2600 and Siemens PBX

[Carlo Poggiarelli]

Hi all,

I've configured voip on two cisco 2610 with an E1 controller and I have two Siemens Hicom as PBX.

When I try to dial a remote number I hear for a mute tone and after about 5 seconds I have a busy tone.

Debugging the Isdn events I receive a Disconnect Cause "66 " and "recovery on timer expiry".

I tried to change ISDN timer value, but the behavior is the same.

This is how I've configured one of my two routers (the second is similar):

isdn switch-type primary-net5

call rsvp-sync

!

controller E1 1/0

clock source internal

pri-group timeslots 1-31

interface Serial1/0:15

no ip address

no logging event link-status

isdn switch-type primary-net5

isdn incoming-voice voice

isdn T309-enable

isdn T310 60000

no cdp enable

voice-port 1/0:15

cptone IT

!

dial-peer cor custom

!

dial-peer voice 2 voip

destination-pattern 5..

session target ipv4:10.0.0.2

!

dial-peer voice 1 pots

destination-pattern 2..

direct-inward-dial

port 1/0:15

prefix 2

Any idea?

Thank you in advance.

Carlo Poggiarelli

[nmilojevic]

Hi!

Not sure, but check following:

controller E1 1/0

clock source line primary

pri-group timeslots 1-31

voice-port 1/0:15

bearer-cap Speech

timeouts interdigit 5

cptone IT

Your dial-peers look a bit strange to me. They both have dest-pattern, so where do you have incoming peer? And on pots peer, you have DID configured (which indicates incoming peer) together with prefix (indicating outgoing peer).

On pots peer, port command should be probably "port 1/0:D", and not 1/0:15. At least it is so with AS5300.

Furthermore, do you have configured network / user sides?

Hope some of this helps,

Nikola

[Carlo Poggiarelli]

Hi!

I've found this kind of configuration on Cisco web site, in a document that

explains the interoperability between cisco 2600 and Siemens PBX.

Command "port 1/0:D" is not supported by 2600 and "line primary" I can't

find either.

My incoming peer are specified in dial-peer voice1 pots.

Why my dial-peers look strange?

Let me know.

Thank you

[anderson-david]

we have a 2621 connected to a siemens realitis this is some of the config

voice class codec 1

codec preference 1 g711alaw

codec preference 2 g711ulaw

interface Serial1/0:15

no ip address

no logging event link-status

isdn switch-type primary-qsig

isdn protocol-emulate network

isdn incoming-voice voice

no isdn T309-enable

fair-queue 64 256 0

no cdp enable

dial-peer voice 202 voip

destination-pattern 20..

progress_ind setup enable 3

voice-class codec 1

session target ipv4:x.x.x.x

dtmf-relay h245-alphanumeric

dial-peer voice 2 pots

destination-pattern 9T

progress_ind alert enable 8

progress_ind progress enable 8

progress_ind connect enable 8

direct-inward-dial

port 1/0:15

forward-digits all

dial-peer voice 3 pots

destination-pattern 5...

direct-inward-dial

port 1/0:15

forward-digits all

we use a DPNSS/QSIG convertor

this works fine, let me know how you get on,

hope this helps

dave

[Carlo Poggiarelli]

Hi Dave.

I tried to configure in this way my router. But when I set ISDN protocol

emulate as network the layer 2 goes down giving me a TEI ASSIGNED.

If I set it as USER layer 2 goes up.

Thnk u

Carlo

[nmilojevic]

Yes, I was looking some more... obviously "port 1/0:15" is the right one. As I said, I use 5300 not 2600, so I wasn't sure.

About dial-peers;

Personally, I like to make strict difference between incoming and outgoing dial-peers.

Direct-inward-dial is inbound command, and on the same dial-peer you have also dest-pattern, which behaves differently whan used as inbound than as when outbound.

For example; if dial-peer 1 pots is your incoming peer, dest-pattern is matched against calling number, not called! I am not sure, wether it strips matched digits, but for sure there is no use of adding that digit with prefix command, since it is calling party number and not called.

Have a look at the following document:

http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml

Now, I am not saying your dial-peers are responsible for not working, it's just I don't find them very clear.

Have you traced the call, to find out, how far does it get? That would be usefull.

[Carlo Poggiarelli]

Hi Nikola,

I've traced a call from the remote site and this is the result:

GENERIC:

SetupTime=11912521 ms

Index=25

PeerAddress=505

PeerSubAddress=

PeerId=2002

PeerIfIndex=73

LogicalIfIndex=0

DisconnectCause=66

DisconnectText=recovery on timer expiry

ConnectTime=0

DisconnectTime=11913332

CallDuration=00:00:00

CallOrigin=1

ChargedUnits=0

InfoType=speech

TransmitPackets=0

TransmitBytes=0

ReceivePackets=0

ReceiveBytes=0

VOIP: onnectionId[0xE3F349D0 0x160311CC 0x8034A670 0xD92CE

D33]

IncomingConnectionId[0xE3F349D0 0x160311CC 0x8034A670 0xD92CED33]

RemoteIPAddress=10.0.0.2

RemoteUDPPort=17602

RoundTripDelay=0 ms

SelectedQoS=best-effort

tx_DtmfRelay=h245-alphanumeric

FastConnect=TRUE

Separate H245 Connection=FALSE

H245 Tunneling=TRUE

SessionProtocol=cisco

SessionTarget=ipv4:10.0.0.2

OnTimeRvPlayout=0

GapFillWithSilence=0 ms

GapFillWithPrediction=0 ms

GapFillWithInterpolation=0 ms

GapFillWithRedundancy=0 ms

HiWaterPlayoutDelay=0 ms

LoWaterPlayoutDelay=0 ms

ReceiveDelay=0 ms

I don't know if it can help you but when I try to do a call from a phone, the display show me the "Inhibited access" message and then I hear for a busy tone.

Hope this help you

Thank You a lot again

Carlo

[nmilojevic]

To tell you the truth, it doesn't help me much. What I meant is - check call step by step... do you receive it on expected incoming pots dial-peer of first router, is it matched to correct outgoing voip dial-peer on first router, and so on on second router. When you find out, where exactly call stops, enable debugging there to find out, why does it stop.

Let me know, when you fing out anything.

Nikola

[Carlo Poggiarelli]

Hi Nikola.

Do you see something wrong on the following text?

1/0:15 31 State Transitions: (state, event) -> (state, event) ...

(S_NULL, E_TSP_PROCEEDING) -> (S_SETUP_REQ_PROC, E_TSP_ALERT) ->

(S_SETUP_REQ_PROC, E_CC_BRIDGE) -> (S_SETUP_REQ_PROC, E_CC_CAPS_IND) ->

(S_SETUP_REQ_PROC, E_CC_CAPS_ACK) -> (S_SETUP_REQ_PROC, E_CC_REQ_PACK_STAT) ->

(S_SETUP_REQ_PROC, E_DSP_GET_TX) -> (S_SETUP_REQ_PROC, E_DSP_GET_RX) ->

(S_SETUP_REQ_PROC, E_DSP_GET_VP_DELAY) -> (S_SETUP_REQ_PROC, E_DSP_GET_VP_ERROR)

->

(S_SETUP_REQ_PROC, E_TSP_CONNECT) -> (S_CONNECT, E_CC_REQ_PACK_STAT) ->

(S_CONNECT, E_DSP_GET_TX) -> (S_CONNECT, E_DSP_GET_RX) ->

(S_CONNECT, E_DSP_GET_VP_DELAY) -> (S_CONNECT, E_DSP_GET_VP_ERROR) ->

Thank u a lot again

Carlo