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Inbound & Outbound VoIP Dial Peers

blcase
Level 1
Level 1

Is it possible to have a gateway accept inbound voip dial-peer and send that call out another voip dial-peer? How might that configuration look?

Thanks!

22 Replies 22

Dear Prasanna,

Are you sure that calls are not even coming to AS?

In fact, maybe calls are coming trough PRI and then tries to dial telco again but telco is giving busy signal to those calls...

example:

1. someone dials PRI number 2720055 and he comes to the AS

2. num-exp or translation rule is applied and now AS dials telco number 160221,,,,1212121212

3. Telco gives busy signal

why?

1. number is really busy

2. number is invalid

3. pause time is not long enough

4. there are some user authentification which needs to be proccesed during a call

...

maybe you can try to debug and see if calls are coming to the AS and are they are going out of AS?

Regards,

Goran

Try this ...

translation-rule 1

rule 0 2720055 160221,,,,12121212

dial-peer voice 2171 pots

incoming called number 217232....

dial-peer voice 2172 pots

destination-pattern 2720055

translate-outgoing called 1

port 1/0/4:d

After you put this in the configuration, make a call and do 'show call hist voice brief' and see what dial-peers the call is matching, as Goran suggested.

K

Tried but still no movement

following obs I got :

sh call hist vo brie

: hs. + + pid:

dur hh:mm:ss tx:/ rx:/ ()

IP : rtt:

delay://ms

MODEMPASS buf:/ loss /

last s dur:/s

FR [int dlci cid] vad: dtmf: seq:

sig: (payload size)

ATM [int vpi/vci cid] vad: dtmf: seq:

sig: (payload size)

Telephony : tx://ms noise:dBm acom:dBm

Proxy :

anf>

bw: / codec:

tx:

tes>

rx:

es>

Total call-legs: 0

I think there is nothing in the above log

following is the dial-peer history :

sh dial-peer voice

VoiceEncapPeer2171

information type = voice,

tag = 2171, destination-pattern = `',

answer-address = `', preference=0,

numbering Type = `unknown'

group = 2171, Admin state is up, Operation state is up,

incoming called-number = `217232....', connections/maximum = 0/unlimited

,

DTMF Relay = disabled,

huntstop = disabled,

in bound application associated: DEFAULT

out bound application associated:

permission :both

incoming COR list:maximum capability

outgoing COR list:minimum requirement

type = pots, prefix = `',

forward-digits default

session-target = `', voice-port = `',

direct-inward-dial = disabled,

digit_strip = enabled,

register E.164 number with GK = TRUE

Connect Time = 0, Charged Units = 0,

Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0

Accepted Calls = 0, Refused Calls = 0,

Last Disconnect Cause is "",

Last Disconnect Text is "",

Last Setup Time = 0.

VoiceEncapPeer2172

information type = voice,

tag = 2172, destination-pattern = `2720055',

answer-address = `', preference=0,

numbering Type = `unknown'

group = 2172, Admin state is up, Operation state is up,

incoming called-number = `', connections/maximum = 0/unlimited,

DTMF Relay = disabled,

huntstop = disabled,

in bound application associated: DEFAULT

out bound application associated:

permission :both

incoming COR list:maximum capability

outgoing COR list:minimum requirement

type = pots, prefix = `',

forward-digits default

session-target = `', voice-port = `1/0/4:D',

direct-inward-dial = disabled,

digit_strip = enabled,

register E.164 number with GK = TRUE

Connect Time = 0, Charged Units = 0,

Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0

Accepted Calls = 0, Refused Calls = 0,

Last Disconnect Cause is "",

Last Disconnect Text is "",

Last Setup Time = 0.

Regds,

Prasanna

1. Were u able to successfully route calls through this gateway ? If yes How ?

2. Did u do PSTN -> IP on this gateway ?

3. R u able to do PSTN -> PSTN using different ports on this gateway ?

I can't see call coming in. Try to do 'debug isdn q931' and see if the call is coming in.

K

1.This AS 5800 is spcifically for Internet Dial up.

Uptill noe not used for any voice application.

2.No PSTN->IP on this gw

3.On all ports I am able to give ring to any telco no. by reverse telneting a modem from MICA modem pool

I tried to debug isdn q931.Output is :

sh debugging

ISDN:

ISDN Q931 packets debugging is on

ISDN Q931 packets debug DSLs. (On/Off/No DSL:1/0/-)

DSL 0 --> 31

1 1 1 1 1 1 - - - - - - - - - - - - - - - - - - - - - - - - - -

DSL 32 --> 55

- - - - - - - - - - - - - - - - - - - - - - - -

Following is the ISDN port status :

sh voice port 1/0/2:d

ISDN 1/0/2:D - 1/0/2:D

Type of VoicePort is ISDN

Operation State is DORMANT

Administrative State is UP

No Interface Down Failure

Description is not set

Noise Regeneration is enabled

Non Linear Processing is enabled

Music On Hold Threshold is Set to -38 dBm

In Gain is Set to 0 dB

Out Attenuation is Set to 0 dB

Echo Cancellation is enabled

Echo Cancel Coverage is set to 8 ms

Playout-delay Mode is set to default

Playout-delay Nominal is set to 60 ms

Playout-delay Maximum is set to 200 ms

Connection Mode is normal

Connection Number is not set

Initial Time Out is set to 10 s

Interdigit Time Out is set to 10 s

Ringing Time Out is set to 180 s

Companding Type is A-law

Region Tone is set for US

Wait Release Time Out is 30 s

Station name None, Station number None

Voice card specific Info Follows:

DS0 channel specific status info:

IN OUT

PORT CH SIG-TYPE OPER STATUS STATUS TIP RING

For pstn->telco house->AS5800->pstn any voice card is required ?

Regds,

Prasanna

Kiran

Following is the debug output of isdn q931

107472: 9w5d: ISDN Se1/0/2:15: RX <- SETUP pd = 8 callref = 0x7097

107473: 9w5d: Sending Complete

107474: 9w5d: Bearer Capability i = 0x8090A3

107475: 9w5d: Channel ID i = 0xA98394

107476: 9w5d: Progress Ind i = 0x8283 - Origination address is non-ISDN

107477: 9w5d: Calling Party Number i = 0x2183, '2172721111', Plan:ISDN,

Type:National

107478: 9w5d: Called Party Number i = 0x81, '2055', Plan:ISDN, Type:Unkn

own

107479: 9w5d: ISDN Se1/0/2:15: TX -> RELEASE_COMP pd = 8 callref = 0xF097

107480: 9w5d: Cause i = 0x80AF - Quality of service unavailable

Regds,

Prasanna

Dan, you over-simplify things.

Every call going through a Cisco H.323 gateway is matched to both incoming and outgoing dial peers, even if the incoming dial peer is not explicitly defined. And Incoming-Called-Number is only 1 of several matching criteria that can be used.

A simply dial peer scheme can be built without this information. But a complex dial plan will send a person to the funny farm in short order unless these concepts are understood.

happykiran
Level 1
Level 1

Voip - Voip is possible in 3745 ip-ip gateway running 12.3 or later. In other gateways you can only do ip-pstn or pstn-ip.

If you are using either AS5300 or AS5400 the best way to do, as explained by someother responder is to send the incoming ip to pstn and loop it back to another port and send it out as IP.

Another way of doing may be to write a TCL script, though I have not tried it myself, I think it can work.

Kiran.