I’m confused as to why this is so difficult. Below is my post to the Cisco NetPro forum. I’m hoping they’ll be able to help eventually, but I thought I’d include it here for the record. I know a few Cisco schmartypantsh people whom might read this.

Hi all,

I am wondering if someone may be able to point out where I’m going wrong.

I want to be able to see/user the boot cycle/ROMMON mode of a 2811 router, remotely.

As I understand it, the only way to achieve this is with an analogue modem connected to the Console port.

I’m doing the testing here with a Cisco 2651 as it’s the only router I have available to me locally, though as most of the configuration is modem-side, I feel that it shouldn’t make much of a difference..

I’m using a Hayes External Serial modem, connected to the Console port with a Cisco-provided 25pin D -> RJ-45 roll-over.

However, after using this guide, I’ve not had any success. The basic principles appear the same as with a router, but when I dial in, the modems handshake and connect but HyperTerminal displays nothing. There is no output at all UNLESS I enable the ‘post-dial window’, and then I can control the console. My issue is that this is not a native console – there’s no xmodem support and I wouldn’t have a clue how to duplicate this work-around in another client (minicom or screen, for example.)

I’ve noticed that the S37=9 (listed in the guide as the setting for 9600kbps) is not a value that is saved in the current profile. Perhaps the Hayes set has been updated since the article was written. Does anyone know a better way to set this? I have found a way of forcing V.32, which is 9600, but .. is it correct?

Here is the current profile stored on the modem:

ACTIVE PROFILE:
B1 E1 L1 M1 N0 Q0 T V1 W2 X4 Y0 &C1 &D0 &G0 &J0 &K0 &Q5 &R1 &S0 &T5 &X0 &Y0
S00:001 S01:000 S02:043 S03:013 S04:010 S05:008 S06:002 S07:050 S08:002 S09:006
S10:014 S11:085 S12:050 S18:000 S25:005 S26:001 S36:007 S38:020 S46:138 S48:007
S95:000

(Note that I set ATE0 and ATQ1 when testing, but these were flipped for the purpose of viewing the config )

Here is the config from the router:

line con 0
exec-timeout 5 0
logging synchronous

I *think* this is a problem with the speed not detecting the speed of the console properly, but I’m not experienced enough with modems at this level. I’ve even consulted my CCNP2 book, only to find that it concentrated more upon the Aux port.

I can get modem -> Aux working, but that’s just not flexible enough (despite its advantages.)

If anyone can help, I’d be extremely grateful. If you require any more information, I’ll be happy to provide it!

Thanks,

Tom

If anyone has any insight into why HyperTerminal is behaving in such a fashion, I’d love to know.

Update: On my last trip to the hosting facility where this router’s stored, I took the opportunity to test the theory of a non-Cisco CF card knobbling the successful restart of a router. I was sure I’d checked already, but I wanted to be sure.

Thankfully, it restarted without a hitch. So I guess there’s zero cause for alarm.

I know I’ve not written on this log in a long while; I’ve been so busy following the departure of my predecessor, that it’s left me with little time (or will) to write about technology. More to come on this though. I’m still meant to be using this blog as a placement log, so I should really document what I’m doing.

Anyhow, as it happens I’m sat with very little to do at the minute, so I thought I’d write about something that really got me annoyed a week or so ago.

OK, so anyone who’s had the pleasure of setting up a VoIP system based around Cisco CME will know that the flash: directory rapidly fills up with a tonne of files. Config files, language files, firmware files… Basically; a lot of individual little files inhabit your CF card.

Now, on a recent trip to our co-location facility of choice, I wanted to take the opportunity to upgrade the flash card in our voice router. This is a 2811, with the basic 64MB CF card pre-installed, and recently it’s become a pain in the arse working with newer phones (7941′s, in my case) because the firmware files were just taking up too much room. Not to mention you’re only left with ~14MB free space after uploading the IOS image.

So I purchased a 256MB CF card, for about £8 inc. delivery (a mere shadow of the £202-400+ that Cisco wanted for a god-damn CF card of the same size!) and grabbed a standard-affair USB flash drive to use as temporary storage whilst shifting files from one card to the other.

Now, when you’re deleting files en masse from a Cisco flash: directory, it’s possible to use wildcards in order to catch multiple, similarly-named files and delete them all in turn. Which is quite useful if you’re working with 20 config files that all begin with ‘SEPSomethingOrOther’. So because of this included [20th Century] functionality, one would also assume that Cisco would have also included the use of wildcards into the copy command…

Have they bollocks!


RouterX# copy flash:its\SEP* usbflash0:
LOL * IS NOT A VALID FILENAME CHARACTER



RouterX# copy flash:its usbflash0:
LOL CAN'T COPY FOLDERS!


So not only is it retarded enough to not recognise ‘*’ as ‘please copy anything that begins with the preceeding phrase’, it’s also completely inable to copy an entire directory! I cannot believe that Cisco are up to release 12.4(15) and haven’t included some way of copying files en-masse.

It took me bloody ages to copy each of the 62 files from flash: to usbflash0:, swap the CF cards over, and then copy 62 files back to the new CF card. What a horrid waste of time, Cisco.

My good friends at Cisco: if you’re reading this, for the sake of all that’s useful, bloody-well sort it out!

The phone registers fine and can make external calls via the SIP trunk absolutely fine. The phone will also accept transfers of internal calls (ephones), but if I attempt to transfer an external call form an ephone, to the Granstream – the external call is cut-off. I’ve isolated the ‘debug ccsip messages’ output which describes what is happening, but I’m by no means an expert in debugging SIP output.

Aug 14 09:16:49.807: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REFER sip:07xxxxxxxxx@cme-router:5060 SIP/2.0
Via: SIP/2.0/UDP cme-router:5060;branch=z9hG4bK3432633
From: ;tag=2CD733C8-216A
To: "Tom" ;tag=as221e2abe
Call-ID: 24524ebd58e984e930f461ff5d490bd9@asterisk-sip-gateway
CSeq: 102 REFER
Max-Forwards: 70
Contact:
User-Agent: Cisco-SIPGateway/IOS-12.x
Timestamp: 1187083009
Refer-To: sip:2007@cme-router?Replaces=E72682C6-497D11DC-90D890FC-79F613F8%40cme-router
%3Bto-tag%3Ddb0457357be0d469%3Bfrom-tag%3D2CD77068-25C3
Referred-By:
Content-Length: 0


Aug 14 09:16:49.811: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP cme-router:5060;branch=z9hG4bK3432633;received=cme-router
From: ;tag=2CD733C8-216A
To: "Tom" ;tag=as221e2abe
Call-ID: 24524ebd58e984e930f461ff5d490bd9@asterisk-sip-gateway
CSeq: 102 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Length: 0


Aug 14 09:16:49.811: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
NOTIFY sip:0116@cme-router:5060 SIP/2.0
Via: SIP/2.0/UDP asterisk-sip-gateway:5060;branch=z9hG4bK49b17aad;rport
From: "Tom" ;tag=as221e2abe
To: ;tag=2CD733C8-216A
Contact:
Call-ID: 24524ebd58e984e930f461ff5d490bd9@asterisk-sip-gateway
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: refer;id=102
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 49


SIP/2.0 481 Call leg/transaction does not exist


Aug 14 09:16:49.815: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP asterisk-sip-gateway:5060;branch=z9hG4bK49b17aad;rport
From: "Tom" ;tag=as221e2abe
To: ;tag=2CD733C8-216A
Date: Tue, 14 Aug 2007 09:16:49 GMT
Call-ID: 24524ebd58e984e930f461ff5d490bd9@asterisk-sip-gateway
CSeq: 104 NOTIFY
Content-Length: 0


Aug 14 09:16:49.843: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:0116@cme-router:5060 SIP/2.0
Via: SIP/2.0/UDP asterisk-sip-gateway:5060;branch=z9hG4bK136ae0fa;rport
From: "Tom" ;tag=as221e2abe
To: ;tag=2CD733C8-216A
Call-ID: 24524ebd58e984e930f461ff5d490bd9@asterisk-sip-gateway
CSeq: 105 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


My voice service voip config is as below:

voice service voip
allow-connections sip to sip
no supplementary-service sip moved-temporarily
sip
registrar server expires max 240 min 60
no call service stop


And the Grandstream phone config is:

voice register dn 1
number 2007
allow watch
refer target dial-peer
mwi
!
voice register pool 1
id mac 000B.820D.0536
number 1 dn 1
template 1
dtmf-relay rtp-nte
voice-class codec 1
description Grandstream


Where ‘cme-router’ is the router on which CME 4.2 is installed and running, and ‘asterisk-sip-gateway’ is our SIP gateway (which the CME 4.2 SIP-UA connects to) that handles more advanced call routing features.

If anyone has any ideas – I’d appreciate the help! It’s not a massive problem, but I’m keen to see this one through.

Don’t assume that just because there is no output from the test, that nothing has been matched. If there are no matches across any of your rules, you will see:

01234555321 Didn't match with any of rules


If there’s no output, then you have a partial match. If you issue ‘debug voice translation’, and watch carefully, you will see output such as this:

Router(cfg-translation-rule)#do test voice translation-rule 1 0123
Router(cfg-translation-rule)#
//-1/xxxxxxxxxxxx/RXRULE/regxrule_match: more digits needed; number=0123 rule precedence=1
//-1/xxxxxxxxxxxx/RXRULE/regxrule_match: more digits needed; number=0123 rule precedence=2
//-1/xxxxxxxxxxxx/RXRULE/regxrule_match: more digits needed; number=0123 rule precedence=3
//-1/xxxxxxxxxxxx/RXRULE/regxrule_match: more digits needed; number=0123 rule precedence=4
//-1/xxxxxxxxxxxx/RXRULE/regxrule_rule_match: partial match found, partial_match 4


There is a problem with this: if your translation rules are mistakenly waiting for any number (as mine was this morning; I’d left a .* on the beginning of the expression!), then your calling will exhibit some odd behaviour.

For instance, when dialling extensions from one ephone to another (simply across CME) your call will be subject to the ‘timeout interdigit’ value (as set in telephony-service). Why? Well, if your translation rules pick up a partial match they wait for a predefined length of time to see if you dial any other digits. The time waited is predefined by the interdigit timeout!

Even if the number matches the correct dial-peer (eg. 20001, 20002) and eventually dials correctly, you will still be subjected to the interdigit timeout, despite the lack of any ‘T’ in your destination-patterns. This is purely because the translation rules are waiting on your dial pattern.

Just thought I’d mention it. Really confused me this morning – many thanks to Tele for helping out!

So the hunt began, and seemingly not long before we’d started looking, Cisco had released IOS version 12.4(15)T, which indeed comes with CME 4.1 bundled in the appropriate voice-related featuresets. I chose this release, as the only alternative was an earlier X release. And I’m still quite un-sure of X releases, as it happens.

Now for the severe warning:

Do NOT attempt to register an Snom 360 SIP phone to the SIP registar with this release. We’ve tested a few non-Cisco SIP phones so far, all with varying degrees of success (the Grandstream GXP2000 worked well, though it’s far from usuable*), but the Snom actually caused the router to crash and reboot each time it attempted to register.

Strange but true – a SIP phone registering caused the router (a 2811) to bomb-out and restart at least six times before I realised what was going on. I guess there’s one reason why the more experienced Cisco geeks shy well away from T-train releases…

To make matters worse, I’ve found another strange, yet seemingly harmless bug with this new image. Whenever a call is made, and it doesn’t matter where it’s orginated or destined for; as soon as the call is answered the CLID is suffixed with seemingly random crap. Due to the fact that most of it is garbage, with the rest resembling bits of caller IDs from previous calls, I’d hazard a guess that this is some sort of memory over/underflow issue (but I’m no experienced programmer, so feel free to correct me.)

I’m also not sure whether it’s the IOS release, or CME 4.1 which is causing the problem. I’ve not had the 12.4(11)XJ3 release to test it with, and I’m more-likely to push fowards instead of backwards as there’s since been a newer IOS released with CME 4.2 and I’m hoping it will solve the problem. The only problem is, it’s another X release.

What I do know though, is that changing the 7960 firmware load makes no difference what-so-ever. I’ve tested 8.0(5.0) (which happens to be the recommended version for use with CME 4.1), 8.0(6.0) and even 8.0(4.0) with no change in behaviour. The only thing I can attest to is that the Grandstream SIP phone I’m using for testing purposes doesn’t exhibit the same problem. That’s one yay for SIP, at least.

If anyone out there is experiencing the same problem, then please let me know any possible solutions that I could test!

*The Grandstream can currently dial extensions registered to both CME, and our Asterisk gateway but as for calling externally – I merely receive a 403 error. Maybe I need to keep playing with it, but I don’t think Cisco’s implementation of SIP still particularly favourable…