Alcatel 4200 E and bri s/t te compatibilty issue

[l.kasneci]

Hi there, i need help!!!

I have one-way voice problem between two cisco 3600 series equiped with NM2V and VIC-2BRI-S/T-TE, in both sides of voip connectivity and 2 alcatel pabx's; the first pabx is omniPCX 4400 and dhe other is 4200 E. On the 4200 Alcatel side, i'm getting this output from "debug isdn q931":

ETSI supplementary service not supported;

Mandatory Information Element missing;

the call is successfully from omnipcx 4400 side to 4200 E side but is only one-way voice. In the opposite side, i'm getting only a short ring and nothing else.

I really need help

chears guys!

[layer3]

Can you post all the debug isdn q931 for a call?

Thanks.

Filipe

[l.kasneci]

Hi Filipe,

Thanks for your response! Please have a look to attachments.

[layer3]

Hi.

Please make a copy-paste of the entire debug trace for the call from the first setup message. (For example, you missed the compand type, since it is before the printscreen)

Select the text instead of print-screens :)

Another thing, the first image is from on call in the opposite direction of the other two?

Regarding the unsupported supplementary service, serivce 30 is Calling Line Identification Presentation

Cheers,

Filipe

[l.kasneci]

Ok Filipe,

i just find out how to make a copy-paste :))

see attachment please

Leonard

[layer3]

Hi.

Am I correct in assuming that the router is emulating user side?

If so, reconfigure the router to emulate network side (and the necessary changes on the pbx).

There is an issue regarding the caller id presentation on the setup message (which seems to be your problem) that is supposed to go away like that.

Cheers,

Filipe

[l.kasneci]

no, cisco is configured as network in both sides;

isdn bri config is:

interface bri1/0

no ip address

no ip route-cache

no ip mroute-cache

isdn switch-type basic-net3

isdn overlap-receiving

isdn protocol-emulate network

isdn layer1-emulate network

isdn incominng-voice voice

isdn T200 1500

no isdn outgoing display-ie

isdn static-tei 0

isdn skipsend-idverify

the same is for next bri1/1

voice port configuration:

voice-port 1/0/0

compand-type a-law

cptone DE

timeouts initial 60

voice-port 1/0/1

compand-type a-law

cptone DE

timeouts initial 60

in alcatel 4400 site:

dial-peer voice 100 pots

destination-pattern 1...

port 1/0/0

dial-peer voice 399 voip

destination-pattern 4...

session target ipv4:x.x.x.x

in alcatel 4200 site:

dial-peer voice 400 pots

destination-pattern 4...

port 1/0/0

dial-peer voice 399 pots

destination-pattern 4...

dial-peer voice 100 voip

destination-pattern 1...

session target ipv4:x.x.x.x

should be any problem IOS with encription?

ios i'm using in both sides is:

c3660-jk9o3s-mz-123-5b.bin

cheers

[layer3]

Hum...

Do you perform two-stage dialing?

I don't see direct-inward-dialling on the pots peers.

You are also missing the port command on the 399 pots peer and the session target on the 399 voip peer. I don't think this is related but you never know.

You might also try to remove the no isdn outgoing display-ie to see if it changes anything (i.e., something else sops working).

Cheers,

Filipe

[layer3]

Maybe I should explain the two-stage dialling question better:

From what I read of your config, a call coming out from a 4200 extension to a 4400 extension should happen this way:

1 - 4200 user picks up phone and gets 4200 dialtone;

2 - 4200 user dials a prefix and seizes trunk;

3 - 4200 user dials 1 plus 3 digits

4 - router connected to 4200 connects to router connected to 4400

5 - 4200 user hears another dialtone

6 - 4200 user dials 1 plus 3 digits

7 - router connected to 4400 sends the last 3 digits to 4400 and a 4400 extension rings

a 4400 to 4200 call should be the same with models reversed :)

Point 2 may be replaced by abbreviated dialing, i.e., user dials "1" and it is expanded to the trunk code plus another "1"

If the 4200 caller is calling from a digital phone he may not hear the dialtone at point 5. In fact he might not hear any progress tones at all (dialtones, ringing, etc.)

Is this what is happening or did you "simplify" the config you posted?

Cheers,

Filipe

[l.kasneci]

voice port is present in original config (i missed it to write it) should i use direct-inward-dial??

By your explanation, 1,2,3 are ok, and for my opinion follows:

4- router connected to 4200 connects to router connected to 4400 and forwords 4 collected digits, to the router in 4400 site;let's say 1... (1149) where last 3 digits is my post telephone number

5- is the cisco on the 4400 site that sends 3 last digits to my telephone (149) and i see in the display of my digital phone, calling number with 3 digits coming from 4200 site.

6-i haven't enough time to pick up the phone, cause the call releases imediately (within 2 seks).

that's it.

should be any one-way voice issue? cause the call goes from 4400 site to 4200 site.

cheer leonard

[l.kasneci]

i resolved the problem :) but i still not have dial tone and ringback on 4200 site. i've added direct-inward-dial, in 4200 site. But when i add direct-inward-dial in 4400 site, the situation is completly like before; i've used also no digit-strip command and i've changed numberin from 4 digits to 3 digits.

any idea how to resolve no dial tone and no ring in 4200 site??

thanks Leonard

[layer3]

Glad to have been of, at least, some help! :)))

Without direct-inward-dial the receiving router will present a dialtone to the caller and collect more digits. It will not use the received digits from the incoming call to complete the connection.

I guess the problems was that the 4200 was allergic to 2 stage dialing.

To solve the tone problems see below:

add to global config

voice call send-alert

add to the pots peers:

progress_ind setup enable 1

progress_ind alert enable 8

progress_ind progress enable 1

progress_ind connect enable 1

progress_ind disconnect enable 8

add to the voip peers:

progress_ind setup enable 3

Cheers,

Filipe

[l.kasneci]

I puted evrth on both routers; but still the same situation; i've restarted bri's but the same. the 4200 site doesn't hear dial-tone and ringback when they call 4400 site. The DID is puted only in 4200 site and i've used "no digit-strip" command. If i remove DID, the 4200 site can call succesfully only from analog phone; if they call from digital phone, in 4400 site is only a short ring and after that, call releases.

Cheers

Leonard

[layer3]

This is the relevant part of the config of one of the our alcatel-alcatel implementations:

pbx1

----

dial-peer voice 5000 voip

destination-pattern 5...

progress_ind setup enable 3

session target ipv4:192.168.5.80

dtmf-relay cisco-rtp

codec g711ulaw

fax-relay ecm disable

fax rate 9600

fax nsf 000000

ip qos dscp cs5 media

no vad

dial-peer voice 5200 pots

destination-pattern 52..

progress_ind setup enable 1

progress_ind alert enable 8

progress_ind progress enable 1

progress_ind connect enable 1

progress_ind disconnect enable 8

direct-inward-dial

port 1:15

prefix 52

pbx2

----

dial-peer voice 5000 pots

destination-pattern 5...

progress_ind setup enable 1

progress_ind alert enable 8

progress_ind progress enable 1

progress_ind connect enable 1

progress_ind disconnect enable 8

direct-inward-dial

port 1:15

prefix 5

dial-peer voice 5200 voip

destination-pattern 52..

progress_ind setup enable 3

session target ipv4:192.168.5.253

dtmf-relay cisco-rtp

codec g711ulaw

fax nsf 000000

ip qos dscp cs5 media

no vad

---

Background

-----------

Extensions 5200 in pbx1

Extensions 5000-5199 and 5300-5999 in pbx2

Call flow:

pbx2 -> pbx1

1 - User picks-up phone in pbx2, gets pbx2 dial-tone;

2 - User dials 5201

3 - Router2 connected to pbx2 collects digits and connects to router1

4 - Router1 connects to pbx1 and completes the call

5 - extension 5201 rings in pbx1

All tones are heard, regardless of analog or digital phone.

pbx2 is configured to seize trunk when users dials 52 and repeat those digits, so that the router gets all 4 digits (where it not be, the router would only get the final 2 digits)

This way there are no secondary dial-tones and multi-stage dialing to worry about. I have many pbxs interconnected with this or variations of this config and is very clean. If you ever use PRIs you can even use ABC-F in the interconnection and keep all the alcatel propriatery supplementary services.

Cheers,

Filipe