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Today's Topics:
1. Prepend Dial Out Digit with exceptions (Matthew Loraditch)
2. Re: Weird MTP issues (Peter Slow)
3. How to globalize national calls based on calling device pool?
([email protected])
4. Re: Weird MTP issues (Kenneth Hayes)
5. Re: ITL file missing CAPF entry (Erick Wellnitz)
6. Re: Comfort noise RTP (Peter Slow)
7. 9971 SIP phone - no audio on calls between cluster to PSTN
for certain phone numbers (Erick B.)
8. Re: 9971 SIP phone - no audio on calls between cluster to
PSTN for certain phone numbers (Wes Sisk)
9. Re: 9971 SIP phone - no audio on calls between cluster to
PSTN for certain phone numbers (Erick B.)
10. Re: 9971 SIP phone - no audio on calls between cluster to
PSTN for certain phone numbers (Wes Sisk)
11. Re: 9971 SIP phone - no audio on calls between cluster to
PSTN for certain phone numbers (Wes Sisk)
12. Re: 9971 SIP phone - no audio on calls between cluster to
PSTN for certain phone numbers (Erick B.)
13. Re: 9971 SIP phone - no audio on calls between cluster to
PSTN for certain phone numbers (Erick B.)
14. Re: 9971 SIP phone - no audio on calls between cluster to
PSTN for certain phone numbers (Erick B.)
15. Re: Weird MTP issues (Mike Lydick)
16. Re: 9971 SIP phone - no audio on calls between cluster to
PSTN for certain phone numbers (Carlo)
17. Cisco Webex App on iPhone - call in issue (Scholz, Torsten)
18. Unknown caller id from incoming PSTN call (Dharambir kumar varma)
----------------------------------------------------------------------
Message: 1
Date: Mon, 12 Aug 2013 16:30:18 +0000
From: Matthew Loraditch <[email protected]>
To: "[email protected]" <[email protected]>
Subject: [cisco-voip] Prepend Dial Out Digit with exceptions
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="us-ascii"
I know how to prepend our dial digit on calls so items from missed and received
calls can be directly dialed, but if want to remove it for passed through
cfwdall and mobility calls to external numbers, where and how is the best place
to do this?
Matthew G. Loraditch - CCNP-Voice, CCNA-R&S, CCDA
1965 Greenspring Drive
Timonium, MD 21093
voice. 410.252.8830
fax. 410.252.9284
Twitter<http://twitter.com/heliontech> |
Facebook<http://www.facebook.com/#!/pages/Helion/252157915296> |
Website<http://www.heliontechnologies.com/> | Email
Support<mailto:[email protected]?subject=Technical%20Support%20Request>
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Message: 2
Date: Mon, 12 Aug 2013 12:43:57 -0400
From: Peter Slow <[email protected]>
To: Kenneth Hayes <[email protected]>
Cc: [email protected]
Subject: Re: [cisco-voip] Weird MTP issues
Message-ID:
<cama5jw5ce3opha1pogwtmjangycaemovdszy-zn-omq-v7a...@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
sounds interesting, I could take a look =)
take the MTP back out and get traces of the call failing the way it
was originally... Also, "debug ccsip mess" from the CUBE. ,,,And make
sure the CCM traces are set to detailed and that the options in both
columns are all enabled except the two labeled "SIP register'
somethingorother and skinny keepalive, please =)
-Pete
On Sun, Aug 11, 2013 at 8:47 PM, Kenneth Hayes <[email protected]> wrote:
> All,
>
>
>
> Currently I'm running CUBE in my enviroment and we have a weird issue.
> Here's my Call Flow first.
>
>
>
> PSTN->CUBE->CUCMBE->AA
>
>
>
> So when callers come in they will reach the Auto Attendant and have
> the option to press 1 for dial-by-name, well when they select option 1
> they get to the Directory Handler and speak the person they are trying
> to reach, well when the call begins to transfer it rings the extension
> for a slight second then it drops. Weird right, well on the SIP trunk
> between CUCM and CUBE I checked MTP required and saved changes and
> reset the trunk and attempt the call it works fine, the phone rings
> normal, and no issues occur. So in my SIP profile that's associated
> with the SIP trunk I checked the box to insert MTP if needed and
> unchecked the MTP required on the SIP trunk and the call drops at a
> slight second.
>
>
>
> In the CUBE I've added a voice class sip profile to see if that
> corrects it, and it has not. Can someone assist me with this issue?
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
------------------------------
Message: 3
Date: Mon, 12 Aug 2013 16:52:24 +0000
From: <[email protected]>
To: <[email protected]>
Subject: [cisco-voip] How to globalize national calls based on calling
device pool?
Message-ID:
<f8e0cc3253a10c4cb137f12f568dad061b6dd81...@gblonz-pmsgem02.emrsn.org>
Content-Type: text/plain; charset="us-ascii"
There is a large CUCM installation, with sites in various European
countries. Devices from different sites are in different Device Pools and
extensive use is
made of the Local Route Groups.
Now there is a connection to a new provider over a SIP Trunk and a CUBE.
However all calls
made to this provider are expected to be sent in +E.164 format (globalized).
There would
be also benefit in sending various national calls as well to this provider.
Current Route Pattern for national calls is:
0.0! towards Local Route Group
with H.323 gateways dropping the leading 0 and sending the call out to PSTN.
How can the dialed numbers (ex. 001234567) be globalized according to the
originating
Device's Device pool?
I.e. if a phone from Germany dials the number above, +4901234567 should be sent
to the SIP
trunk, but if a phone from the UK dials the same digits, +4401234567 should be
sent to the
SIP Trunk.
The translations could be done either on the CUCM cluster or on the CUBE,
however, cluster
based solutions would be preferred.
There is also a larger number of sites/countries involved, so some scalable
solution would
be best.
Outbound callerIDs are correctly set.
Any ideas would be appreciated :)
Zoltan Kelemen
ETS & Information Security
Implementation Engineering
Cluj-Napoca, Romania
[email protected]<mailto:[email protected]>
w: +40 374 132356
m: +40 757 039093
[cid:[email protected]]
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Message: 4
Date: Mon, 12 Aug 2013 12:48:12 -0400
From: Kenneth Hayes <[email protected]>
To: Peter Slow <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Weird MTP issues
Message-ID: <-2084884859269467938@unknownmsgid>
Content-Type: text/plain; charset=ISO-8859-1
If you like we can setup a webex.
Sent from my iPhone
On Aug 12, 2013, at 12:43 PM, Peter Slow <[email protected]> wrote:
> sounds interesting, I could take a look =)
>
> take the MTP back out and get traces of the call failing the way it
> was originally... Also, "debug ccsip mess" from the CUBE. ,,,And make
> sure the CCM traces are set to detailed and that the options in both
> columns are all enabled except the two labeled "SIP register'
> somethingorother and skinny keepalive, please =)
>
> -Pete
>
> On Sun, Aug 11, 2013 at 8:47 PM, Kenneth Hayes <[email protected]>
> wrote:
>> All,
>>
>>
>>
>> Currently I'm running CUBE in my enviroment and we have a weird issue.
>> Here's my Call Flow first.
>>
>>
>>
>> PSTN->CUBE->CUCMBE->AA
>>
>>
>>
>> So when callers come in they will reach the Auto Attendant and have
>> the option to press 1 for dial-by-name, well when they select option 1
>> they get to the Directory Handler and speak the person they are trying
>> to reach, well when the call begins to transfer it rings the extension
>> for a slight second then it drops. Weird right, well on the SIP trunk
>> between CUCM and CUBE I checked MTP required and saved changes and
>> reset the trunk and attempt the call it works fine, the phone rings
>> normal, and no issues occur. So in my SIP profile that's associated
>> with the SIP trunk I checked the box to insert MTP if needed and
>> unchecked the MTP required on the SIP trunk and the call drops at a
>> slight second.
>>
>>
>>
>> In the CUBE I've added a voice class sip profile to see if that
>> corrects it, and it has not. Can someone assist me with this issue?
>> _______________________________________________
>> cisco-voip mailing list
>> [email protected]
>> https://puck.nether.net/mailman/listinfo/cisco-voip
------------------------------
Message: 5
Date: Mon, 12 Aug 2013 12:16:02 -0500
From: Erick Wellnitz <[email protected]>
To: "Ryan Ratliff (rratliff)" <[email protected]>
Cc: cisco-voip <[email protected]>
Subject: Re: [cisco-voip] ITL file missing CAPF entry
Message-ID:
<CAK0wOsAHZsUWuiwpT3va1jDSQBq=sjr0dk8q+p_ydwrbmlu...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Thanks guys! We will be testing with the network group this week.
On Mon, Aug 12, 2013 at 10:47 AM, Ryan Ratliff (rratliff) <
[email protected]> wrote:
> CAPF-(characters) is fine, as long as the subject matches what's in the
> OS Admin Cert Manager or System->Security->Certs in CCMAdmin.
>
> -Ryan
>
> On Aug 9, 2013, at 5:16 PM, Erick Wellnitz <[email protected]>
> wrote:
>
> Version 9.1
>
> Restarting the services added the CAPF entry but the invalid certificate
> error remains. Should the CAPF entry show up as CAPF-(number) or should it
> be an actual hostname?
>
>
> On Fri, Aug 9, 2013 at 2:41 PM, Wes Sisk <[email protected]> wrote:
>
>> Seems like a common and yet intermittent issues in CUCM 8.0:
>> CSCta66968 After TVS cert is re-generated, ITL file does not contain
>> CAPF cert
>> (bug not visible in bug toolkit yet)
>>
>> <quote>
>> I see that TVS finds the CAPF cert but it does not include it in the ITL
>> File.
>> 05:10:33.859 | debug updateITLFile - Checking for ITL Related role :4,
>> Count - 1, Server - sec2-mars|*^*^*
>> 05:10:33.859 |<--debug |*^*^*
>> </quote>
>>
>> What version? Is CAPF certificate present, valid, and not expired?
>>
>> Looks like it was related to regenerating CAPF vert. Done that recently?
>>
>> restart TFTP, TVS, and capf services?
>>
>> Regards,
>> Wes
>>
>> On Aug 9, 2013, at 3:23 PM, Erick Wellnitz <[email protected]>
>> wrote:
>>
>> Strange issue.
>>
>> My ITL files do not have the CAPF entry in them. How do I get the CAPF
>> entry in the ITL file and will it cause problems for existing phones?
>>
>> Thanks!
>> _______________________________________________
>> cisco-voip mailing list
>> [email protected]
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Message: 6
Date: Mon, 12 Aug 2013 13:45:16 -0400
From: Peter Slow <[email protected]>
To: Seth McGuinness <[email protected]>
Cc: [email protected]
Subject: Re: [cisco-voip] Comfort noise RTP
Message-ID:
<CAMa5Jw7mPA5CA-4K4iMTi_uttJgNRBVgAGwGiE2=qzh+avx...@mail.gmail.com>
Content-Type: text/plain; charset=windows-1252
You need the rest of the SIP dialogue. you sent the SDP offer in the
invite, but that doesnt mean they didnt counter with different caps or
a= lines. please post everything from the INVITE though the ACK to the
200 OK. This should shed some light on the behavior.
-Peter
On Thu, Aug 8, 2013 at 6:33 AM, Seth McGuinness
<[email protected]> wrote:
> Hello All,
>
>
>
> We have a customer complaining about Comfort Noise generating from our Cisco
> AS5400, strange thing is that the packet traces suggest that the request for
> comfort noise generation comes from the client SIP invite, I can?t see where
> this is the case in the invite.
>
>
>
> Client invite
>
>
>
> No. Time Source Destination Protocol Length Info
>
> 1 2013-08-07 09:21:34.500695 10.10.1.1 192.168.1.1 SIP/SDP 1131
> Request: INVITE sip:[email protected];user=phone, with session
> description
>
>
>
> Frame 1: 1131 bytes on wire (9048 bits), 1131 bytes captured (9048 bits)
>
> Ethernet II, Src: WwPcbaTe_11:11:11 (00:0f:1f:11:11:11), Dst:
> BrocadeC_22:22:22 (00:1b:ed:22:22:22)
>
> Internet Protocol Version 4, Src: 10.10.1.1(10.10.1.1), Dst: 192.168.1.1
> (192.168.1.1)
>
> User Datagram Protocol, Src Port: sip-tls (5061), Dst Port: sip (5060)
>
> Session Initiation Protocol
>
> Request-Line: INVITE sip:[email protected];user=phone SIP/2.0
>
> Message Header
>
> Message Body
>
> Session Description Protocol
>
> Session Description Protocol Version (v): 0
>
> Owner/Creator, Session Id (o): - 1375863694 1375863694 IN IP4
> 10.10.1.1
>
> Owner Username: -
>
> Session ID: 1375863694
>
> Session Version: 1375863694
>
> Owner Network Type: IN
>
> Owner Address Type: IP4
>
> Owner Address: 10.10.1.1
>
> Session Name (s): -
>
> Connection Information (c): IN IP4 10.10.1.1
>
> Connection Network Type: IN
>
> Connection Address Type: IP4
>
> Connection Address: 10.10.1.1
>
> Time Description, active time (t): 0 0
>
> Session Start Time: 0
>
> Session Stop Time: 0
>
> Media Description, name and address (m): audio 33870 RTP/AVP 8 0
> 18 101
>
> Media Type: audio
>
> Media Port: 33870
>
> Media Protocol: RTP/AVP
>
> Media Format: ITU-T G.711 PCMA
>
> Media Format: ITU-T G.711 PCMU
>
> Media Format: ITU-T G.729
>
> Media Format: DynamicRTP-Type-101
>
> Media Attribute (a): rtpmap:8 PCMA/8000
>
> Media Attribute Fieldname: rtpmap
>
> Media Format: 8
>
> MIME Type: PCMA
>
> Sample Rate: 8000
>
> Media Attribute (a): rtpmap:0 PCMU/8000
>
> Media Attribute Fieldname: rtpmap
>
> Media Format: 0
>
> MIME Type: PCMU
>
> Sample Rate: 8000
>
> Media Attribute (a): rtpmap:18 G729/8000
>
> Media Attribute Fieldname: rtpmap
>
> Media Format: 18
>
> MIME Type: G729
>
> Sample Rate: 8000
>
> Media Attribute (a): fmtp:18 annexb=no
>
> Media Attribute Fieldname: fmtp
>
> Media Format: 18 [G729]
>
> Media format specific parameters: annexb=no
>
> Media Attribute (a): rtpmap:101 telephone-event/8000
>
> Media Attribute Fieldname: rtpmap
>
> Media Format: 101
>
> MIME Type: telephone-event
>
> Sample Rate: 8000
>
> Media Attribute (a): fmtp:101 0-15
>
> Media Attribute Fieldname: fmtp
>
> Media Format: 101 [telephone-event]
>
> Media format specific parameters: 0-15
>
> Media Attribute (a): sendrecv
>
> Media Attribute (a): silenceSupp:off - - - -
>
> Media Attribute Fieldname: silenceSupp
>
> Media Attribute Value: off - - - -
>
>
>
>
>
> Our response
>
>
>
> No. Time Source Destination
> Protocol Length Info
>
> 2 2013-08-07 09:21:41.862610 192.168.1.1 10.10.1.1 RTP
> 60 PT=Comfort noise, SSRC=0x8FED3F6, Seq=28259, Time=1374861414
>
>
>
> Frame 7: 60 bytes on wire (480 bits), 60 bytes captured (480 bits)
>
> Ethernet II, Src: BrocadeC_22:22:22 (00:1b:ed:22:22:22), Dst:
> WwPcbaTe_11:11:11 (00:0f:1f:11:11:11)
>
> Internet Protocol Version 4, Src: 192.168.1.1 (192.168.1.1), Dst:
> 10.10.1.1(10.10.1.1)
>
> User Datagram Protocol, Src Port: 19578 (19578), Dst Port: 33870 (33870)
>
> Real-Time Transport Protocol
>
> [Stream setup by SDP (frame 1)]
>
> 10.. .... = Version: RFC 1889 Version (2)
>
> ..0. .... = Padding: False
>
> ...0 .... = Extension: False
>
> .... 0000 = Contributing source identifiers count: 0
>
> 0... .... = Marker: False
>
> Payload type: Comfort noise (13)
>
> Sequence number: 28259
>
> [Extended sequence number: 93795]
>
> Timestamp: 1374861414
>
> Synchronization Source identifier: 0x08fed3f6 (150918134)
>
> Payload: 48
>
>
>
> In our RTP Comfort noise payload type 13 in frame(2) it states that the
> Stream has been setup by SDP in frame 1, I can?t see this in frame 1, it has
> silence suppression turned off and annexb=no, the RTP stream is G.711
> anyway, I left out the whole capture for brevity. Has anyone come across
> this? Any suggestions appreciated and apologies if I post this in the wrong
> place!
>
>
>
> Cheers
>
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
------------------------------
Message: 7
Date: Mon, 12 Aug 2013 14:22:12 -0500
From: "Erick B." <[email protected]>
To: voip puck <[email protected]>
Subject: [cisco-voip] 9971 SIP phone - no audio on calls between
cluster to PSTN for certain phone numbers
Message-ID:
<CAHSnBQwehJVcwRdoOCeQ-Cj5C8=NT-MNUynRhRH=rc2y9mx...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hi,
Anyone have idea here?
Have a system with 2 CUCM clusters.
Cluster 1 is CUCM version 8.6.2.22900-9 with SIP CUBE to ITSP
Cluster 2 is CUCM version 8.6.2.20000-2
ICT between the clusters (Non Gatekeeper controlled).
SCCP phones on Cluster 2 can call outbound to numbers over the ICT and out
CUBE GW fine. When a 9971 places call to same number, the call connects but
no audio is heard.
If we put the 9971 on Cluster 1 (no ICT) and same number is called it works
fine.
I've been over the release notes for 8.62a SU1 to SU3 and not finding
anything yet and have recreated the ICT and reset it. The trunk seems to be
fine since the SCCP phones work for calling the few numbers the 9971 has
trouble with. The 9971 can call most numbers just fine. 9971 is on current
firmware version to. Looking for ideas before trying to do SU3 to see if it
helps.
The number being called is a IVR with menu, etc.
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Message: 8
Date: Mon, 12 Aug 2013 15:38:52 -0400
From: Wes Sisk <[email protected]>
To: "Erick B." <[email protected]>
Cc: voip puck <[email protected]>
Subject: Re: [cisco-voip] 9971 SIP phone - no audio on calls between
cluster to PSTN for certain phone numbers
Message-ID: <[email protected]>
Content-Type: text/plain; charset=iso-8859-1
Does the IVR cut through audio before the call goes into the connected state?
-Wes
On Aug 12, 2013, at 3:22 PM, Erick B. <[email protected]> wrote:
Hi,
Anyone have idea here?
Have a system with 2 CUCM clusters.
Cluster 1 is CUCM version 8.6.2.22900-9 with SIP CUBE to ITSP
Cluster 2 is CUCM version 8.6.2.20000-2
ICT between the clusters (Non Gatekeeper controlled).
SCCP phones on Cluster 2 can call outbound to numbers over the ICT and out CUBE
GW fine. When a 9971 places call to same number, the call connects but no audio
is heard.
If we put the 9971 on Cluster 1 (no ICT) and same number is called it works
fine.
I've been over the release notes for 8.62a SU1 to SU3 and not finding anything
yet and have recreated the ICT and reset it. The trunk seems to be fine since
the SCCP phones work for calling the few numbers the 9971 has trouble with. The
9971 can call most numbers just fine. 9971 is on current firmware version to.
Looking for ideas before trying to do SU3 to see if it helps.
The number being called is a IVR with menu, etc.
_______________________________________________
cisco-voip mailing list
[email protected]
https://puck.nether.net/mailman/listinfo/cisco-voip
------------------------------
Message: 9
Date: Mon, 12 Aug 2013 14:47:17 -0500
From: "Erick B." <[email protected]>
To: Wes Sisk <[email protected]>
Cc: voip puck <[email protected]>
Subject: Re: [cisco-voip] 9971 SIP phone - no audio on calls between
cluster to PSTN for certain phone numbers
Message-ID:
<CAHSnBQyzpJ6ZC5nhRfFK34uPFRAOC14xdChs=cdcm7-3i05...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
It is an external number (not ours) and audio starts right away so
possibly. I could verify with SIP debug.
On Mon, Aug 12, 2013 at 2:38 PM, Wes Sisk <[email protected]> wrote:
> Does the IVR cut through audio before the call goes into the connected
> state?
> -Wes
>
> On Aug 12, 2013, at 3:22 PM, Erick B. <[email protected]> wrote:
>
> Hi,
>
> Anyone have idea here?
>
> Have a system with 2 CUCM clusters.
>
> Cluster 1 is CUCM version 8.6.2.22900-9 with SIP CUBE to ITSP
>
> Cluster 2 is CUCM version 8.6.2.20000-2
>
> ICT between the clusters (Non Gatekeeper controlled).
>
> SCCP phones on Cluster 2 can call outbound to numbers over the ICT and out
> CUBE GW fine. When a 9971 places call to same number, the call connects but
> no audio is heard.
> If we put the 9971 on Cluster 1 (no ICT) and same number is called it
> works fine.
>
> I've been over the release notes for 8.62a SU1 to SU3 and not finding
> anything yet and have recreated the ICT and reset it. The trunk seems to be
> fine since the SCCP phones work for calling the few numbers the 9971 has
> trouble with. The 9971 can call most numbers just fine. 9971 is on current
> firmware version to. Looking for ideas before trying to do SU3 to see if it
> helps.
>
> The number being called is a IVR with menu, etc.
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Message: 10
Date: Mon, 12 Aug 2013 15:48:12 -0400
From: Wes Sisk <[email protected]>
To: "Erick B." <[email protected]>
Cc: voip puck <[email protected]>
Subject: Re: [cisco-voip] 9971 SIP phone - no audio on calls between
cluster to PSTN for certain phone numbers
Message-ID: <[email protected]>
Content-Type: text/plain; charset="iso-8859-1"
Watch the phone UI, does audio start before the status line goes to connected?
On Aug 12, 2013, at 3:47 PM, Erick B. <[email protected]> wrote:
It is an external number (not ours) and audio starts right away so possibly. I
could verify with SIP debug.
On Mon, Aug 12, 2013 at 2:38 PM, Wes Sisk <[email protected]> wrote:
Does the IVR cut through audio before the call goes into the connected state?
-Wes
On Aug 12, 2013, at 3:22 PM, Erick B. <[email protected]> wrote:
Hi,
Anyone have idea here?
Have a system with 2 CUCM clusters.
Cluster 1 is CUCM version 8.6.2.22900-9 with SIP CUBE to ITSP
Cluster 2 is CUCM version 8.6.2.20000-2
ICT between the clusters (Non Gatekeeper controlled).
SCCP phones on Cluster 2 can call outbound to numbers over the ICT and out CUBE
GW fine. When a 9971 places call to same number, the call connects but no audio
is heard.
If we put the 9971 on Cluster 1 (no ICT) and same number is called it works
fine.
I've been over the release notes for 8.62a SU1 to SU3 and not finding anything
yet and have recreated the ICT and reset it. The trunk seems to be fine since
the SCCP phones work for calling the few numbers the 9971 has trouble with. The
9971 can call most numbers just fine. 9971 is on current firmware version to.
Looking for ideas before trying to do SU3 to see if it helps.
The number being called is a IVR with menu, etc.
_______________________________________________
cisco-voip mailing list
[email protected]
https://puck.nether.net/mailman/listinfo/cisco-voip
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Message: 11
Date: Mon, 12 Aug 2013 15:53:41 -0400
From: Wes Sisk <[email protected]>
To: "Erick B." <[email protected]>
Cc: voip puck <[email protected]>
Subject: Re: [cisco-voip] 9971 SIP phone - no audio on calls between
cluster to PSTN for certain phone numbers
Message-ID: <[email protected]>
Content-Type: text/plain; charset=iso-8859-1
Is the ICT SIP or H.323?
-wes
On Aug 12, 2013, at 3:22 PM, Erick B. <[email protected]> wrote:
Hi,
Anyone have idea here?
Have a system with 2 CUCM clusters.
Cluster 1 is CUCM version 8.6.2.22900-9 with SIP CUBE to ITSP
Cluster 2 is CUCM version 8.6.2.20000-2
ICT between the clusters (Non Gatekeeper controlled).
SCCP phones on Cluster 2 can call outbound to numbers over the ICT and out CUBE
GW fine. When a 9971 places call to same number, the call connects but no audio
is heard.
If we put the 9971 on Cluster 1 (no ICT) and same number is called it works
fine.
I've been over the release notes for 8.62a SU1 to SU3 and not finding anything
yet and have recreated the ICT and reset it. The trunk seems to be fine since
the SCCP phones work for calling the few numbers the 9971 has trouble with. The
9971 can call most numbers just fine. 9971 is on current firmware version to.
Looking for ideas before trying to do SU3 to see if it helps.
The number being called is a IVR with menu, etc.
_______________________________________________
cisco-voip mailing list
[email protected]
https://puck.nether.net/mailman/listinfo/cisco-voip
------------------------------
Message: 12
Date: Mon, 12 Aug 2013 15:28:20 -0500
From: "Erick B." <[email protected]>
To: Wes Sisk <[email protected]>
Cc: voip puck <[email protected]>
Subject: Re: [cisco-voip] 9971 SIP phone - no audio on calls between
cluster to PSTN for certain phone numbers
Message-ID:
<cahsnbqy0dm8rp9pyju9+j0rn6hgufg+m48h_gbi0cn6mcdj...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
correction to call flow, Cluster 1 has H.323 gateway with PRI (no CUBE/SIP
to provider). So the only SIP device is the 9971 phone.
On Mon, Aug 12, 2013 at 3:13 PM, Erick B. <[email protected]> wrote:
> H.323 , Inter-Cluster Trunk (Non-Gatekeeper Controlled).
>
> I can call the number from here fine on my 9971 with no issue. It'll be a
> bit before I can have someone watch the phone at that location and test
> (time difference).
>
>
> On Mon, Aug 12, 2013 at 2:53 PM, Wes Sisk <[email protected]> wrote:
>
>> Is the ICT SIP or H.323?
>>
>> -wes
>>
>> On Aug 12, 2013, at 3:22 PM, Erick B. <[email protected]> wrote:
>>
>> Hi,
>>
>> Anyone have idea here?
>>
>> Have a system with 2 CUCM clusters.
>>
>> Cluster 1 is CUCM version 8.6.2.22900-9 with SIP CUBE to ITSP
>>
>> Cluster 2 is CUCM version 8.6.2.20000-2
>>
>> ICT between the clusters (Non Gatekeeper controlled).
>>
>> SCCP phones on Cluster 2 can call outbound to numbers over the ICT and
>> out CUBE GW fine. When a 9971 places call to same number, the call connects
>> but no audio is heard.
>> If we put the 9971 on Cluster 1 (no ICT) and same number is called it
>> works fine.
>>
>> I've been over the release notes for 8.62a SU1 to SU3 and not finding
>> anything yet and have recreated the ICT and reset it. The trunk seems to be
>> fine since the SCCP phones work for calling the few numbers the 9971 has
>> trouble with. The 9971 can call most numbers just fine. 9971 is on current
>> firmware version to. Looking for ideas before trying to do SU3 to see if it
>> helps.
>>
>> The number being called is a IVR with menu, etc.
>>
>> _______________________________________________
>> cisco-voip mailing list
>> [email protected]
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>
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Message: 13
Date: Mon, 12 Aug 2013 15:13:08 -0500
From: "Erick B." <[email protected]>
To: Wes Sisk <[email protected]>
Cc: voip puck <[email protected]>
Subject: Re: [cisco-voip] 9971 SIP phone - no audio on calls between
cluster to PSTN for certain phone numbers
Message-ID:
<cahsnbqz81j6dnny0dcifs+ie-mguets4f8dvabq9j_xxbqu...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
H.323 , Inter-Cluster Trunk (Non-Gatekeeper Controlled).
I can call the number from here fine on my 9971 with no issue. It'll be a
bit before I can have someone watch the phone at that location and test
(time difference).
On Mon, Aug 12, 2013 at 2:53 PM, Wes Sisk <[email protected]> wrote:
> Is the ICT SIP or H.323?
>
> -wes
>
> On Aug 12, 2013, at 3:22 PM, Erick B. <[email protected]> wrote:
>
> Hi,
>
> Anyone have idea here?
>
> Have a system with 2 CUCM clusters.
>
> Cluster 1 is CUCM version 8.6.2.22900-9 with SIP CUBE to ITSP
>
> Cluster 2 is CUCM version 8.6.2.20000-2
>
> ICT between the clusters (Non Gatekeeper controlled).
>
> SCCP phones on Cluster 2 can call outbound to numbers over the ICT and out
> CUBE GW fine. When a 9971 places call to same number, the call connects but
> no audio is heard.
> If we put the 9971 on Cluster 1 (no ICT) and same number is called it
> works fine.
>
> I've been over the release notes for 8.62a SU1 to SU3 and not finding
> anything yet and have recreated the ICT and reset it. The trunk seems to be
> fine since the SCCP phones work for calling the few numbers the 9971 has
> trouble with. The 9971 can call most numbers just fine. 9971 is on current
> firmware version to. Looking for ideas before trying to do SU3 to see if it
> helps.
>
> The number being called is a IVR with menu, etc.
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Message: 14
Date: Mon, 12 Aug 2013 14:49:56 -0500
From: "Erick B." <[email protected]>
To: Wes Sisk <[email protected]>
Cc: voip puck <[email protected]>
Subject: Re: [cisco-voip] 9971 SIP phone - no audio on calls between
cluster to PSTN for certain phone numbers
Message-ID:
<cahsnbqw1twswodki6b_bwi+6ry2ugkgxyka+3v6pokmv-xg...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
I would see that on a SCCP phone to right? I'll have them check the 9971
(not at the location I am at).
On Mon, Aug 12, 2013 at 2:48 PM, Wes Sisk <[email protected]> wrote:
> Watch the phone UI, does audio start before the status line goes to
> connected?
>
> On Aug 12, 2013, at 3:47 PM, Erick B. <[email protected]> wrote:
>
> It is an external number (not ours) and audio starts right away so
> possibly. I could verify with SIP debug.
>
>
> On Mon, Aug 12, 2013 at 2:38 PM, Wes Sisk <[email protected]> wrote:
>
>> Does the IVR cut through audio before the call goes into the connected
>> state?
>> -Wes
>>
>> On Aug 12, 2013, at 3:22 PM, Erick B. <[email protected]> wrote:
>>
>> Hi,
>>
>> Anyone have idea here?
>>
>> Have a system with 2 CUCM clusters.
>>
>> Cluster 1 is CUCM version 8.6.2.22900-9 with SIP CUBE to ITSP
>>
>> Cluster 2 is CUCM version 8.6.2.20000-2
>>
>> ICT between the clusters (Non Gatekeeper controlled).
>>
>> SCCP phones on Cluster 2 can call outbound to numbers over the ICT and
>> out CUBE GW fine. When a 9971 places call to same number, the call connects
>> but no audio is heard.
>> If we put the 9971 on Cluster 1 (no ICT) and same number is called it
>> works fine.
>>
>> I've been over the release notes for 8.62a SU1 to SU3 and not finding
>> anything yet and have recreated the ICT and reset it. The trunk seems to be
>> fine since the SCCP phones work for calling the few numbers the 9971 has
>> trouble with. The 9971 can call most numbers just fine. 9971 is on current
>> firmware version to. Looking for ideas before trying to do SU3 to see if it
>> helps.
>>
>> The number being called is a IVR with menu, etc.
>>
>> _______________________________________________
>> cisco-voip mailing list
>> [email protected]
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>
>
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------------------------------
Message: 15
Date: Mon, 12 Aug 2013 18:21:02 -0400
From: Mike Lydick <[email protected]>
To: Kenneth Hayes <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Weird MTP issues
Message-ID:
<cafjjozd76vcz9rm_gyvcanb2oovf3qj4suq1na4kubqvfhe...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
You left out a few details of the call flow? I thought I would add some
details for the group
PSTN(Internet SIP Trunk)->CUBE->CUCMBE->AA->PSTN(Your SIP Provider)
>From the debug we took, your provider was sending back Cause Code 41
(Temporary Network Failure) with no MTP. However the call flow works if you
add MTP.
This is most likely is because the MTP will terminate the outbound call leg
from an SIP call signalling perspective with the registration address of
the MTP. This address is most likely trusted by your carrier and they
accept the call.
When you remove the MTP, the originating IP Address is passed along to your
carrier. Most carriers use the invite and re-invite information (IP
addressing in the URI) to prevent toll fraud. Hence why the use of the MTP
allows the call to work.
We tried Address-hiding with no change.
We tried sip profiles and modifying the SIP invite header with some success
but you reported that the codec was not correct on the successful call and
the quality was poor.
This does not have bearing on the use of MTP as you were using a software
MTP on successful calls. This resource will allow for terminating call
legs, dtmf mismatch or translation, packetization mismatch or translation
but not codec translation. Packet drop is a QOS issue and should look to
correcting the dial-peer and/or CUCM to get the call back to a G729r8 call
and the issue may improve but we did not look over the QOS configurations
so there maybe some room for improvement.
I think you are close to resolution. Either provide a software (IOS) mtp
for the call volume we discussed or if you the the SIP profiles are
working just should look to getting calls back to g729r8 and possible
review the QoS policy end-to-end.
Mike
Best Regards,
Mike Lydick
On Mon, Aug 12, 2013 at 12:48 PM, Kenneth Hayes <[email protected]>wrote:
> If you like we can setup a webex.
>
> Sent from my iPhone
>
> On Aug 12, 2013, at 12:43 PM, Peter Slow <[email protected]> wrote:
>
> > sounds interesting, I could take a look =)
> >
> > take the MTP back out and get traces of the call failing the way it
> > was originally... Also, "debug ccsip mess" from the CUBE. ,,,And make
> > sure the CCM traces are set to detailed and that the options in both
> > columns are all enabled except the two labeled "SIP register'
> > somethingorother and skinny keepalive, please =)
> >
> > -Pete
> >
> > On Sun, Aug 11, 2013 at 8:47 PM, Kenneth Hayes <[email protected]>
> wrote:
> >> All,
> >>
> >>
> >>
> >> Currently I'm running CUBE in my enviroment and we have a weird issue.
> >> Here's my Call Flow first.
> >>
> >>
> >>
> >> PSTN->CUBE->CUCMBE->AA
> >>
> >>
> >>
> >> So when callers come in they will reach the Auto Attendant and have
> >> the option to press 1 for dial-by-name, well when they select option 1
> >> they get to the Directory Handler and speak the person they are trying
> >> to reach, well when the call begins to transfer it rings the extension
> >> for a slight second then it drops. Weird right, well on the SIP trunk
> >> between CUCM and CUBE I checked MTP required and saved changes and
> >> reset the trunk and attempt the call it works fine, the phone rings
> >> normal, and no issues occur. So in my SIP profile that's associated
> >> with the SIP trunk I checked the box to insert MTP if needed and
> >> unchecked the MTP required on the SIP trunk and the call drops at a
> >> slight second.
> >>
> >>
> >>
> >> In the CUBE I've added a voice class sip profile to see if that
> >> corrects it, and it has not. Can someone assist me with this issue?
> >> _______________________________________________
> >> cisco-voip mailing list
> >> [email protected]
> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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Message: 16
Date: Mon, 12 Aug 2013 15:38:19 -0700
From: Carlo <[email protected]>
To: "Erick B." <[email protected]>
Cc: voip puck <[email protected]>
Subject: Re: [cisco-voip] 9971 SIP phone - no audio on calls between
cluster to PSTN for certain phone numbers
Message-ID: <[email protected]>
Content-Type: text/plain; charset="us-ascii"
are you running mulitcast?
I ran into almost the same problem. calls going from SIP to H323 back to SIP.
works find until I have to transfer or put them on hold.
When you make the call and you don't get any audio, try putting the call on
hold then pick it back up and see if you get audio.
Sent from my iPad
On Aug 12, 2013, at 12:49 PM, "Erick B." <[email protected]> wrote:
> I would see that on a SCCP phone to right? I'll have them check the 9971
> (not at the location I am at).
>
>
> On Mon, Aug 12, 2013 at 2:48 PM, Wes Sisk <[email protected]> wrote:
> Watch the phone UI, does audio start before the status line goes to connected?
>
> On Aug 12, 2013, at 3:47 PM, Erick B. <[email protected]> wrote:
>
> It is an external number (not ours) and audio starts right away so possibly.
> I could verify with SIP debug.
>
>
> On Mon, Aug 12, 2013 at 2:38 PM, Wes Sisk <[email protected]> wrote:
> Does the IVR cut through audio before the call goes into the connected state?
> -Wes
>
> On Aug 12, 2013, at 3:22 PM, Erick B. <[email protected]> wrote:
>
> Hi,
>
> Anyone have idea here?
>
> Have a system with 2 CUCM clusters.
>
> Cluster 1 is CUCM version 8.6.2.22900-9 with SIP CUBE to ITSP
>
> Cluster 2 is CUCM version 8.6.2.20000-2
>
> ICT between the clusters (Non Gatekeeper controlled).
>
> SCCP phones on Cluster 2 can call outbound to numbers over the ICT and out
> CUBE GW fine. When a 9971 places call to same number, the call connects but
> no audio is heard.
> If we put the 9971 on Cluster 1 (no ICT) and same number is called it works
> fine.
>
> I've been over the release notes for 8.62a SU1 to SU3 and not finding
> anything yet and have recreated the ICT and reset it. The trunk seems to be
> fine since the SCCP phones work for calling the few numbers the 9971 has
> trouble with. The 9971 can call most numbers just fine. 9971 is on current
> firmware version to. Looking for ideas before trying to do SU3 to see if it
> helps.
>
> The number being called is a IVR with menu, etc.
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
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Message: 17
Date: Tue, 13 Aug 2013 08:08:21 +0000
From: "Scholz, Torsten" <[email protected]>
To: "[email protected]" <[email protected]>
Subject: [cisco-voip] Cisco Webex App on iPhone - call in issue
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="us-ascii"
Hi,
I have an issue with an mobile iphone user and the webex app.
If the user wants to call into a webex via iphone app the number which was
dialed begins only with 49 w/o + (Germany).... without any DTMF parameters and
the call cannot be established.
Other user have no issues with the webex iphone app and the iphone dials
+49..... with all needed parameters to join the webex audio conference.
The iPhone version is the newest and the Webex App was also updated. New added
Webex users have no issues only this one.
Does anyone experience the same or similar issue and knows how it can be solved?
Thank you all in advance
Best Regards
Torsten
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Message: 18
Date: Tue, 13 Aug 2013 19:47:23 +0530
From: Dharambir kumar varma <[email protected]>
To: [email protected]
Subject: [cisco-voip] Unknown caller id from incoming PSTN call
Message-ID:
<ca+iwkjtpb6ryfsed2dk8ugm5gvquqqzz-_gwnwduou78g+g...@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
Hi
My cisco ip phone is displaying Unknown caller id when PSTN calls come
via MGCP gateway..
Can anyone share idea..
--
Regards,
Dharambir Kumar
------------------------------
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------------------------------
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