problem with sip trunk between gw and ToIP ISP in incoming calls

[juanluis]

Hi, I have this VoIP network: SIP Server -- GW -- PSTN, with E1 link to PSTN, the gw and SIP server and GW are in the same LAN.

I can see the sip-ua is registered:

GW#show sip-ua register status

Line                              peer        expires(sec)  registered

================================  ==========  ============  ==========

.*                                2           154           no

90001                             -1          94            yes

GW#show sip-ua connections udp detail

Remote-Agent:192.168.4.21, Connections-Count:1

  Remote-Port Conn-Id Conn-State  WriteQ-Size

  =========== ======= =========== ===========

         6060       2 Established           0

The outgoing calls are working fine with phones registered in sip server. But I dont get work the incoming calls from pstn. This is the involved config:

!

voice service pots

supported-language es

supplementary-service qsig call-forward

!

voice service voip

dtmf-interworking rtp-nte

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to sip

redirect ip2ip

signaling forward unconditional

fax protocol t38 nse force ls-redundancy 0 hs-redundancy 0 fallback none

sip

  bind control source-interface GigabitEthernet0/0

  bind media source-interface GigabitEthernet0/0

  registrar server expires max 3600 min 3600

  no update-callerid

  early-offer forced

  midcall-signaling passthru

!

voice class codec 1

codec preference 1 transparent

codec preference 2 g711alaw

codec preference 3 g711ulaw

!

!

controller E1 0/3/1

framing NO-CRC4

pri-group timeslots 1-31

!

!

interface Serial0/3/1:15

no ip address

encapsulation hdlc

isdn switch-type primary-qsig

isdn incoming-voice voice

no cdp enable

!

bearer-cap 3100Hz

!

voice-port 0/3/1:15

echo-cancel coverage 64

cptone ES

connection plar 951016650

!

!

dial-peer voice 2 pots

description outgoing calls

destination-pattern .T

port 0/3/1:15

!

dial-peer voice 1 voip

description incoming calls

destination-pattern 951016650

voice-class codec 1

session protocol sipv2

session target ipv4:192.168.4.21:6060

dtmf-relay sip-notify h245-alphanumeric

!

!

sip-ua

credentials username 90001 password ****** realm 3CXPhoneSystem

keepalive target ipv4:192.168.4.21:6060

authentication username 90001 password ******

retry invite 3

retry response 3

retry bye 3

retry cancel 3

timers expires 300000

registrar ipv4:192.168.4.21:6060 expires 3600

sip-server ipv4:192.168.4.21:6060

!

I attach debug ccsip file.

I call from 952029343 to 951016650

Please, anyone can help me?, I have tried really a lot of commands and config but I dont get work the incoming calls.

Thanks in advance.

[juanluis]

It seems that the private extension operator number is 90009

How can I test if calling directly with translation rule the call is working?

I have set destination-pattern 90001 on dial-peer voice 1 voip and translate called number on voice-port, but it doesn work.

Must I configure the rule at other level?

Regards,

[chrysostomos1980]

Can is see your translation rule/profile/voice-port ,etc?

[juanluis]

!

translation-rule 1

Rule 0 951016650 90009

!

!

voice-port 0/3/1:15

translate called 1

!

dial-peer voice 1 voip

destination-pattern 90009

voice-class codec 1

session protocol sipv2

session target ipv4:192.168.4.21:6060

dtmf-relay sip-notify h245-alphanumeric

!

[chrysostomos1980]

I Prefer to use this format of translation

You can use debug voice translation  to check whats going

voice translation-rule 20987

rule 1 /951016650/ /90001/

!

voice translation-profile PROFILE-INCOMING

translate called 20987

!

voice-port 0/3/1:15

translation-profile incoming PROFILE-INCOMING

////////////////////////////////////////////////////

OR apply the profile in dial-peer voice 2 pots

Use voice-port for general incoming calls OR dial peer pots 2 for specific calls

dial-peer voice 2 pots

description outgoing calls

incoming called-number .

destination-pattern .T

translation-profile incoming PROFILE-INCOMING

port 0/3/1:15

/////////////////////////////////////////////////////

its good also to use this dial peer with two more commands

!

dial-peer voice 1 voip

description incoming calls

destination-pattern 90001

incoming called-number.

voice-class codec 1

session protocol sipv2

session target ipv4:192.168.4.21:6060

dtmf-relay sip-notify h245-alphanumeric

no vad

[juanluis]

Today the system doesnt wat working. There is no way.

Here is the last debug, you can see the call is going to 90009, but it doesn work.

I have a phone software registered with the sip server and call directly to 90009 and the call is OK. But that I need, the incoming call from pstn doesn want to work today.

Tomorrow I´ll try to contact with sip server administrator.

Thanks. If you review this debug and see something can help me, please let me know.

Regards,

[chrysostomos1980]

OK

Now here is the problem

Received:

SIP/2.0 404 User unknown

Sip server dont know what is 90009

So check it again with teh sip server administrator and verify what sip server expecting to recieve

[juanluis]

OK, thanks

But can I ask you a last question?, why I need the connection plar on the voice-port setup?

I can see that if I dont config that command, the incoming call doesnt progress. Must I config connection plar or translation?

I call to a number that is on the destination-pattern of the dial-peer associated with sip server, but the system seems that dont "call" to that dial-peer, why?

Regards.

[chrysostomos1980]

Hi

I believe that you dont need the plar ,

You need translation ONLY IF the incoming number should to be translated to a different number

Verify with the sip server administrator what number is expecting to recieve from you

Also if you want to check if you hit the correct dial peerand if the translation working correct then  you can use

debug voice dialpeer

debug voice translation

Regards

cc

[juanluis]

This is the debug dialpeer output:

GW#

Oct  5 11:11:15.256: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

   Calling Number=952029343, Called Number=951016650, Voice-Interface=0x6918D8B8,

   Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search Type=PEER_TYPE_VOICE,

   Peer Info Type=DIALPEER_INFO_SPEECH

Oct  5 11:11:15.256: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

   Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=2

Oct  5 11:11:15.256: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

   Calling Number=952029343, Called Number=951016650, Voice-Interface=0x0,

   Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search Type=PEER_TYPE_VOICE,

   Peer Info Type=DIALPEER_INFO_FAX

Oct  5 11:11:15.256: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

   Result=NO_MATCH(-1) After All Match Rules Attempt

Oct  5 11:11:15.264: %ISDN-6-CONNECT: Interface Serial0/3/1:17 is now connected to 952029343 N/A

Oct  5 11:11:33.865: %ISDN-6-DISCONNECT: Interface Serial0/3/1:17  disconnected from 952029343 , call lasted 18 seconds

Oct  5 11:11:34.153: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

   Calling Number=.T, Called Number=, Voice-Interface=0x0,

   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,

   Peer Info Type=DIALPEER_INFO_SPEECH

Oct  5 11:11:34.157: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

   Result=NO_MATCH(-1) After All Match Rules Attempt

GW#

I have no connectio plar config, and it seems that the system doesnt find the voip dialpeer with destination pattern 951016650, why?

This is the dialpeer config:

!

dial-peer voice 2 pots

destination-pattern .T

incoming called-number .

port 0/3/1:15

!

dial-peer voice 1 voip

destination-pattern 951016650

voice-class codec 1

session protocol sipv2

session target ipv4:192.168.4.21:6060

dtmf-relay sip-notify h245-alphanumeric

!

Regards.

[chrysostomos1980]

Its supposed that you have remove the translation profile or translation rule from the voice-port right?

Also put the no vad into ALL the voip dial peer.VAD is bad

[juanluis]

Yes, no translation. And no vad on dialpeer voip.

I dont understand why the system needs connection plar to progress the call from pstn voice-port to voip sip dialpeer.

Really I must doing something wrong, but what?

[chrysostomos1980]

I am sure that is not nee it

The call coming from E1 and then it should be go to dial-peer voice 1 voip

If you are sure that you dont missed any misconfiguration , then pls reload the VG

[juanluis]

I can not reload the gw now, I have to wait this afternoon. I'll try it.

Thats the question, the call should go from E1 to voip dial-peer, but the call seems stop his travel.

In the phone I test the call, it seems to be established or connected immediately after dialing the number, it is as the system doesnt know there is a voip dialpeer with that destination number.

My gw has his own will.

Although all is different when I set up connection plar, as you can see in the last debug. Why?

Regards,