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.

[chrysostomos1980]

Hi

add incoming called-number. in this dial peer

dial-peer voice 2 pots

description outgoing calls

destination-pattern .T

incoming called-mumber .

port 0/3/1:15

Questions:

What is the extension number in sip server?(951016650)?

We need full running config

Need also debug isdn q931 to see how the incoming call coming from PSTN.

[juanluis]

Thanks

I have tried the incoming called-number but it doesnt work. I send you the debug isdn q931.

That debug shows that in the beginnig the calling number is the real I use to test, 628436338, but then it changes to the same called number, 951016650. I dont know why it happens and if that is the reason the incoming call doesnt work.

The sip server doesnt have a extension with the called number, it has my gw defined and an incoming rule to send the calls to 951016650 to a operator private extension.

[chrysostomos1980]

Hi

What is the version of the voice gateway ios

[juanluis]

c3825-spservicesk9-mz.124-24.T3.bin

[chrysostomos1980]

Hi

Can you remove codec preference 1 transparent and recheck?

Is really strange the debug for isdn

is any reason that you change the default port for sip?Sip server is configured also with this port? (6060)

Also why do you have a plar configured?

If you want all the incomimg numbers to translated to 951016650, then create a simple translation rule

[juanluis]

I have rechecked but no success.

Here is the new debug. It keeps showing the change of calling number to the same called number, why?

[juanluis]

Yes, there is a reason, the sip server have to listen in 6060 port because another instance of the same server is listen in 5060 port to other customer.

I have connection plar because I thought I need it, if it is not necessary please tell me another way to get the incoming calls work.

I have checked without the connection plar and this is the debug, it seems that the call doesnt progress. In calling phone now I can hear a continous tone, but with the connection plar command set I can hear nothing.

So, must I delete the connection plar config?

GW#

GW#

Oct  4 15:05:44.030: ISDN Se0/3/1:15 Q931: RX <- SETUP pd = 8  callref = 0x1A19

        Sending Complete

        Bearer Capability i = 0x8090A3

                Standard = CCITT

                Transfer Capability = Speech

                Transfer Mode = Circuit

                Transfer Rate = 64 kbit/s

        Channel ID i = 0xA18386

                Preferred, Channel 6

        Calling Party Number i = 0x2183, '628436338'

                Plan:ISDN, Type:National

        Called Party Number i = 0xA1, '951016650'

                Plan:ISDN, Type:National

Oct  4 15:05:44.034: %ISDN-6-CONNECT: Interface Serial0/3/1:5 is now connected to 628436338 N/A

Oct  4 15:05:44.038: ISDN Se0/3/1:15 Q931: TX -> CALL_PROC pd = 8  callref = 0x9A19

        Channel ID i = 0xA98386

                Exclusive, Channel 6

Oct  4 15:05:44.038: ISDN Se0/3/1:15 Q931: TX -> CONNECT pd = 8  callref = 0x9A19

        Connected Number i = 0x80, '951016650'

Oct  4 15:05:44.106: ISDN Se0/3/1:15 Q931: RX <- CONNECT_ACK pd = 8  callref = 0x1A19

GW_INGENIA#

GW_INGENIA#

GW_INGENIA#

Oct  4 15:06:04.386: ISDN Se0/3/1:15 Q931: RX <- DISCONNECT pd = 8  callref = 0x1A19

        Cause i = 0x829000000000 - Normal call clearing

        Progress Ind i = 0x8288 - In-band info or appropriate now available

Oct  4 15:06:04.386: %ISDN-6-DISCONNECT: Interface Serial0/3/1:5  disconnected from 628436338 , call lasted 20 seconds

Oct  4 15:06:04.386: ISDN Se0/3/1:15 Q931: TX -> RELEASE pd = 8  callref = 0x9A19

Oct  4 15:06:04.718: ISDN Se0/3/1:15 Q931: RX <- RELEASE_COMP pd = 8  callref = 0x1A19

GW#

[chrysostomos1980]

Hi

Yes i believe that you dont need the conenction plar.

can you try another one call with debug ccsip messages?

I need to have , incoming called-number . in pots dial peer

Remove the tranlsrent codec

Remove plar

The incoming call coming from PSTN is always 951016650?You dont need any translation?

Also the connection with the PSTN network is E1 PRI

So the correct

interface Serial0/3/1:15

no ip address

encapsulation hdlc

isdn switch-type primary-net5 (If is europe)

isdn incoming-voice voice

no cdp enable

!

!

voice-port 0/3/1:15

cptone (cyptone of your country)

bearer-cap Speech

!

[juanluis]

I have done all the config changes but it doesn work, I only can hear a continous tone.

That is to test the solution, when I get to work it is possible I have to do translation.

This is the debug:

GWA#

GW#

Oct  4 15:39:46.613: %ISDN-6-CONNECT: Interface Serial0/3/1:17 is now connected to 628436338 N/A

Oct  4 15:39:49.621: %LINEPROTO-5-UPDOWN: Line protocol on Interface Serial0/3/1:17, changed state to up

Oct  4 15:39:49.621: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_bind_if_comingup: The interface which is bound to SIP is comin up

Oct  4 15:39:58.389: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAddContextToTable: Added context(0x68B13C00) with key=[9961] to tabl

e

Oct  4 15:39:58.389: //20409/000000000000/SIP/State/sipSPIChangeState: 0x68B13C00 : State change from (STATE_NONE, SUBSTAT

E_NONE)  to (STATE_IDLE, SUBSTATE_NONE)

Oct  4 15:39:58.389: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for e

vent 37

Oct  4 15:39:58.389: //20409/000000000000/SIP/Info/sipSPIUaddccCallIdToTable: Adding call id 4FB9 to table

Oct  4 15:39:58.389: //20409/000000000000/SIP/Info/sipSPIUaddCcbToUACTable: ****Adding to UAC table.

Oct  4 15:39:58.389: //20409/000000000000/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x68B13C00 key=D2C7C864-D5F11

E2-9DCE983E-D9D8E858@192.168.4.2

Oct  4 15:39:58.389: //20409/000000000000/SIP/State/sipSPIChangeState: 0x68B13C00 : State change from (STATE_IDLE, SUBSTAT

E_NONE)  to (SIP_STATE_OPTIONS_WAIT, SUBSTATE_NONE)

Oct  4 15:39:58.389: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone GMT+1 to SIP default timezone = G

MT

Oct  4 15:39:58.389: //20409/000000000000/SIP/Error/sipSPIAddPrivacyHeader: Orig Container is NULL...should have value

Oct  4 15:39:58.389: //20409/000000000000/SIP/Error/sipSPIAddAssertedIDHeader: Orig Container is NULL...should have value

Oct  4 15:39:58.389: //20409/000000000000/SIP/Error/sipSPIAddPreferredIDHeader: Orig Container is NULL...should have value

Oct  4 15:39:58.389: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

OPTIONS sip:192.168.4.21:6060 SIP/2.0

Via: SIP/2.0/UDP 192.168.4.2:5060;branch=z9hG4bK2B1126B1

From: <192.168.4.2>;tag=28CB535C-F19

To: <192.168.4.21>

Date: Thu, 04 Oct 2012 13:39:58 GMT

Call-ID: D2C7C864-D5F11E2-9DCE983E-D9D8E858@192.168.4.2

User-Agent: Cisco-SIPGateway/IOS-12.x

Max-Forwards: 70

CSeq: 101 OPTIONS

Contact: <192.168.4.2:5060>

Content-Length: 0

Oct  4 15:39:58.493: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg enqueued for SPI with IP addr: [192.168.4.21

]:6060

Oct  4 15:39:58.493: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for e

vent 1

Oct  4 15:39:58.493: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog

Oct  4 15:39:58.493: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.4.2:5060;branch=z9hG4bK2B1126B1

Contact: <192.168.4.21:6060>

To: <192.168.4.21>;tag=f5355f15

From: <192.168.4.2>;tag=28CB535C-F19

Call-ID: D2C7C864-D5F11E2-9DCE983E-D9D8E858@192.168.4.2

CSeq: 101 OPTIONS

Accept: application/sdp

Accept-Language: en

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE

Supported: replaces

User-Agent: 3CXPhoneSystem 10.0.23053.0

Allow-Events: presence, message-summary, dialog, call-info, line-seize

Content-Length: 0

Oct  4 15:39:58.493: //20409/000000000000/SIP/Info/sipSPICheckResponse: non-INVITE response with no RSEQ - do not disable

IS_REL1XX

Oct  4 15:39:58.493: //20409/000000000000/SIP/Info/sipSPIUdeleteccCallIdFromTable: Removing call id 4FB9

Oct  4 15:39:58.493: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIDeleteContextFromTable: Context for key=[9961] removed.

Oct  4 15:39:58.493: //20409/000000000000/SIP/Info/sipSPIUdeleteCcbFromUACTable: ****Deleting from UAC table.

Oct  4 15:39:58.493: //20409/000000000000/SIP/Info/sipSPIUdeleteCcbFromTable: Deleting from table. ccb=0x68B13C00 key=D2C7

C864-D5F11E2-9DCE983E-D9D8E858@192.168.4.2

Oct  4 15:39:58.493: //20409/000000000000/SIP/Info/sipSPIFlushEventBufferQueue: There are 0 events on the internal queue t

hat are going to be free'd

Oct  4 15:39:58.493: //20409/000000000000/SIP/Info/ccsip_qos_cleanup: Entry

Oct  4 15:39:58.493: //20409/000000000000/SIP/Info/sipSPI_ipip_free_codec_profile: Codec Profiles Freed

Oct  4 15:39:58.493: //20409/000000000000/SIP/Info/sipSPIUfreeOneCCB: Freeing ccb 68B13C00

Oct  4 15:39:58.493: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContextFromTable: NO context for key[9961]

Oct  4 15:40:07.525: %ISDN-6-DISCONNECT: Interface Serial0/3/1:17  disconnected from 628436338 , call lasted 20 seconds

GW#

[chrysostomos1980]

Hi

It seems that the CUBE(GW) send the call succesfully to sip server.From there i dont know how is working

What is the ip:

192.168.4.2

Also just to understand:

you wrote:

The sip server doesnt have a extension with the called number, it has my  gw defined and an incoming rule to send the calls to 951016650 to a  operator private extension.

Operator private extension where is register.

The sip server what number expecting to recieve from gw??(951016650)??

Operator private extension does not have a number>??

Regards

[juanluis]

That is the ip address of the GW lan interface, and 192.168.4.21 is assigned to sip server. 

[chrysostomos1980]

Ok

From the traces i see that 192.168.4.2 has port 5060.From what you say you want 6060 and not 5060

Are you sure that the sip server in listening ONLY for 6060 port with this connection?

Can you answer the above and also the previous post>?

Can i see also one outgoing call from the operator private extension?

Debug isdn q931

ccsip messages

[juanluis]

The gw is using port 5060, but sip server is working on port 6060. I think that's not a problem, is fact outgoing calls work fine, and the gw seems to be registered on the sip server.

The operator private extension is registered on sip server, I dont know the number, is a private number. The gw is expecting to receive calls on the number 951016650, and has a incoming rule to transfer that calls to the private extension.

I send you the debug for a outgoing succesfully call.

Thats very strange, I am really desperate.

Regards,

[chrysostomos1980]

Hi

From the outgoing you can see that all the sip messages are there.Invite , ACK, etc

In incoming call i dont see full sip messages

Most propably the sip server is not configure correct for this connection

Second :Gw is sending 951016650 to sip server.I believe that the call stuck there and never trying to go to the endpoint

So i want from you to be sure that sip server expecting to recieve 951016650 and not something else..

From outgoing call the calling number is 90001.So why would not be also and the incoming number 90001

You have to be sure what number expecting to recieve the sip server