[cisco-voip] 7821 phones issue

Ahmed Abd EL-Rahman Ahmed.Rahman at bmbgroup.com
Thu Oct 29 08:34:00 EDT 2015


Many Thanks Brian, stripping out the authorization header fixed the issue.

Really appreciated.



Best Regards

Ahmed Abd EL-Rahman
Senior Network Engineer - KSA


From: bmeade90 at gmail.com [mailto:bmeade90 at gmail.com] On Behalf Of Brian Meade
Sent: Thursday, October 29, 2015 12:21 AM
To: Ahmed Abd EL-Rahman <Ahmed.Rahman at bmbgroup.com>
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 7821 phones issue

You could try a sip-profile under voice service voip to strip it out.

voice class sip-profiles 100
 request Invite sip-header Authorization remove

voice service voip
 sip
  sip-profiles 100

I've never tried this for CME messaging though.  It's normally a CUBE feature.  It may still work though.

You should be able to remove the username/password under the SIP ephone configuration for the calling phone if it's on the same subnet as CME which may also help but doesn't help if you have those 78XX phones registering over the WAN.


On Wed, Oct 28, 2015 at 3:41 PM, Ahmed Abd EL-Rahman <Ahmed.Rahman at bmbgroup.com<mailto:Ahmed.Rahman at bmbgroup.com>> wrote:
Nothing configured under the sip-ua.

How can I strip out the authorization header please?





Best Regards

Ahmed Abd EL-Rahman
Senior Network Engineer - BMB KSA

On Oct 28, 2015, at 7:29 PM, Brian Meade <bmeade90 at vt.edu<mailto:bmeade90 at vt.edu><mailto:bmeade90 at vt.edu<mailto:bmeade90 at vt.edu>>> wrote:

I'm guessing there's a device with SIP inspection in the middle that doesn't like the Authorization header and is dropping the packets.  You could either go through the full path and fix that issue or somehow strip out that Authorization header.

What do you have configured under sip-ua?

On Wed, Oct 28, 2015 at 11:49 AM, Ahmed Abd EL-Rahman <Ahmed.Rahman at bmbgroup.com<mailto:Ahmed.Rahman at bmbgroup.com><mailto:Ahmed.Rahman at bmbgroup.com<mailto:Ahmed.Rahman at bmbgroup.com>>> wrote:
Hi Brian,

If this is the case, so how this can be fixed ?

Also if this is what makes the call fail, how these calls are working between the 7821 phones on the same LAN (both HQ and Branch)? Also it’s working from branch 7821 to HQ 7821, it just not working from HQ 7821 to branch 7821.



Best Regards

Ahmed Abd EL-Rahman
Senior Network Engineer - KSA


From: bmeade90 at gmail.com<mailto:bmeade90 at gmail.com><mailto:bmeade90 at gmail.com<mailto:bmeade90 at gmail.com>> [mailto:bmeade90 at gmail.com<mailto:bmeade90 at gmail.com><mailto:bmeade90 at gmail.com<mailto:bmeade90 at gmail.com>>] On Behalf Of Brian Meade
Sent: Wednesday, October 28, 2015 6:43 PM
To: Ahmed Abd EL-Rahman <Ahmed.Rahman at bmbgroup.com<mailto:Ahmed.Rahman at bmbgroup.com><mailto:Ahmed.Rahman at bmbgroup.com<mailto:Ahmed.Rahman at bmbgroup.com>>>
Cc: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net><mailto:cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>>
Subject: Re: [cisco-voip] 7821 phones issue

In the non-working Invite, CME is sending an Authorization header that doesn't exist in the working case:
Authorization: Digest username="204",realm="all",uri="sip:201 at 10.10.3.2<mailto:sip%3A201 at 10.10.3.2><mailto:sip%3A201 at 10.10.3.2<mailto:sip%253A201 at 10.10.3.2>>;user=phone",response="13d0e80e52a41bc8642290d5744ca66d",nonce="9D2BD5F30012BAFA",cnonce="570b883d",qop=auth,nc=00000003,algorithm=MD5

This is forwarded from the incoming Invite from the 7821.  I'm guessing you have some sort of header-passthrough configured here resulting in the problem occurring.




On Wed, Oct 28, 2015 at 8:07 AM, Ahmed Abd EL-Rahman <Ahmed.Rahman at bmbgroup.com<mailto:Ahmed.Rahman at bmbgroup.com><mailto:Ahmed.Rahman at bmbgroup.com<mailto:Ahmed.Rahman at bmbgroup.com>>> wrote:
I know that there is a lack of a response to the SIP invite message which is being sent from HQ 7821 phone to the branch 7821 phone during the call setup which may point to a network connectivity (or blocking) in the middle, but can anyone give me a reasonable answer for why the same messages are flowing normally when the 8831 HQ phone initiate the call to the 7821 branch phone ?? if something blocked it will be blocked for all.



Best Regards

Ahmed Abd EL-Rahman
Senior Network Engineer - KSA

From: cisco-voip [mailto:cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at puck.nether.net><mailto:cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at puck.nether.net>>] On Behalf Of Ahmed Abd EL-Rahman
Sent: Wednesday, October 28, 2015 3:02 PM
To: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net><mailto:cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>>
Subject: [cisco-voip] 7821 phones issue

Hi Gents,


I have a setup of CME 10.5 on 2911 ISR in the HQ, then I have a DMVPN to the branch site.

I have the following phone types:

- 7945 skinny based phone.

- 8831 SIP conference station.

-7821 SIP phones.

- 7861 SIP phone.

the setup is working fine in the HQ (on the same LAN) but when i took some of the 7821 phones and placed them in the branch site the following happens:

- They are able to register to the CME in the HQ.

- Branch phones can call all HQ phones and can also call each others’ in the branch normally.

- The issue is that HQ 7821 & 7861 phones cannot call 7821 branch phones (no ring back is heard then the call disconnects after around 1 minute), despite the fact that HQ 7945 and 8831 (which is also SIP based) can call Branch 7821 phones normally with no issues ???



I have upgraded the router IOS to the latest safe harbor version 15.4(3)M4 and upgraded the 7821 firmware to the latest version 10.3.1 with no success.

I have enabled the debug voice ccapi inout and collected the output from 2 calls, 1 is from HQ 8831 SIP phone to a branch 7821 phone which was successful and the other from HQ 7821 phone to a branch 7821 phone which was unsuccessful and i compared the two outputs line by line and discovered that in the successful call the alerting singal (ring back) is received normally, but in the unsuccessful one it's not received and a disconnect signal is received as shown below:

- successful call:

*Oct 25 10:43:55.258: cc_api_get_xcode_stream : 4982
*Oct 25 10:43:55.258: //678/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:

*Oct 25 10:43:55.258: cc_api_get_xcode_stream : 4982
*Oct 25 10:43:55.258: //678/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:

*Oct 25 10:43:55.258: cc_api_get_xcode_stream : 4982
*Oct 25 10:43:55.258: //678/1F64117E85C0/CCAPI/cc_api_call_proceeding:
   Interface=0x3E81FC6C, Progress Indication=NULL(0)
*Oct 25 10:43:55.466: //678/1F64117E85C0/CCAPI/cc_api_call_alert:
   Interface=0x3E81FC6C, Progress Indication=NULL(0), Signal Indication=SIGNAL RINGBACK(1)
*Oct 25 10:43:55.466: //678/1F64117E85C0/CCAPI/cc_api_call_alert:
   Call Entry(Retry Count=0, Responsed=TRUE)
*Oct 25 10:43:55.470: //677/1F64117E85C0/CCAPI/ccCallAlert:
   Progress Indication=NULL(0), Signal Indication=SIGNAL RINGBACK(1)
*Oct 25 10:43:55.470: //677/1F64117E85C0/CCAPI/ccCallAlert:
   Call Entry(Responsed=TRUE, Alert Sent=TRUE)
*Oct 25 10:43:55.470: //677/1F64117E85C0/CCAPI/ccCallNotify:
   Data Bitmask=0x7, Call Id=677



- unsuccessful call:

*Oct 25 08:18:21.359: cc_api_get_xcode_stream : 4982
*Oct 25 08:18:21.359: //357/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:

*Oct 25 08:18:21.359: cc_api_get_xcode_stream : 4982
*Oct 25 08:18:21.359: //357/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:

*Oct 25 08:18:21.359: cc_api_get_xcode_stream : 4982
*Oct 25 08:18:21.359: //357/C9AD33DD82E1/CCAPI/cc_api_call_proceeding:
   Interface=0x3E81FC6C, Progress Indication=NULL(0)

*Oct 25 08:19:24.863: //357/C9AD33DD82E1/CCAPI/cc_api_call_disconnected:
   Cause Value=102, Interface=0x3E81FC6C, Call Id=357
*Oct 25 08:19:24.863: //357/C9AD33DD82E1/CCAPI/cc_api_call_disconnected:
   Call Entry(Responsed=TRUE, Cause Value=102, Retry Count=0)
*Oct 25 08:19:24.863: //356/C9AD33DD82E1/CCAPI/ccCallReleaseResources:
   release reserved xcoding resource.
*Oct 25 08:19:24.863: //357/C9AD33DD82E1/CCAPI/ccCallSetAAA_Accounting:
   Accounting=0, Call Id=357
*Oct 25 08:19:24.863: //357/C9AD33DD82E1/CCAPI/ccCallDisconnect:
   Cause Value=102, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=102)
*Oct 25 08:19:24.863: //357/C9AD33DD82E1/CCAPI/ccCallDisconnect:
   Cause Value=102, Call Entry(Responsed=TRUE, Cause Value=102)
*Oct 25 08:19:24.863: //357/C9AD33DD82E1/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x3E81FC6C, Tag=0x0, Call Id=357,
   Call Entry(Disconnect Cause=102, Voice Class Cause Code=0, Retry Count=0)
*Oct 25 08:19:24.863: //357/C9AD33DD82E1/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
*Oct 25 08:19:24.863: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

*Oct 25 08:19:24.863: :cc_free_feature_vsa freeing 24DE2870
*Oct 25 08:19:24.863: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

*Oct 25 08:19:24.863:  vsacount in free is 1
*Oct 25 08:19:24.863: //356/C9AD33DD82E1/CCAPI/ccCallDisconnect:
   Cause Value=102, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
*Oct 25 08:19:24.863: //356/C9AD33DD82E1/CCAPI/ccCallDisconnect:
   Cause Value=102, Call Entry(Responsed=TRUE, Cause Value=102)
*Oct 25 08:19:24.871: //356/C9AD33DD82E1/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x3E81FC6C, Tag=0x0, Call Id=356,
   Call Entry(Disconnect Cause=102, Voice Class Cause Code=0, Retry Count=0)
*Oct 25 08:19:24.871: //356/C9AD33DD82E1/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
*Oct 25 08:19:24.871: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

*Oct 25 08:19:24.871: :cc_free_feature_vsa freeing 24DE25D0
*Oct 25 08:19:24.871: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:



Also please find the debug ccsip message output for both successful call and unsuccessful call attached.

So any ideas please ?




Best Regards

Ahmed Abd EL-Rahman
Senior Network Engineer - KSA


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net><mailto:cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>>
https://puck.nether.net/mailman/listinfo/cisco-voip


-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20151029/129347ca/attachment.html>


More information about the cisco-voip mailing list