SIP-to-TDM...

Answered Question
Feb 24th, 2009

I am trying to do SIP-to-TDM with an AS5300 running 12.3(26) (that's the latest the AS5300 can run). I have a SIP trunk pointed at the FE port of the AS5300. I have a dial-peer configured like this:

dial-peer voice 100 pots

trunkgroup CME

destination-pattern 8005551122

I have one of my T1 controllers like this:

controller T1 0

framing esf

linecode b8zs

pri-group timeslots 1-24

interface Serial0:23

no ip address

encapsulation hdlc

trunk-group CME

isdn switch-type primary-5ess

no cdp enable

On the other end of the T1 is a 1760 router with an 1MFT-1T VWIC installed. That side is configured like:

controller T1 1/0

framing esf

linecode b8zs

pri-group timeslots 1,24

interface Serial1/0:23

no ip address

encapsulation hdlc

no logging event link-status

isdn switch-type primary-5ess

isdn incoming-voice voice

no cdp enable

When I do a 'show ip interface brief' on the 1760 I get Serial1/0:0 in "Reset/Down" and Serial1/0:23 in "Up/Up".

When I run a 'debug voip dialpeer all' on the 1760 I never see anything come up. Can someone lead me in the right direction to go from SIP to TDM? TIA.

I have this problem too.
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Correct Answer by Nicholas Matthews about 7 years 9 months ago

I agree - unless you have some restrictions on your CME setup to prevent you from SIP trunking (can't think of any good ones), CME is going to be better suited to terminate the SIP trunk. If you have a large amount of other pots interfaces you're wanting to use with the SIP trunk it could make sense.

'debug voip ccapi inout' is my favorite for troubleshooting call routing. 'debug voip dialpeer' is the stripped down version that may be all you need.

You'll need these dial peers for your current setup:

CME:

dial-peer voice 1 pots

destination-pattern 9.T

port 0/0:23

incoming called-number .

direct-inward-dial

5300:

dial-peer voice 1 pots

destination-pattern 444555....

port 0/0:D

incoming called-number .

direct-inward-dial

dial-peer voice 1 voip

incoming called-number .

session protocol sip

session target ipv4:x.x.x.x

destination-pattern .T

codec g711ulaw

no vad

dtmf-relay rtp-nte

In this case, your 10 digit numbers are 4445550000 - 4445559999 and you're using 4 digit extensions. Any digits explicitly matched in a pots dial peer will be automatically stripped. .... sends 4 digits, 34.. sends 2 digits.

hth,

nick

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Paolo Bevilacqua Tue, 02/24/2009 - 14:24

Make one side clock source internal, and protocol-emulate network. You should be good to go after. Watch the calls with debug isdn q931.

snickered Tue, 02/24/2009 - 15:14

Thanks a lot. These two commands made the 1760 receive data from the AS5300. Now if I could just get it to make a phone ring. I'm going to play with it a lot tonight and post more later.

Nicholas Matthews Tue, 02/24/2009 - 14:35

Where is your voip dial peer?

You should be able to pretty quickly determine if the 5300 will support SIP if the voip dial peer command "session protocol sip" exists.

Interface S0:23 should have isdn incoming-voice voice (or modem).

Agree with Paolo, you need one side to emulate the network side with isdn protocol-emulate network.

hth,

nick

snickered Tue, 02/24/2009 - 15:11

I'm not sure what to put in the voip dial-peer. I do have the 'session protocol sipv2' though. 'debug dialpeer' shows it matching my pots dial-peer and then I see a bunch of traffic on the 1760. Should the voip dial-peer have a session target of my carrier or what? Like this:

dial-peer voice 101 voip

destination-pattern 8005551122

session target ipv4:ip.add.of.isp

then I would need to check the incoming called-number in the pots dial-peer, right? Like this:

dial-peer voice 100 pots

incoming called-number 8005551122

trunkgroup CME

Thanks for your help.

Paolo Bevilacqua Tue, 02/24/2009 - 15:19

Ok, at this point I've lost track of how things are hooked up and what you're trying to do. Seems to me you're throwing commands almost random, eg trunkgroup what's for ?

snickered Tue, 02/24/2009 - 15:50

HAH! I don't doubt it.

Here's what I'm doing. I'm taking a SIP trunk from my carrier and attempting to convert it to TDM with an AS5300. Then I'd like to send it to a 1760 with CME installed where it will distribute the calls to my VoIP phones.

The reason I used trunkgroup was to route the call out any interface that is in that trunkgroup. In my case only one interface is in that group.

controller T1 0

framing esf

linecode b8zs

pri-group timeslots 1-24

interface Serial0:23

no ip address

encapsulation hdlc

trunk-group CME

isdn switch-type primary-5ess

no cdp enable

Am I not using that command correctly?

Paolo Bevilacqua Tue, 02/24/2009 - 16:19

Well first of all the 1760/CME should be able to sip trunk directly to the ITSP without the 5300 in between.

Then as you noticed, usefulness of trunkgroup is negated by having one only interface in it.

If you really want to have the 5300 blowing in your closet, as mentioned above, need to configure clock internal and network side. Observe show isdn status, if not multiframe established, something not right yet.

snickered Tue, 02/24/2009 - 16:52

I must have grabbed an old config. This is actually what I have:

controller T1 0

framing esf

linecode b8zs

pri-group timeslots 1-24

interface Serial0:23

no ip address

encapsulation hdlc

isdn switch-type primary-5ess

isdn protocol-emulate network

no cdp enable

I also see "State = MULTIPLE_FRAME_ESTABLISHED" in 'show isdn status'.

I don't really want this blowing in my closet. I will need to convert to TDM for another computer that only takes TDM in the near future. I just figured if I could get the 1760/CME router to do things with it I could get the computer to do things with it. I'm going to work on this some more to see if I can get the dial-peers to play nice and make a phone ring. I'll post more tomorrow or tonight. Thanks for your help.

Correct Answer
Nicholas Matthews Tue, 02/24/2009 - 16:31

I agree - unless you have some restrictions on your CME setup to prevent you from SIP trunking (can't think of any good ones), CME is going to be better suited to terminate the SIP trunk. If you have a large amount of other pots interfaces you're wanting to use with the SIP trunk it could make sense.

'debug voip ccapi inout' is my favorite for troubleshooting call routing. 'debug voip dialpeer' is the stripped down version that may be all you need.

You'll need these dial peers for your current setup:

CME:

dial-peer voice 1 pots

destination-pattern 9.T

port 0/0:23

incoming called-number .

direct-inward-dial

5300:

dial-peer voice 1 pots

destination-pattern 444555....

port 0/0:D

incoming called-number .

direct-inward-dial

dial-peer voice 1 voip

incoming called-number .

session protocol sip

session target ipv4:x.x.x.x

destination-pattern .T

codec g711ulaw

no vad

dtmf-relay rtp-nte

In this case, your 10 digit numbers are 4445550000 - 4445559999 and you're using 4 digit extensions. Any digits explicitly matched in a pots dial peer will be automatically stripped. .... sends 4 digits, 34.. sends 2 digits.

hth,

nick

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