IPO to IPO over MPLS 5

[Markee2171]

Hi folks, apologies if this has been covered elsewhere I'm down to tearing my hair out looking for the correct combination of keywords.

Basically I have 2 sites, connected by MPLS-VPN with QoS and all IP routes are sorted. Both sites have IPO 406's ( albeit differing FW ) I'm trying to set up an IP Line between the 2 for the transfer of internal calls.

created new line, ( line 10, inc grp 10, out grp 10, IP of site B IPO )

shortcode of 2N, ., Dial, Group ID 10

watching callstatus - dialled number gets sent to ISDN ( line 5 ) not IP ( Line 10 )

Any suggestions

TIA

Mark
============================

Config

Site A - 192.168.3.0/255.255.255.0 - 406 FW 3.2(55)
Site B - 192.168.0.0/255.255.255.0 - 406 FW 21.(27)

 

[PlatinumGuy]

Correct.

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[Markee2171]

Hi Guys ( girls )

now have IP lines set up on BOTH IPO's -

calling from a to b - engaged tone
calling from b to a - "number busy " on screen

haven't bothered with the shortcodes as voice networking is ticked

any ideas

Cheers

Mark

 

[PlatinumGuy]

Run a sys mon and trace for AVRIP TX and RX in the routing tab, are these packets getting through?

If you dont have short codes any more then it is these packets that are updating the users lists on both systems. If this isnt getting through it wont know who is at the other end.

Also open up PM and try and "add user" to the speed dial tab, can you see the users on the remote system?

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[Markee2171]

RIGHTO - UPDATE

Onsite "manager" had decided to add users from siteB as users on SiteA's IPO !!

After I fell off my chair with laughter she told me that she'd only just done it ( I can't complain too much she owns the comapny and is usually pretty good )

Waiting for a reboot ( after removing all the users )

YES - Can see SiteB's users if I go to AddUSer in PM

Cheers - keep you informed

Mark

 

[Markee2171]

Here's more

got the Sysmon log as requested ( tbh dont know what i'm looking for )

AVRIP Tx: v=192.168.3.100 send=0 receive=0
PLATFORM: requiresvoicemail 0 operational 0 version 0 recordsupported 0
104021mS AVRIP Rx: v=192.168.3.100 send=1 receive=0
PLATFORM: requiresvoicemail 0 operational 0 version 0 recordsupported 0
113714mS AVRIP Tx: v=192.168.3.100 send=0 receive=0
PLATFORM: requiresvoicemail 0 operational 0 version 0 recordsupported 0
113776mS AVRIP Rx: v=192.168.3.100 send=1 receive=0
PLATFORM: requiresvoicemail 0 operational 0 version 0 recordsupported 0
119907mS PRN: National Number in prefix 0
119924mS PRN: TFTPServer::RRQ(from 192.168.0.82) nasystem/call_info/0.1104.0
123540mS AVRIP Rx: v=192.168.3.100 send=0 receive=0
PLATFORM: requiresvoicemail 0 operational 0 version 0 recordsupported 0
123712mS AVRIP Tx: v=192.168.3.100 send=0 receive=1
PLATFORM: requiresvoicemail 0 operational 0 version 0 recordsupported 0
126782mS PRN: TFTPServer::RRQ(from 192.168.0.82) nasystem/call_info/0.1104.0
129846mS PRN: TFTPServer::RRQ(from 192.168.0.63) nasystem/user_info/01244545907
131811mS PRN: TFTPServer::RRQ(from 192.168.0.63) nasystem/user_info/01244545907
133137mS PRN: TFTPServer::RRQ(from 192.168.0.63) nasystem/user_info/01244545907
133282mS AVRIP Rx: v=192.168.3.100 send=0 receive=0
PLATFORM: requiresvoicemail 0 operational 0 version 0 recordsupported 0
133585mS PRN: TFTPServer::RRQ(from 192.168.0.82) nasystem/call_info/0.1105.0
133718mS AVRIP Tx: v=192.168.3.100 send=0 receive=1
PLATFORM: requiresvoicemail 0 operational 0 version 0 recordsupported 0
134090mS PRN: TFTPServer::RRQ(from 192.168.0.63) nasystem/user_info/01244545907
136145mS PRN: TFTPServer::RRQ(from 192.168.0.63) nasystem/call_info/0.1108.0
137032mS PRN: TFTPServer::RRQ(from 192.168.0.82) nasystem/call_info/0.1105.0
141909mS PRN: TFTPServer::RRQ(from 192.168.0.63) nasystem/call_info/0.1108.0
143050mS AVRIP Rx: v=192.168.3.100 send=0 receive=0
PLATFORM: requiresvoicemail 0 operational 0 version 0 recordsupported 0
143630mS PRN: TFTPServer::RRQ(from 192.168.0.82) nasystem/call_info/0.1110.0
143717mS AVRIP Tx: v=192.168.3.100 send=0 receive=1
PLATFORM: requiresvoicemail 0 operational 0 version 0 recordsupported 0
150108mS PRN: TFTPServer::RRQ(from 192.168.0.82) nasystem/call_info/0.1114.0
152182mS PRN: TFTPServer::RRQ(from 192.168.0.63) nasystem/call_info/0.1108.0
152806mS AVRIP Rx: v=192.168.3.100 send=0 receive=0
PLATFORM: requiresvoicemail 0 operational 0 version 0 recordsupported 0
153712mS AVRIP Tx: v=192.168.3.100 send=0 receive=1
PLATFORM: requiresvoicemail 0 operational 0 version 0 recordsupported 0
157105mS PRN: TFTPServer::RRQ(from 192.168.0.82) nasystem/call_info/0.1114.0
159242mS PRN: TFTPServer::RRQ(from 192.168.0.82) nasystem/call_info/0.1114.0
162567mS AVRIP Rx: v=192.168.3.100 send=0 receive=0
PLATFORM: requiresvoicemail 0 operational 0 version 0 recordsupported 0
163712mS AVRIP Tx: v=192.168.3.100 send=0 receive=1
PLATFORM: requiresvoicemail 0 operational 0 version 0 recordsupported 0
172330mS AVRIP Rx: v=192.168.3.100 send=0 receive=0
PLATFORM: requiresvoicemail 0 operational 0 version 0 recordsupported 0
173716mS AVRIP Tx: v=192.168.3.100 send=0 receive=1
PLATFORM: requiresvoicemail 0 operational 0 version 0 recordsupported 0
182074mS AVRIP Rx: v=192.168.3.100 send=1 receive=1
PLATFORM: requiresvoicemail 0 operational 0 version 0 recordsupported 0
183718mS AVRIP Tx: v=192.168.3.100 send=1 receive=2
PLATFORM: requiresvoicemail 0 operational 0 version 0 recordsupported 0
191839mS AVRIP Rx: v=192.168.3.100 send=2 receive=2
PLATFORM: requiresvoicemail 0 operational 0 version 0 recordsupported 0
193263mS PRN: TFTPServer::RRQ(from 192.168.0.82) nasystem/call_info/0.1118.0
193721mS AVRIP Tx: v=192.168.3.100 send=0 receive=0
PLATFORM: requiresvoicemail 0 operational 0 version 0 recordsupported 0
201590mS AVRIP Rx: v=192.168.3.100 send=0 receive=0
PLATFORM: requiresvoicemail 0 operational 0 version 0 recordsupported 0


make any sense?? not enough ?

anything else I can try before I go bald heh heh, cheers again

MArk

 

[Markee2171]

Here's the hosted configs

Site A -

http://remsup.dnsalias.net/configs/thorncliffe.cfg

Site B -

http://remsup.dnsalias.net/configs/Fencing.cfg

Any more info please let me know

Mark

 

[PlatinumGuy]

Markee, are you saying that users from a remote site are appearing in the cfg at the local site etc. ie users are swapping across the SCN?

If yes, this is a known issue.

Did you use the wizard to create the cfgs? That causes this problem and can screw up SCN big style.

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[Markee2171]

Hi platinumguy

the users are appearing in the "adduser" bit of the "phone manager" as per your suggested test.

the admin at site a had added the users from site b in as if they were local users. I've resolved that.

Wizards to create a config, please i'm way too much of a techie to use such things as wizards ;o)

Cheers again

configs are hosted at

http://remsup.dnsalias.net/configs/

if that helps

Mark

 

[PlatinumGuy]

Ok, got ya.

If you can see the users in PM then SCN is working.

There has to be something else over riding that if you still cannot dial across.

You can see from the trace that the avrip packets are getting through, this is backed by the fact that you the visibility in PM. Have you tried the PM thing both ways?

Im just trying to get the cfgs now, can you run a trace with h323 events of a failed call and post it here telling us what number was dialled etc?

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[Markee2171]

Not tried PM on site B as there aren't any PC's. That'll have to wait until I'm on site next with my laptop ( maybe this afternoon )

I'm on site this AM, I'll try the trace when I get back to my desk

Virtual Beers all round for help so far ;o)


Mark

 

[PlatinumGuy]

I cant get those cfgs, the web browser seraches for ages then comes back page not found?

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[Markee2171]

aaaww cr@p

home internet gone down for the first time in about 2 years

I'll swing past on my way back to the office and give it a kick.

Cheers

 

[PlatinumGuy]

Ill try again throughout the day.

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[Markee2171]

It's back up now - apologies for the inconvenience

Cheers

 

[PlatinumGuy]

I cant see anything major in the cfgs that would stop this working.

You can see the users in PM so the scn is working.

I know this is MPLS but is there a firewall in the mix here?

Can you run a trace of a call that fails.

I cant see anything wrong here.

Your software levels are not supported for scn, only one major revision apart is supported, anything further is not. However i know of lots of sites that work between 2.1 and 3.2.

I think we need to see a trace of a failed call please. Can you enable the default options plus h323. And everything in the scn tab.

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[Markee2171]

Cheers

I'm on site all day next tuesday - I'll upgrade sieta to 3.2 ( they have an alchemy phone30 is that supoprted still )

I'll be better placed to run any checks and tests and stuff from there

Thanks again

Mark

 

[Markee2171]

Mornin

I'm on site, please advise what it was you wanted to be monitored again

Cheers

Mark

 

[Markee2171]

Hi

just to say thanks for all your help and support whilst I went bald over the past few days.

I have convinced the client to purchase a new 406v2 for Site A along with a proper IP400 Phone 30

If I was to have upgraded the firmware on ther original box, I would have lost all the DT phones and quite possibly the Network Alchemy phone 30 too.

The new kit should arrive today - I'll let you know how it goes

Thanks again, you've all been fantastic

Mark

 

[Markee2171]

Thanks for bearing with me, just finished replacing the 403/argentphone with a 406v2 andIP400Phone30

after a hair raising panic over the analog phones not ringing ( firmware ) all now seems well.

EXCEPT

the SCN Site to site ( original issue ) now if i choose extn 401 call status shows Dial and silence for minutes at a time, but if I call other extn's then come up as Busy - Disconected

all sites are 406v2 3.2(55)

any help gratefully received

Mark

 

[raknas]

Have you set compression mode to G711 64k at both sites?