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Today's Topics:
1. Re: Call drop issue (Nick Matthews)
2. Can a CUBE be used to connect CUCM and Asterisk to a SIP
Trunk (Dieter Jansen)
3. Re: alert when PRI goes down (Shaihan Jaffrey)
4. Re: Can a CUBE be used to connect CUCM and Asterisk to a SIP
Trunk (Matthew Saskin)
----------------------------------------------------------------------
Message: 1
Date: Sat, 1 Dec 2012 12:16:34 -0500
From: Nick Matthews <[email protected]>
To: John Van Laecke <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Call drop issue
Message-ID:
<cam-k-nokcip7u-j3q0wlmnd53mkd0ocftuxndnb4wqsuq_n...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Packet capture looks normal, it's a bit odd that Avaya continually sends
empty H323 messages, but that doesn't appear to be root of the problem.
Debugs you sent seem to show a TCP FIN message coming from Avaya for the
call drop.
-nick
On Wed, Nov 28, 2012 at 10:09 PM, John Van Laecke <[email protected]>wrote:
> This normally happens when the router can't see the call manager.
>
> Or you are missing the h323 bindings.
>
>
> -----Original Message-----
> From: [email protected] [mailto:
> [email protected]] On Behalf Of Kenneth Hayes
> Sent: Thursday, 29 November 2012 6:12 AM
> To: Michele Russo (AM)
> Cc: [email protected]
> Subject: Re: [cisco-voip] Call drop issue
>
> Do you have the proper codecs in place?
>
> Sent from my iPhone
>
> On Nov 28, 2012, at 2:08 PM, "Michele Russo (AM)"
> <[email protected]> wrote:
>
> > All,
> >
> > I am working with a customer who is seeing sporadic call drops on their
> systems. They following is the setup:
> >
> > CME - H323 Trunk to Avaya - call routed out a LD PRI registered to the
> Avaya system.
> >
> > The CME has 12 7937's and 4 biamp devices registered to it, the Biamps
> are SIP based phones. The off-net calls drop sporadically and the
> seemingly relevant messages we see in the Wire Shark and Syslog traces
> show:
> >
> >
> > 10.146.128.50 (Avaya) 10.146.149.65 TCP [TCP ACKed lost
> segment] h323hostcall > hfcs-manager [ACK] Seq=1 Ack=2 Win=8736 Len=0
> > TCP0: bad seg from 10.146.128.50 -- bad sequence number: port 35839 seq
> 2919846836 ack 1534185920 rcvnxt 2919846837 rcvwnd 3146 len 0
> > TCP0: ACK timeout timer expired
> >
> > With a disconnect Cause Value=38 (meaning network out of order).
> >
> > I am attaching the router config, the syslog file and the Wireshark
> capture.
> >
> > Note: the CME and the Avaya re on the same 4507E switch and a review of
> both interfaces show literally zero CRC, Frame, Overrun or other L2 issues.
> So it seems unlikely that there are any cabling issues, and the switch
> itself seems quite healthy.
> >
> > Any thoughts/suggestions would be very helpful! So far TAC has
> suggested we add the 'no vad' statement to the dial-peer 11 and
> allow-connections h323 to h323. I am not sure either one of those
> configuration changes will fix this issue.
> >
> > Thanks!
> >
> > Michele Russo
> > Consultant
> > Dimension Data NA
> > 202-460-3965 (cell)
> > 571-203-4007 (desk)
> > [email protected]
> > <mime-attachment>
> > <dropped conf call.pcap>
> > <SyslogCatchAll.txt>
> > <WASWAN1_show_run.11_26_12.txt>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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Message: 2
Date: Sun, 2 Dec 2012 16:10:03 +1030
From: Dieter Jansen <[email protected]>
To: [email protected]
Subject: [cisco-voip] Can a CUBE be used to connect CUCM and Asterisk
to a SIP Trunk
Message-ID:
<CAFdJyunwzHLHMdQ_0YOXQ=hvyirf3tbuvgqahvdh-m2tkmm...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hi All,
We are looking to replace an existing Asterisk PBX with about 40 SIP
handsets with a CUCM UC solution (BE6K) with about 50 new CP-8945s.
The current installation uses a SIP trunk and we plan to keep using this to
connect to the PSTN. We have a 100 DID block for our incoming calls.
As we have comparatively little experience with the Cisco solution we're
looking at a soft introduction and will probably have both sets of handsets
in place in parallel for a while.
I've got all the edge switches and VLANing working - independent data,
voice and legacy voice VLANs/subnets and all required routing between.
We're at the point where test phones can place video calls to each other
so we now need to get routing and PSTN access happening
Originally I was thinking I'd set up a CUCM-behind-Asterisk
or Asterisk behind-CUCM situation whilst testing but on reading the Cisco
documents and books I've noted the suggested use of a CUBE SBC. As we have
a spare router that can probably do CUBE I was wondering if I should set
that up to register with our VSP.
In the first instance I'd get the Asterisk box to pass calls to the CUBE
and have the CUBE forward all incoming calls to the Asterisk.
What I'm hoping is that I could then also have the CUCM pass calls to the
same CUBE and route a subset of the inward DIDs to the CUCM whilst testing
- allowing me to get all the setup sorted whilst easily flipping incoming
calls to specific DID between Asterisk and CUCM.
I've not seen an example of the CUBE being used in this way - is it
possible?
Regards, JustTemprament.
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Message: 3
Date: Sun, 2 Dec 2012 11:32:03 +0500
From: Shaihan Jaffrey <[email protected]>
To: Tim Reimers <[email protected]>
Cc: Cisco VOIP <[email protected]>
Subject: Re: [cisco-voip] alert when PRI goes down
Message-ID:
<capxwygrofayt+eetme_qqvjgc_zgjaetpnhkw2n6bbtq9ch...@mail.gmail.com>
Content-Type: text/plain; charset="windows-1252"
Thanks!
On Fri, Nov 30, 2012 at 7:16 PM, Tim Reimers <[email protected]>wrote:
> Here?s a few sample EEM scripts:****
>
> ** **
>
> Notice that EEM is watching for specific syslog messages****
>
> **- ** for the T1 controller****
>
> **- **for a Serial Interface changing state****
>
> **- **for EIGRP creating/dropping ajacencies****
>
> ** **
>
> I usually do an applet for down and up, or recovery and failure..****
>
> That way you know when something was brief?****
>
> ** **
>
> I?m running 12.3 and 12.4 code on these guys? older stuff, so EEM should
> be available to you.****
>
> You may have to enable some stuff under SNMP traps to start seeing it in
> buffered logfiles on the routers for EEM to catch it.****
>
> ** **
>
> Start with just enabling the SNMP in local logging, watch for the messages
> you want to trigger an EEM on, and go from there.****
>
> ** **
>
> ** **
>
> event manager applet T1Control_0-0-0_Down****
>
> event syslog pattern "%CONTROLLER-5-UPDOWN: Controller T1 0/0/0, changed
> state to down"****
>
> action email mail server "my-mailservers-ip-addr" to "
> [email protected]" from "
> [email protected]" subject "T1 controller 0/0/0
> on C2821 10.53.2.5 is DOWN " body "The T1 controller 0/0/0 has gone down on
> 10.53.2.5"****
>
> ** **
>
> event manager applet T1Control_0-0-0_Up****
>
> event syslog pattern "%CONTROLLER-5-UPDOWN: Controller T1 0/0/0, changed
> state to up"****
>
> action email mail server "my-mailservers-ip-addr" to "
> [email protected]" from "
> [email protected]" subject "T1 controller 0/0/0
> on C2821 10.53.2.5 is BACK UP " body "The T1 controller 0/0/0 is up up on
> 10.53.2.5"****
>
> ** **
>
> event manager applet RemoteSite-EIGRP-up****
>
> event syslog pattern "%DUAL-5-NBRNetworkCoreANGE: IP-EIGRP(0) 12: Neighbor
> 10.25.43.5 (Serial0/3/0) is up: new adjacency"****
>
> action email mail server "your-mailserver-addr" to "
> [email protected]" from "[email protected]"
> subject "EIGRP adjacency created between NetworkCore and RemoteSite" body
> "The EIGRP adjacency between NetworkCore and RemoteSite has been created"*
> ***
>
> ** **
>
> event manager applet MySite-Serial010-down****
>
> event syslog pattern "%LINK-3-UPDOWN: Interface Serial0/1/0,
> NetworkCoreanged state to down"****
>
> action email mail server "your-mailserver-addr" to "
> [email protected],[email protected]" from "
> [email protected]" subject "Interface Serial0/1/0,
> NetworkCoreanged state to down - MY SITE T1 DOWN" body "Ser0/1/0 -- MY
> SITE down"****
>
> ** **
>
> ** **
>
> Tim Reimers****
>
> Systems Analyst II****
>
> Information Technology Services****
>
> City of Asheville****
>
> 70 Court Plaza****
>
> Asheville, NC 28801****
>
> Direct phone - 828-259-5512****
>
> IT Services main front desk - 828-259-5510****
>
> [email protected] <[email protected]>****
>
> ** **
>
> ** **
>
> ** **
>
> *From:* [email protected] [mailto:
> [email protected]] *On Behalf Of *Shaihan Jaffrey
> *Sent:* Thursday, November 29, 2012 10:01 AM
> *To:* Erick B.
> *Cc:* Cisco VOIP
> *Subject:* Re: [cisco-voip] alert when PRI goes down****
>
> ** **
>
> Yes PRI setup for MGCP.****
>
> ** **
>
> On Wed, Nov 28, 2012 at 11:26 PM, Erick B. <[email protected]> wrote:****
>
> The RTMT MGCP alert will only be useful if your PRI is setup for MGCP. If
> it is H323 then you need to use syslogs, snmp traps. You could also put
> together a EEM script that runs on the router to email you. ****
>
> ** **
>
> On Wed, Nov 28, 2012 at 1:20 AM, Abdul Salam . <[email protected]> wrote:*
> ***
>
> see attached counter****
>
>
>
>
> *---AS*****
>
>
>
>
>
> ****
>
> On Wed, Nov 28, 2012 at 12:42 PM, Abdul Salam . <[email protected]> wrote:
> ****
>
> I think you can look for RTMT alert D-channel oos ****
>
>
>
>
> *---AS*****
>
>
>
>
>
> ****
>
> On Wed, Nov 28, 2012 at 11:14 AM, Lelio Fulgenzi <[email protected]>
> wrote:****
>
> Both the router and CUCM will send syslog messages to this affect. Once
> you configure syslog target, you can setup your syslog host with the
> appropriate software to send email. We use SEC in conjunction with a Linux
> based syslog daemon. There are other options like Solarwinds and Logzilla.
>
> I also believe RTMT can be configured to send email based on thresholds.
>
> Sent from my iPhone...
>
> "There's no place like 127.0.0.1"****
>
>
> On Nov 28, 2012, at 12:33 AM, Shaihan Jaffrey <[email protected]> wrote:
>
> > Hi Team,
> > Is there any way to generate an alert on a specific email as soon as
> the PRI goes down in cisco voice gateway.
> >
> > Regards,
> >****
>
> > _______________________________________________
> > 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****
>
> ** **
>
> ** **
>
>
> _______________________________________________
> 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: 4
Date: Sun, 2 Dec 2012 10:33:21 -0500
From: Matthew Saskin <[email protected]>
To: Dieter Jansen <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Can a CUBE be used to connect CUCM and
Asterisk to a SIP Trunk
Message-ID:
<camsv-mtsmor9x26zvsbdcje_xd1dvoojrhan8ho8qcnbqtj...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
This would not be a problem at all. The short answer is you'd end up
having dial peers on the CUBE that route (via destination pattern, etc.)
specific calls/DID's to the asterisk IP address and specific calls/DID's to
the be6k address.
-matthew
Matthew Saskin
[email protected]
203-253-9571
On Sun, Dec 2, 2012 at 12:40 AM, Dieter Jansen <[email protected]>wrote:
> Hi All,
>
> We are looking to replace an existing Asterisk PBX with about 40 SIP
> handsets with a CUCM UC solution (BE6K) with about 50 new CP-8945s.
>
> The current installation uses a SIP trunk and we plan to keep using this
> to connect to the PSTN. We have a 100 DID block for our incoming calls.
>
> As we have comparatively little experience with the Cisco solution we're
> looking at a soft introduction and will probably have both sets of handsets
> in place in parallel for a while.
>
> I've got all the edge switches and VLANing working - independent data,
> voice and legacy voice VLANs/subnets and all required routing between.
> We're at the point where test phones can place video calls to each other
> so we now need to get routing and PSTN access happening
>
> Originally I was thinking I'd set up a CUCM-behind-Asterisk
> or Asterisk behind-CUCM situation whilst testing but on reading the Cisco
> documents and books I've noted the suggested use of a CUBE SBC. As we have
> a spare router that can probably do CUBE I was wondering if I should set
> that up to register with our VSP.
>
> In the first instance I'd get the Asterisk box to pass calls to the CUBE
> and have the CUBE forward all incoming calls to the Asterisk.
>
> What I'm hoping is that I could then also have the CUCM pass calls to the
> same CUBE and route a subset of the inward DIDs to the CUCM whilst testing
> - allowing me to get all the setup sorted whilst easily flipping incoming
> calls to specific DID between Asterisk and CUCM.
>
> I've not seen an example of the CUBE being used in this way - is it
> possible?
>
> Regards, JustTemprament.
>
>
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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End of cisco-voip Digest, Vol 110, Issue 2
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