: application/sdp
v=0
o=CiscoSystemsCCM-SIP 429005 1 IN IP4 172.27.0.12
s=MitE1x Call
c=IN IP4 172.27.0.12
t=0 0
m=audio 36508 RTP/AVP 8 101
a=sendrecv
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

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Re: [cisco-voip] SIp Trunk call failing after PBX upgrade

2019-03-25 Thread NateCCIE

Non working call shows G711u and a, working call shows only a.  there is
also a difference of the /1 at the end not sure what that indicates.

 

a=rtpmap:0 PCMU/8000/1

a=rtpmap:8 PCMA/8000

 

 

From: cisco-voip  On Behalf Of ROZA,
Ariel
Sent: Monday, March 25, 2019 11:17 AM
To: cisco-voip (cisco-voip@puck.nether.net) 
Subject: [cisco-voip] SIp Trunk call failing after PBX upgrade

 

Hi, guys and gals.

 

I have a customer with a CUCM 9.0(2) cluster.

It is connected to a SIP PBX via a direct SIP TRUNK (No SBC, CUBE or
otherwise). The PBX has four different nodes, all configured in the SIP
TRUNK

 

They claim it was working fine until last Thursday, where they did an
upgrade to one of the nodes of the PBX. After that, calls going from PBX to
CUCM fail with a 488 Media Not Acceptable error.

They also have tried making calls from one of the not upgraded nodes, with
the same error.

I have been looking into the SIP traces, and I see nothing really telling of
a problem there.

 

We reseted the SIP trunk with no success.

I have looked at the región configuration, and all regions are set to the
System Default (G722, G711)

I also tried changing the preferred codec in the SIP trunk, with no success.

 

Following this, I am pasting the SIP messages of a failed call from PBX ->
CUCM and a successfull call in the reverse, from CUCM -> PBX.

 

Can you see if anything is wrong or odd?

 

Regards,

 

Ariel.

 

Failed Call from PBX



 

INVITE sip:3366@10.4.128.27 SIP/2.0

Via: SIP/2.0/UDP
172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm

From: " " ;tag=2792862

To: 

Call-ID: 501227892-15@172.27.0.15  

CSeq: 1 INVITE

Contact: 

Max-Forwards: 70

User-Agent: MitE1x v4.4.5.1062

Expires: 300

Allow: INVITE,ACK,CANCEL,BYE,REGISTER,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO

P-Early-Media: Supported

P-Asserted-Identity: " " 

P-Mitrol-idLlamada: 190322160050689_MIT_07437

P-Mitrol-LoginID: 

P-Mitrol-PerfilRuteo: 100

Content-Length: 233

Content-Type: application/sdp

v=0

o=86329 -835641967 1 IN IP4 172.27.0.15

s=MitE1x Call

c=IN IP4 172.27.0.15

t=0 0

m=audio 36112 RTP/AVP 0 8 101

a=sendrecv

a=rtpmap:0 PCMU/8000/1

a=rtpmap:8 PCMA/8000/1

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

 

Reply from CUCM

---

 

SIP/2.0 488 Not Acceptable Media

Via: SIP/2.0/UDP
172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm

From: "Gabriel Querol" ;tag=2792862

To: ;tag=573234994

Date: Fri, 22 Mar 2019 19:00:23 GMT

Call-ID: 501227892-15@172.27.0.15  

CSeq: 1 INVITE

Allow-Events: presence

Warning: 304 10.4.128.27 "Media Type(s) Unavailable"

Reason: Q.850;cause=65

Content-Length: 0

 

 

 

 

SUCESSFULL CALL FROM CUCM

-

INVITE sip:*86329@172.27.0.12:5060 SIP/2.0

Via: SIP/2.0/UDP 10.4.128.27:5060;branch=z9hG4bK668c2eabb0b8

From: "  (3307)"
;tag=429005~b085ab57-efd9-4eb3-9a97-adbf28ac4c95-50893
220

To: 

Date: Mon, 25 Mar 2019 10:40:36 GMT

Call-ID: 6b366f80-c981b024-4f13-1b80040a@10.4.128.27
 

Supported: timer,resource-priority,replaces

Min-SE:  1800

User-Agent: Cisco-CUCM9.1

Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
SUBSCRIBE, NOTIFY

CSeq: 101 INVITE

Expires: 180

Allow-Events: presence, kpml

Supported: X-cisco-srtp-fallback,X-cisco-original-called

Cisco-Guid: 1798729600-065536-010811-0461374474

Session-Expires:  1800

P-Asserted-Identity: "  (3307)" 

Remote-Party-ID: "  (3307)"
;party=calling;screen=yes;privacy=off

Contact: 

Max-Forwards: 69

Content-Type: application/sdp

Content-Length: 212

v=0

o=CiscoSystemsCCM-SIP 429005 1 IN IP4 10.4.128.27

s=SIP Call

c=IN IP4 10.4.128.12

t=0 0

m=audio 30530 RTP/AVP 8 101

a=rtpmap:8 PCMA/8000

a=ptime:20

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

 

Answer from the PBX

--

 

SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 10.4.128.27:5060;branch=z9hG4bK668c2eabb0b8

From: "Gabriel Querol (3307)"
;tag=429005~b085ab57-efd9-4eb3-9a97-adbf28ac4c95-50893
220

To: ;tag=43743456

Call-ID: 6b366f80-c981b024-4f13-1b80040a@10.4.128.27
 

CSeq: 101 INVITE

Server: MitE1x v4.4.5.1062

Allow: INVITE,ACK,CANCEL,BYE,REGISTER,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO

P-Mitrol-idLlamada: 190325074112281_MIT_02447

Content-Length: 217

Content-Type: application/sdp

v=0

o=CiscoSystemsCCM-SIP 429005 1 IN IP4 172.27.0.12

s=MitE1x Call

c=IN IP4 172.27.0.12

t=0 0

m=audio 36508 RTP/AVP 8 101

a=sendrecv

a=rtpmap:8 PCMA/8000/1

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

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Re: [cisco-voip] SIp Trunk call failing after PBX upgrade

2019-03-25 Thread ROZA, Ariel

Yes I already looked at that /1. According to the RFC, the /1 denotes the 
quantity of channels and it is optional when the codec uses only one channel.

I looked up posible bugs related to that in the Bug Search Tool and did not 
find anything suitable.
Already tried changing the Preferred codec to G711U and got the same results, 
except the output now shows PCMU/8000 from CUCM side, as expected.

Thanks, Nate.

De: NateCCIE 
Enviado el: lunes, 25 de marzo de 2019 14:33
Para: ROZA, Ariel ; 'cisco-voip' 

Asunto: RE: [cisco-voip] SIp Trunk call failing after PBX upgrade

Non working call shows G711u and a, working call shows only a.  there is also a 
difference of the /1 at the end not sure what that indicates.

a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000


From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of ROZA, Ariel
Sent: Monday, March 25, 2019 11:17 AM
To: cisco-voip (cisco-voip@puck.nether.net) 
mailto:cisco-voip@puck.nether.net>>
Subject: [cisco-voip] SIp Trunk call failing after PBX upgrade

Hi, guys and gals.

I have a customer with a CUCM 9.0(2) cluster.
It is connected to a SIP PBX via a direct SIP TRUNK (No SBC, CUBE or 
otherwise). The PBX has four different nodes, all configured in the SIP TRUNK

They claim it was working fine until last Thursday, where they did an upgrade 
to one of the nodes of the PBX. After that, calls going from PBX to CUCM fail 
with a 488 Media Not Acceptable error.
They also have tried making calls from one of the not upgraded nodes, with the 
same error.
I have been looking into the SIP traces, and I see nothing really telling of a 
problem there.

We reseted the SIP trunk with no success.
I have looked at the región configuration, and all regions are set to the 
System Default (G722, G711)
I also tried changing the preferred codec in the SIP trunk, with no success.

Following this, I am pasting the SIP messages of a failed call from PBX -> CUCM 
and a successfull call in the reverse, from CUCM -> PBX.

Can you see if anything is wrong or odd?

Regards,

Ariel.

Failed Call from PBX


INVITE sip:3366@10.4.128.27 SIP/2.0
Via: SIP/2.0/UDP 
172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm
From: " " ;tag=2792862
To: 
Call-ID: 501227892-15@172.27.0.15
CSeq: 1 INVITE
Contact: 
Max-Forwards: 70
User-Agent: MitE1x v4.4.5.1062
Expires: 300
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO
P-Early-Media: Supported
P-Asserted-Identity: " " 
P-Mitrol-idLlamada: 190322160050689_MIT_07437
P-Mitrol-LoginID: 
P-Mitrol-PerfilRuteo: 100
Content-Length: 233
Content-Type: application/sdp
v=0
o=86329 -835641967 1 IN IP4 172.27.0.15
s=MitE1x Call
c=IN IP4 172.27.0.15
t=0 0
m=audio 36112 RTP/AVP 0 8 101
a=sendrecv
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15


Reply from CUCM
---

SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/UDP 
172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm
From: "Gabriel Querol" ;tag=2792862
To: ;tag=573234994
Date: Fri, 22 Mar 2019 19:00:23 GMT
Call-ID: 501227892-15@172.27.0.15
CSeq: 1 INVITE
Allow-Events: presence
Warning: 304 10.4.128.27 "Media Type(s) Unavailable"
Reason: Q.850;cause=65
Content-Length: 0




SUCESSFULL CALL FROM CUCM
-
INVITE sip:*86329@172.27.0.12:5060 SIP/2.0
Via: SIP/2.0/UDP 10.4.128.27:5060;branch=z9hG4bK668c2eabb0b8
From: "  (3307)" 
;tag=429005~b085ab57-efd9-4eb3-9a97-adbf28ac4c95-50893220
To: 
Date: Mon, 25 Mar 2019 10:40:36 GMT
Call-ID: 
6b366f80-c981b024-4f13-1b80040a@10.4.128.27
Supported: timer,resource-priority,replaces
Min-SE:  1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, 
SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Cisco-Guid: 1798729600-065536-010811-0461374474
Session-Expires:  1800
P-Asserted-Identity: "  (3307)" 
Remote-Party-ID: "  (3307)" 
;party=calling;screen=yes;privacy=off
Contact: 
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 212
v=0
o=CiscoSystemsCCM-SIP 429005 1 IN IP4 10.4.128.27
s=SIP Call
c=IN IP4 10.4.128.12
t=0 0
m=audio 30530 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15


Answer from the PBX
--

SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.4.128.27:5060;branch=z9hG4bK668c2eabb0b8
From: "Gabriel Querol (3307)" 
;tag=429005~b085ab57-efd9-4eb3-9a97-adbf28ac4c95-50893220
To: ;tag=43743456
Call-ID: 
6b366f80-c981b024-4f13-1b80040a@10.4.128.27
CSeq: 101 INVITE
Server: MitE1x v4.4.5.1062
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,SUBSCRIBE,NOTIFY,REF

Re: [cisco-voip] SIp Trunk call failing after PBX upgrade

2019-03-25 Thread NateCCIE

I would change preferred codec to 711a and see what happens.

From: ROZA, Ariel  
Sent: Monday, March 25, 2019 1:37 PM
To: NateCCIE ; 'cisco-voip' 
Subject: RE: [cisco-voip] SIp Trunk call failing after PBX upgrade

Yes I already looked at that /1. According to the RFC, the /1 denotes the
quantity of channels and it is optional when the codec uses only one
channel.

I looked up posible bugs related to that in the Bug Search Tool and did not
find anything suitable.

Already tried changing the Preferred codec to G711U and got the same
results, except the output now shows PCMU/8000 from CUCM side, as expected.

Thanks, Nate.

De: NateCCIE mailto:natec...@gmail.com> > 
Enviado el: lunes, 25 de marzo de 2019 14:33
Para: ROZA, Ariel mailto:ariel.r...@la.logicalis.com> >; 'cisco-voip'
mailto:cisco-voip@puck.nether.net> >
Asunto: RE: [cisco-voip] SIp Trunk call failing after PBX upgrade

Non working call shows G711u and a, working call shows only a.  there is
also a difference of the /1 at the end not sure what that indicates.

a=rtpmap:0 PCMU/8000/1

a=rtpmap:8 PCMA/8000

From: cisco-voip mailto:cisco-voip-boun...@puck.nether.net> > On Behalf Of ROZA, Ariel
Sent: Monday, March 25, 2019 11:17 AM
To: cisco-voip (cisco-voip@puck.nether.net
 ) mailto:cisco-voip@puck.nether.net> >
Subject: [cisco-voip] SIp Trunk call failing after PBX upgrade

Hi, guys and gals.

I have a customer with a CUCM 9.0(2) cluster.

It is connected to a SIP PBX via a direct SIP TRUNK (No SBC, CUBE or
otherwise). The PBX has four different nodes, all configured in the SIP
TRUNK

They claim it was working fine until last Thursday, where they did an
upgrade to one of the nodes of the PBX. After that, calls going from PBX to
CUCM fail with a 488 Media Not Acceptable error.

They also have tried making calls from one of the not upgraded nodes, with
the same error.

I have been looking into the SIP traces, and I see nothing really telling of
a problem there.

We reseted the SIP trunk with no success.

I have looked at the región configuration, and all regions are set to the
System Default (G722, G711)

I also tried changing the preferred codec in the SIP trunk, with no success.

Following this, I am pasting the SIP messages of a failed call from PBX ->
CUCM and a successfull call in the reverse, from CUCM -> PBX.

Can you see if anything is wrong or odd?

Regards,

Ariel.

Failed Call from PBX

INVITE sip:3366@10.4.128.27 SIP/2.0

Via: SIP/2.0/UDP
172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm

From: " " ;tag=2792862

To: 

Call-ID: 501227892-15@172.27.0.15  

CSeq: 1 INVITE

Contact: 

Max-Forwards: 70

User-Agent: MitE1x v4.4.5.1062

Expires: 300

Allow: INVITE,ACK,CANCEL,BYE,REGISTER,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO

P-Early-Media: Supported

P-Asserted-Identity: " " 

P-Mitrol-idLlamada: 190322160050689_MIT_07437

P-Mitrol-LoginID: 

P-Mitrol-PerfilRuteo: 100

Content-Length: 233

Content-Type: application/sdp

v=0

o=86329 -835641967 1 IN IP4 172.27.0.15

s=MitE1x Call

c=IN IP4 172.27.0.15

t=0 0

m=audio 36112 RTP/AVP 0 8 101

a=sendrecv

a=rtpmap:0 PCMU/8000/1

a=rtpmap:8 PCMA/8000/1

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

Reply from CUCM

---

SIP/2.0 488 Not Acceptable Media

Via: SIP/2.0/UDP
172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm

From: "Gabriel Querol" ;tag=2792862

To: ;tag=573234994

Date: Fri, 22 Mar 2019 19:00:23 GMT

Call-ID: 501227892-15@172.27.0.15  

CSeq: 1 INVITE

Allow-Events: presence

Warning: 304 10.4.128.27 "Media Type(s) Unavailable"

Reason: Q.850;cause=65

Content-Length: 0

SUCESSFULL CALL FROM CUCM

-

INVITE sip:*86329@172.27.0.12:5060 SIP/2.0

Via: SIP/2.0/UDP 10.4.128.27:5060;branch=z9hG4bK668c2eabb0b8

From: "  (3307)"
;tag=429005~b085ab57-efd9-4eb3-9a97-adbf28ac4c95-50893
220

To: 

Date: Mon, 25 Mar 2019 10:40:36 GMT

Call-ID: 6b366f80-c981b024-4f13-1b80040a@10.4.128.27

Supported: timer,resource-priority,replaces

Min-SE:  1800

User-Agent: Cisco-CUCM9.1

Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
SUBSCRIBE, NOTIFY

CSeq: 101 INVITE

Expires: 180

Allow-Events: presence, kpml

Supported: X-cisco-srtp-fallback,X-cisco-original-called

Cisco-Guid: 1798729600-065536-010811-0461374474

Session-Expires:  1800

P-Asserted-Identity: "  (3307)" 

Remote-Party-ID: "  (3307)"
;party=calling;screen=yes;privacy=off

Contact: 

Max-Forwards: 69

Content-Type: application/sdp

Content-Length: 212

v=0

o=CiscoSystemsCCM-SIP 429005 1 IN IP4 10.4.128.27

s=SIP Call

c=IN IP4 10.4.128.12

t=0 0

m=audio 30530 RTP/AVP 8 101

a=rtpmap:8 PCMA/8000

a=ptime:20

a=rtpmap:101 telephone-event/8000

a=fmtp

Re: [cisco-voip] SIp Trunk call failing after PBX upgrade

2019-03-25 Thread ROZA, Ariel

That was the original setting, and the results is what I included in the mail

De: NateCCIE 
Enviado el: lunes, 25 de marzo de 2019 17:01
Para: ROZA, Ariel ; 'cisco-voip' 

Asunto: RE: [cisco-voip] SIp Trunk call failing after PBX upgrade

I would change preferred codec to 711a and see what happens.

From: ROZA, Ariel 
mailto:ariel.r...@la.logicalis.com>>
Sent: Monday, March 25, 2019 1:37 PM
To: NateCCIE mailto:natec...@gmail.com>>; 'cisco-voip' 
mailto:cisco-voip@puck.nether.net>>
Subject: RE: [cisco-voip] SIp Trunk call failing after PBX upgrade

Yes I already looked at that /1. According to the RFC, the /1 denotes the 
quantity of channels and it is optional when the codec uses only one channel.

I looked up posible bugs related to that in the Bug Search Tool and did not 
find anything suitable.
Already tried changing the Preferred codec to G711U and got the same results, 
except the output now shows PCMU/8000 from CUCM side, as expected.

Thanks, Nate.

De: NateCCIE mailto:natec...@gmail.com>>
Enviado el: lunes, 25 de marzo de 2019 14:33
Para: ROZA, Ariel 
mailto:ariel.r...@la.logicalis.com>>; 'cisco-voip' 
mailto:cisco-voip@puck.nether.net>>
Asunto: RE: [cisco-voip] SIp Trunk call failing after PBX upgrade

Non working call shows G711u and a, working call shows only a.  there is also a 
difference of the /1 at the end not sure what that indicates.

a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000


From: cisco-voip 
mailto:cisco-voip-boun...@puck.nether.net>> 
On Behalf Of ROZA, Ariel
Sent: Monday, March 25, 2019 11:17 AM
To: cisco-voip (cisco-voip@puck.nether.net) 
mailto:cisco-voip@puck.nether.net>>
Subject: [cisco-voip] SIp Trunk call failing after PBX upgrade

Hi, guys and gals.

I have a customer with a CUCM 9.0(2) cluster.
It is connected to a SIP PBX via a direct SIP TRUNK (No SBC, CUBE or 
otherwise). The PBX has four different nodes, all configured in the SIP TRUNK

They claim it was working fine until last Thursday, where they did an upgrade 
to one of the nodes of the PBX. After that, calls going from PBX to CUCM fail 
with a 488 Media Not Acceptable error.
They also have tried making calls from one of the not upgraded nodes, with the 
same error.
I have been looking into the SIP traces, and I see nothing really telling of a 
problem there.

We reseted the SIP trunk with no success.
I have looked at the región configuration, and all regions are set to the 
System Default (G722, G711)
I also tried changing the preferred codec in the SIP trunk, with no success.

Following this, I am pasting the SIP messages of a failed call from PBX -> CUCM 
and a successfull call in the reverse, from CUCM -> PBX.

Can you see if anything is wrong or odd?

Regards,

Ariel.

Failed Call from PBX


INVITE sip:3366@10.4.128.27 SIP/2.0
Via: SIP/2.0/UDP 
172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm
From: " " ;tag=2792862
To: 
Call-ID: 501227892-15@172.27.0.15
CSeq: 1 INVITE
Contact: 
Max-Forwards: 70
User-Agent: MitE1x v4.4.5.1062
Expires: 300
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO
P-Early-Media: Supported
P-Asserted-Identity: " " 
P-Mitrol-idLlamada: 190322160050689_MIT_07437
P-Mitrol-LoginID: 
P-Mitrol-PerfilRuteo: 100
Content-Length: 233
Content-Type: application/sdp
v=0
o=86329 -835641967 1 IN IP4 172.27.0.15
s=MitE1x Call
c=IN IP4 172.27.0.15
t=0 0
m=audio 36112 RTP/AVP 0 8 101
a=sendrecv
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15


Reply from CUCM
---

SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/UDP 
172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm
From: "Gabriel Querol" ;tag=2792862
To: ;tag=573234994
Date: Fri, 22 Mar 2019 19:00:23 GMT
Call-ID: 501227892-15@172.27.0.15
CSeq: 1 INVITE
Allow-Events: presence
Warning: 304 10.4.128.27 "Media Type(s) Unavailable"
Reason: Q.850;cause=65
Content-Length: 0




SUCESSFULL CALL FROM CUCM
-
INVITE sip:*86329@172.27.0.12:5060 SIP/2.0
Via: SIP/2.0/UDP 10.4.128.27:5060;branch=z9hG4bK668c2eabb0b8
From: "  (3307)" 
;tag=429005~b085ab57-efd9-4eb3-9a97-adbf28ac4c95-50893220
To: 
Date: Mon, 25 Mar 2019 10:40:36 GMT
Call-ID: 
6b366f80-c981b024-4f13-1b80040a@10.4.128.27
Supported: timer,resource-priority,replaces
Min-SE:  1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, 
SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Cisco-Guid: 1798729600-065536-010811-0461374474
Session-Expires:  1800
P-Asserted-Identity: "  (3307)" 
Remote-Party-ID: "  (3307)" 
;party=calling;screen=yes;privacy=off
Contact: 
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 212

Re: [cisco-voip] SIp Trunk call failing after PBX upgrade

2019-03-25 Thread NateCCIE

 

Cause No. 65 - bearer capability not implemented.
This cause indicates that the equipment sending this cause does not support
the bearer capability requested.

What it means:



1.  In most cases, the number being called is not an ISDN number but an
analog destination.
2.  The equipment is dialing at a faster rate than the circuitry allows,
for example, dialing at 64K when only 56K is supported.

 

Where is the call going, out a gateway or just a Cisco phone?

 

From: ROZA, Ariel  
Sent: Monday, March 25, 2019 2:03 PM
To: NateCCIE ; 'cisco-voip' 
Subject: RE: [cisco-voip] SIp Trunk call failing after PBX upgrade

 

That was the original setting, and the results is what I included in the
mail

 

De: NateCCIE mailto:natec...@gmail.com> > 
Enviado el: lunes, 25 de marzo de 2019 17:01
Para: ROZA, Ariel mailto:ariel.r...@la.logicalis.com> >; 'cisco-voip'
mailto:cisco-voip@puck.nether.net> >
Asunto: RE: [cisco-voip] SIp Trunk call failing after PBX upgrade

 

I would change preferred codec to 711a and see what happens.

 

From: ROZA, Ariel mailto:ariel.r...@la.logicalis.com> > 
Sent: Monday, March 25, 2019 1:37 PM
To: NateCCIE mailto:natec...@gmail.com> >; 'cisco-voip'
mailto:cisco-voip@puck.nether.net> >
Subject: RE: [cisco-voip] SIp Trunk call failing after PBX upgrade

 

Yes I already looked at that /1. According to the RFC, the /1 denotes the
quantity of channels and it is optional when the codec uses only one
channel.

 

I looked up posible bugs related to that in the Bug Search Tool and did not
find anything suitable.

Already tried changing the Preferred codec to G711U and got the same
results, except the output now shows PCMU/8000 from CUCM side, as expected.

 

Thanks, Nate.

 

De: NateCCIE mailto:natec...@gmail.com> > 
Enviado el: lunes, 25 de marzo de 2019 14:33
Para: ROZA, Ariel mailto:ariel.r...@la.logicalis.com> >; 'cisco-voip'
mailto:cisco-voip@puck.nether.net> >
Asunto: RE: [cisco-voip] SIp Trunk call failing after PBX upgrade

 

Non working call shows G711u and a, working call shows only a.  there is
also a difference of the /1 at the end not sure what that indicates.

 

a=rtpmap:0 PCMU/8000/1

a=rtpmap:8 PCMA/8000

 

 

From: cisco-voip mailto:cisco-voip-boun...@puck.nether.net> > On Behalf Of ROZA, Ariel
Sent: Monday, March 25, 2019 11:17 AM
To: cisco-voip (cisco-voip@puck.nether.net
 ) mailto:cisco-voip@puck.nether.net> >
Subject: [cisco-voip] SIp Trunk call failing after PBX upgrade

 

Hi, guys and gals.

 

I have a customer with a CUCM 9.0(2) cluster.

It is connected to a SIP PBX via a direct SIP TRUNK (No SBC, CUBE or
otherwise). The PBX has four different nodes, all configured in the SIP
TRUNK

 

They claim it was working fine until last Thursday, where they did an
upgrade to one of the nodes of the PBX. After that, calls going from PBX to
CUCM fail with a 488 Media Not Acceptable error.

They also have tried making calls from one of the not upgraded nodes, with
the same error.

I have been looking into the SIP traces, and I see nothing really telling of
a problem there.

 

We reseted the SIP trunk with no success.

I have looked at the región configuration, and all regions are set to the
System Default (G722, G711)

I also tried changing the preferred codec in the SIP trunk, with no success.

 

Following this, I am pasting the SIP messages of a failed call from PBX ->
CUCM and a successfull call in the reverse, from CUCM -> PBX.

 

Can you see if anything is wrong or odd?

 

Regards,

 

Ariel.

 

Failed Call from PBX



 

INVITE sip:3366@10.4.128.27 SIP/2.0

Via: SIP/2.0/UDP
172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm

From: " " ;tag=2792862

To: 

Call-ID: 501227892-15@172.27.0.15  

CSeq: 1 INVITE

Contact: 

Max-Forwards: 70

User-Agent: MitE1x v4.4.5.1062

Expires: 300

Allow: INVITE,ACK,CANCEL,BYE,REGISTER,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO

P-Early-Media: Supported

P-Asserted-Identity: " " 

P-Mitrol-idLlamada: 190322160050689_MIT_07437

P-Mitrol-LoginID: 

P-Mitrol-PerfilRuteo: 100

Content-Length: 233

Content-Type: application/sdp

v=0

o=86329 -835641967 1 IN IP4 172.27.0.15

s=MitE1x Call

c=IN IP4 172.27.0.15

t=0 0

m=audio 36112 RTP/AVP 0 8 101

a=sendrecv

a=rtpmap:0 PCMU/8000/1

a=rtpmap:8 PCMA/8000/1

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

 

Reply from CUCM

---

 

SIP/2.0 488 Not Acceptable Media

Via: SIP/2.0/UDP
172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm

From: "Gabriel Querol" ;tag=2792862

To: ;tag=573234994

Date: Fri, 22 Mar 2019 19:00:23 GMT

Call-ID: 501227892-15@172.27.0.15  

CSeq: 1 INVITE

Allow-Events: presence

Warning: 304 10.4.128.27 "Media Type(s) Unavailable"

Reason: Q.850;cause=65

Content-Length: 0

 

 

 

 

SUCESSFULL CALL FROM CUCM

-

INVITE sip:*86329@172.27.0.12:506

Re: [cisco-voip] SIp Trunk call failing after PBX upgrade

2019-03-25 Thread Jonatan Quezada

we are seeing a similar issues to one of our nodes. we did our during
production, Brave but totally doable. After figuring out that we needed to
point the EM profiles to the node we were keeping up for the upgrade, we
took down the other ucs down, all went well for upgrade. All VM on my ucs
are all done now, but there is this huge jitter issues that has risen from
the ashes of the upgrade. Its as if my media RTP streams are being forked
and the forking is causing the jitter and delay?

I have calls where I lose second of audio but signaling seems fine, Im just
losing a ton of packets between the nodes now that they(the pub and sub)
are load balancing the media resources, or rather seeming to load ballance.

After some dial peer and server group re pointing, all devices finally were
on the one node and we were able to upgrade the UCS, but the other is left
to do. all of my CUCM

On Mon, Mar 25, 2019 at 10:17 AM ROZA, Ariel 
wrote:

> Hi, guys and gals.
>
>
>
> I have a customer with a CUCM 9.0(2) cluster.
>
> It is connected to a SIP PBX via a direct SIP TRUNK (No SBC, CUBE or
> otherwise). The PBX has four different nodes, all configured in the SIP
> TRUNK
>
>
>
> They claim it was working fine until last Thursday, where they did an
> upgrade to one of the nodes of the PBX. After that, calls going from PBX to
> CUCM fail with a 488 Media Not Acceptable error.
>
> They also have tried making calls from one of the not upgraded nodes, with
> the same error.
>
> I have been looking into the SIP traces, and I see nothing really telling
> of a problem there.
>
>
>
> We reseted the SIP trunk with no success.
>
> I have looked at the región configuration, and all regions are set to the
> System Default (G722, G711)
>
> I also tried changing the preferred codec in the SIP trunk, with no
> success.
>
>
>
> Following this, I am pasting the SIP messages of a failed call from PBX ->
> CUCM and a successfull call in the reverse, from CUCM -> PBX.
>
>
>
> Can you see if anything is wrong or odd?
>
>
>
> Regards,
>
>
>
> Ariel.
>
>
>
> Failed Call from PBX
>
> 
>
>
>
> INVITE sip:3366@10.4.128.27 SIP/2.0
>
> Via: SIP/2.0/UDP 172.27.0.15:11347
> ;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm
>
> From: " " ;tag=2792862
>
> To: 
>
> Call-ID: 501227892-15@172.27.0.15
>
> CSeq: 1 INVITE
>
> Contact: 
>
> Max-Forwards: 70
>
> User-Agent: MitE1x v4.4.5.1062
>
> Expires: 300
>
> Allow: INVITE,ACK,CANCEL,BYE,REGISTER,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO
>
> P-Early-Media: Supported
>
> P-Asserted-Identity: " " 
>
> P-Mitrol-idLlamada: 190322160050689_MIT_07437
>
> P-Mitrol-LoginID: 
>
> P-Mitrol-PerfilRuteo: 100
>
> Content-Length: 233
>
> Content-Type: application/sdp
>
> v=0
>
> o=86329 -835641967 1 IN IP4 172.27.0.15
>
> s=MitE1x Call
>
> c=IN IP4 172.27.0.15
>
> t=0 0
>
> m=audio 36112 RTP/AVP 0 8 101
>
> a=sendrecv
>
> a=rtpmap:0 PCMU/8000/1
>
> a=rtpmap:8 PCMA/8000/1
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-15
>
>
>
>
>
> Reply from CUCM
>
> ---
>
>
>
> SIP/2.0 488 Not Acceptable Media
>
> Via: SIP/2.0/UDP 172.27.0.15:11347
> ;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm
>
> From: "Gabriel Querol" ;tag=2792862
>
> To: ;tag=573234994
>
> Date: Fri, 22 Mar 2019 19:00:23 GMT
>
> Call-ID: 501227892-15@172.27.0.15
>
> CSeq: 1 INVITE
>
> Allow-Events: presence
>
> Warning: 304 10.4.128.27 "Media Type(s) Unavailable"
>
> Reason: Q.850;cause=65
>
> Content-Length: 0
>
>
>
>
>
>
>
>
>
> SUCESSFULL CALL FROM CUCM
>
> -
>
> INVITE sip:*86329@172.27.0.12:5060 SIP/2.0
>
> Via: SIP/2.0/UDP 10.4.128.27:5060;branch=z9hG4bK668c2eabb0b8
>
> From: "  (3307)"  >;tag=429005~b085ab57-efd9-4eb3-9a97-adbf28ac4c95-50893220
>
> To: 
>
> Date: Mon, 25 Mar 2019 10:40:36 GMT
>
> Call-ID: 6b366f80-c981b024-4f13-1b80040a@10.4.128.27
>
> Supported: timer,resource-priority,replaces
>
> Min-SE:  1800
>
> User-Agent: Cisco-CUCM9.1
>
> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
> SUBSCRIBE, NOTIFY
>
> CSeq: 101 INVITE
>
> Expires: 180
>
> Allow-Events: presence, kpml
>
> Supported: X-cisco-srtp-fallback,X-cisco-original-called
>
> Cisco-Guid: 1798729600-065536-010811-0461374474
>
> Session-Expires:  1800
>
> P-Asserted-Identity: "  (3307)" 
>
> Remote-Party-ID: "  (3307)"  >;party=calling;screen=yes;privacy=off
>
> Contact: 
>
> Max-Forwards: 69
>
> Content-Type: application/sdp
>
> Content-Length: 212
>
> v=0
>
> o=CiscoSystemsCCM-SIP 429005 1 IN IP4 10.4.128.27
>
> s=SIP Call
>
> c=IN IP4 10.4.128.12
>
> t=0 0
>
> m=audio 30530 RTP/AVP 8 101
>
> a=rtpmap:8 PCMA/8000
>
> a=ptime:20
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-15
>
>
>
>
>
> Answer from the PBX
>
> --
>
>
>
> SIP/2.0 183 Session Progress
>
> Via: SIP/2.0/UDP 10.4.128.27:5060;branch=z9hG4bK668c2eabb0b8
>
> From: "Gabriel Querol (3307)"  >;tag=429005~b085ab57-efd9-4eb3-9a97-adbf28ac4c95-50893220
>
> To: ;tag=437

Re: [cisco-voip] SIp Trunk call failing after PBX upgrade

2019-03-25 Thread ROZA, Ariel

My issue is not a CUCM upgrade. The other side from the SIP Trunk was the one 
that was updated (a local in-house development, called Mitrol). The system 
worked fine before the upgrade, and after that it went bonkers.

De: Jonatan Quezada 
Enviado el: lunes, 25 de marzo de 2019 19:24
Para: ROZA, Ariel 
CC: cisco-voip (cisco-voip@puck.nether.net) 
Asunto: Re: [cisco-voip] SIp Trunk call failing after PBX upgrade

we are seeing a similar issues to one of our nodes. we did our during 
production, Brave but totally doable. After figuring out that we needed to 
point the EM profiles to the node we were keeping up for the upgrade, we took 
down the other ucs down, all went well for upgrade. All VM on my ucs are all 
done now, but there is this huge jitter issues that has risen from the ashes of 
the upgrade. Its as if my media RTP streams are being forked and the forking is 
causing the jitter and delay?

I have calls where I lose second of audio but signaling seems fine, Im just 
losing a ton of packets between the nodes now that they(the pub and sub) are 
load balancing the media resources, or rather seeming to load ballance.

After some dial peer and server group re pointing, all devices finally were on 
the one node and we were able to upgrade the UCS, but the other is left to do. 
all of my CUCM

On Mon, Mar 25, 2019 at 10:17 AM ROZA, Ariel 
mailto:ariel.r...@la.logicalis.com>> wrote:
Hi, guys and gals.

I have a customer with a CUCM 9.0(2) cluster.
It is connected to a SIP PBX via a direct SIP TRUNK (No SBC, CUBE or 
otherwise). The PBX has four different nodes, all configured in the SIP TRUNK

They claim it was working fine until last Thursday, where they did an upgrade 
to one of the nodes of the PBX. After that, calls going from PBX to CUCM fail 
with a 488 Media Not Acceptable error.
They also have tried making calls from one of the not upgraded nodes, with the 
same error.
I have been looking into the SIP traces, and I see nothing really telling of a 
problem there.

We reseted the SIP trunk with no success.
I have looked at the región configuration, and all regions are set to the 
System Default (G722, G711)
I also tried changing the preferred codec in the SIP trunk, with no success.

Following this, I am pasting the SIP messages of a failed call from PBX -> CUCM 
and a successfull call in the reverse, from CUCM -> PBX.

Can you see if anything is wrong or odd?

Regards,

Ariel.

Failed Call from PBX


INVITE sip:3366@10.4.128.27 SIP/2.0
Via: SIP/2.0/UDP 
172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm
From: " " 
mailto:sip%3A86329@172.27.0.15>>;tag=2792862
To: mailto:sip%3A3366@10.4.128.27>>
Call-ID: 501227892-15@172.27.0.15
CSeq: 1 INVITE
Contact: 
Max-Forwards: 70
User-Agent: MitE1x v4.4.5.1062
Expires: 300
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO
P-Early-Media: Supported
P-Asserted-Identity: " " 
mailto:sip%3A86329@172.27.0.15>>
P-Mitrol-idLlamada: 190322160050689_MIT_07437
P-Mitrol-LoginID: 
P-Mitrol-PerfilRuteo: 100
Content-Length: 233
Content-Type: application/sdp
v=0
o=86329 -835641967 1 IN IP4 172.27.0.15
s=MitE1x Call
c=IN IP4 172.27.0.15
t=0 0
m=audio 36112 RTP/AVP 0 8 101
a=sendrecv
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15


Reply from CUCM
---

SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/UDP 
172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm
From: "Gabriel Querol" 
mailto:sip%3A86329@172.27.0.15>>;tag=2792862
To: mailto:sip%3A3366@10.4.128.27>>;tag=573234994
Date: Fri, 22 Mar 2019 19:00:23 GMT
Call-ID: 501227892-15@172.27.0.15
CSeq: 1 INVITE
Allow-Events: presence
Warning: 304 10.4.128.27 "Media Type(s) Unavailable"
Reason: Q.850;cause=65
Content-Length: 0




SUCESSFULL CALL FROM CUCM
-
INVITE 
sip:*86329@172.27.0.12:5060
 SIP/2.0
Via: SIP/2.0/UDP 10.4.128.27:5060;branch=z9hG4bK668c2eabb0b8
From: "  (3307)" 
mailto:sip%3A3307@10.4.128.27>>;tag=429005~b085ab57-efd9-4eb3-9a97-adbf28ac4c95-50893220
To: mailto:86329@172.27.0.12>>
Date: Mon, 25 Mar 2019 10:40:36 GMT
Call-ID: 
6b366f80-c981b024-4f13-1b80040a@10.4.128.27
Supported: timer,resource-priority,replaces
Min-SE:  1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, 
SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Cisco-Guid: 1798729600-065536-0

Re: [cisco-voip] SIp Trunk call failing after PBX upgrade

2019-03-25 Thread UC Penguin

Is there a SIP normalization profile attached to the SIP trunk used for “Failed 
Call from PBX”?

Are changes required to that profile after the remote PBX was modified?

For the “Failed Call from PBX”:
This is a SIP early offer invite. Does the CUCM trunk support early offer?

This invite has advertises it supports early media. Does the CUCM SIP trunk 
support early media?
 
There is no ptime listed in the SIP invite. How does CUCM know what ptime to 
use?

Are MTP resources available for this trunk? 

Have you pulled CallManager SDL Logs?

> On Mar 25, 2019, at 18:13, ROZA, Ariel  wrote:
> 
> My issue is not a CUCM upgrade. The other side from the SIP Trunk was the one 
> that was updated (a local in-house development, called Mitrol). The system 
> worked fine before the upgrade, and after that it went bonkers.
>  
> De: Jonatan Quezada  
> Enviado el: lunes, 25 de marzo de 2019 19:24
> Para: ROZA, Ariel 
> CC: cisco-voip (cisco-voip@puck.nether.net) 
> Asunto: Re: [cisco-voip] SIp Trunk call failing after PBX upgrade
>  
> we are seeing a similar issues to one of our nodes. we did our during 
> production, Brave but totally doable. After figuring out that we needed to 
> point the EM profiles to the node we were keeping up for the upgrade, we took 
> down the other ucs down, all went well for upgrade. All VM on my ucs are all 
> done now, but there is this huge jitter issues that has risen from the ashes 
> of the upgrade. Its as if my media RTP streams are being forked and the 
> forking is causing the jitter and delay?
>  
> I have calls where I lose second of audio but signaling seems fine, Im just 
> losing a ton of packets between the nodes now that they(the pub and sub) are 
> load balancing the media resources, or rather seeming to load ballance.
>  
> After some dial peer and server group re pointing, all devices finally were 
> on the one node and we were able to upgrade the UCS, but the other is left to 
> do. all of my CUCM 
>  
> On Mon, Mar 25, 2019 at 10:17 AM ROZA, Ariel  
> wrote:
> Hi, guys and gals.
>  
> I have a customer with a CUCM 9.0(2) cluster.
> It is connected to a SIP PBX via a direct SIP TRUNK (No SBC, CUBE or 
> otherwise). The PBX has four different nodes, all configured in the SIP TRUNK
>  
> They claim it was working fine until last Thursday, where they did an upgrade 
> to one of the nodes of the PBX. After that, calls going from PBX to CUCM fail 
> with a 488 Media Not Acceptable error.
> They also have tried making calls from one of the not upgraded nodes, with 
> the same error.
> I have been looking into the SIP traces, and I see nothing really telling of 
> a problem there.
>  
> We reseted the SIP trunk with no success.
> I have looked at the región configuration, and all regions are set to the 
> System Default (G722, G711)
> I also tried changing the preferred codec in the SIP trunk, with no success.
>  
> Following this, I am pasting the SIP messages of a failed call from PBX -> 
> CUCM and a successfull call in the reverse, from CUCM -> PBX.
>  
> Can you see if anything is wrong or odd?
>  
> Regards,
>  
> Ariel.
>  
> Failed Call from PBX
> 
>  
> INVITE sip:3366@10.4.128.27 SIP/2.0
> Via: SIP/2.0/UDP 
> 172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm
> From: " " ;tag=2792862
> To: 
> Call-ID: 501227892-15@172.27.0.15
> CSeq: 1 INVITE
> Contact: 
> Max-Forwards: 70
> User-Agent: MitE1x v4.4.5.1062
> Expires: 300
> Allow: INVITE,ACK,CANCEL,BYE,REGISTER,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO
> P-Early-Media: Supported
> P-Asserted-Identity: " " 
> P-Mitrol-idLlamada: 190322160050689_MIT_07437
> P-Mitrol-LoginID: 
> P-Mitrol-PerfilRuteo: 100
> Content-Length: 233
> Content-Type: application/sdp
> v=0
> o=86329 -835641967 1 IN IP4 172.27.0.15
> s=MitE1x Call
> c=IN IP4 172.27.0.15
> t=0 0
> m=audio 36112 RTP/AVP 0 8 101
> a=sendrecv
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
>  
>  
> Reply from CUCM
> ---
>  
> SIP/2.0 488 Not Acceptable Media
> Via: SIP/2.0/UDP 
> 172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm
> From: "Gabriel Querol" ;tag=2792862
> To: ;tag=573234994
> Date: Fri, 22 Mar 2019 19:00:23 GMT
> Call-ID: 501227892-15@172.27.0.15
> CSeq: 1 INVITE
> Allow-Events: presence
> Warning: 304 10.4.128.27 "Media Type(s) Unavailable"
> Reason: Q.850;cause=65
> Content-Length: 0
>  
>  
>  
>  
> SUCESSFULL CALL FROM CUCM
> -
> INVITE sip:*86329@172.27.0.12:5060 SIP/2.0
> Via: SIP/2.0/UDP 10.4.128.27:5060;branch=z9hG4bK668c2eabb0b8
> From: "  (3307)" 
> ;tag=429005~b085ab57-efd9-4eb3-9a97-adbf28ac4c95-50893220
> To: 
> Date: Mon, 25 Mar 2019 10:40:36 GMT
> Call-ID: 6b366f80-c981b024-4f13-1b80040a@10.4.128.27
> Supported: timer,resource-priority,replaces
> Min-SE:  1800
> User-Agent: Cisco-CUCM9.1
> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, 
> SUBSCRIBE

Re: [cisco-voip] SIp Trunk call failing after PBX upgrade

2019-03-25 Thread Brian Meade

Can you send the trace file you pulled the bad call from?

Is MTP Required set on the SIP Trunk?

On Mon, Mar 25, 2019 at 7:14 PM ROZA, Ariel 
wrote:

> My issue is not a CUCM upgrade. The other side from the SIP Trunk was the
> one that was updated (a local in-house development, called Mitrol). The
> system worked fine before the upgrade, and after that it went bonkers.
>
>
>
> *De:* Jonatan Quezada 
> *Enviado el:* lunes, 25 de marzo de 2019 19:24
> *Para:* ROZA, Ariel 
> *CC:* cisco-voip (cisco-voip@puck.nether.net) 
> *Asunto:* Re: [cisco-voip] SIp Trunk call failing after PBX upgrade
>
>
>
> we are seeing a similar issues to one of our nodes. we did our during
> production, Brave but totally doable. After figuring out that we needed to
> point the EM profiles to the node we were keeping up for the upgrade, we
> took down the other ucs down, all went well for upgrade. All VM on my ucs
> are all done now, but there is this huge jitter issues that has risen from
> the ashes of the upgrade. Its as if my media RTP streams are being forked
> and the forking is causing the jitter and delay?
>
>
>
> I have calls where I lose second of audio but signaling seems fine, Im
> just losing a ton of packets between the nodes now that they(the pub and
> sub) are load balancing the media resources, or rather seeming to load
> ballance.
>
>
>
> After some dial peer and server group re pointing, all devices finally
> were on the one node and we were able to upgrade the UCS, but the other is
> left to do. all of my CUCM
>
>
>
> On Mon, Mar 25, 2019 at 10:17 AM ROZA, Ariel 
> wrote:
>
> Hi, guys and gals.
>
>
>
> I have a customer with a CUCM 9.0(2) cluster.
>
> It is connected to a SIP PBX via a direct SIP TRUNK (No SBC, CUBE or
> otherwise). The PBX has four different nodes, all configured in the SIP
> TRUNK
>
>
>
> They claim it was working fine until last Thursday, where they did an
> upgrade to one of the nodes of the PBX. After that, calls going from PBX to
> CUCM fail with a 488 Media Not Acceptable error.
>
> They also have tried making calls from one of the not upgraded nodes, with
> the same error.
>
> I have been looking into the SIP traces, and I see nothing really telling
> of a problem there.
>
>
>
> We reseted the SIP trunk with no success.
>
> I have looked at the región configuration, and all regions are set to the
> System Default (G722, G711)
>
> I also tried changing the preferred codec in the SIP trunk, with no
> success.
>
>
>
> Following this, I am pasting the SIP messages of a failed call from PBX ->
> CUCM and a successfull call in the reverse, from CUCM -> PBX.
>
>
>
> Can you see if anything is wrong or odd?
>
>
>
> Regards,
>
>
>
> Ariel.
>
>
>
> Failed Call from PBX
>
> 
>
>
>
> INVITE sip:3366@10.4.128.27 SIP/2.0
>
> Via: SIP/2.0/UDP 172.27.0.15:11347
> ;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm
>
> From: " " ;tag=2792862
>
> To: 
>
> Call-ID: 501227892-15@172.27.0.15
>
> CSeq: 1 INVITE
>
> Contact: 
>
> Max-Forwards: 70
>
> User-Agent: MitE1x v4.4.5.1062
>
> Expires: 300
>
> Allow: INVITE,ACK,CANCEL,BYE,REGISTER,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO
>
> P-Early-Media: Supported
>
> P-Asserted-Identity: " " 
>
> P-Mitrol-idLlamada: 190322160050689_MIT_07437
>
> P-Mitrol-LoginID: 
>
> P-Mitrol-PerfilRuteo: 100
>
> Content-Length: 233
>
> Content-Type: application/sdp
>
> v=0
>
> o=86329 -835641967 1 IN IP4 172.27.0.15
>
> s=MitE1x Call
>
> c=IN IP4 172.27.0.15
>
> t=0 0
>
> m=audio 36112 RTP/AVP 0 8 101
>
> a=sendrecv
>
> a=rtpmap:0 PCMU/8000/1
>
> a=rtpmap:8 PCMA/8000/1
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-15
>
>
>
>
>
> Reply from CUCM
>
> ---
>
>
>
> SIP/2.0 488 Not Acceptable Media
>
> Via: SIP/2.0/UDP 172.27.0.15:11347
> ;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm
>
> From: "Gabriel Querol" ;tag=2792862
>
> To: ;tag=573234994
>
> Date: Fri, 22 Mar 2019 19:00:23 GMT
>
> Call-ID: 501227892-15@172.27.0.15
>
> CSeq: 1 INVITE
>
> Allow-Events: presence
>
> Warning: 304 10.4.128.27 "Media Type(s) Unavailable"
>
> Reason: Q.850;cause=65
>
> Content-Length: 0
>
>
>
>
>
>
>
>
>
> SUCESSFULL CALL FROM CUCM
>
> -
>
> INVITE sip:*86329@172.27.0.12:5060
> 
> SIP/2.0
>
> Via: SIP/2.0/UDP 10.4.128.27:5060;branch=z9hG4bK668c2eabb0b8
>
> From: "  (3307)"  >;tag=429005~b085ab57-efd9-4eb3-9a97-adbf28ac4c95-50893220
>
> To: 
>
> Date: Mon, 25 Mar 2019 10:40:36 GMT
>
> Call-ID: 6b366f80-c981b024-4f13-1b80040a@10.4.128.27
>
> Supported: timer,resource-priority,replaces
>
> Min-SE:  1800
>
> User-Agent: Cisco-CUCM9.1
>
> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
> SUBSCRIBE, N

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Re: [cisco-voip] SIp Trunk call failing after PBX upgrade

Re: [cisco-voip] SIp Trunk call failing after PBX upgrade

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Re: [cisco-voip] SIp Trunk call failing after PBX upgrade

Re: [cisco-voip] SIp Trunk call failing after PBX upgrade

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