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New Member

Outgoing Calls SIP Provider (CME)

Hi,

i have a problem with outgoing calls on my CME router. I don`t know why. :-/ Incoming works well. I think i missed some little think but i don`t know what.

Here is the config and debug.

**CONFIG**

voice call send-alert

voice call convert-discpi-to-prog

voice call carrier capacity active

voice rtp send-recv

!

voice service voip

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

redirect ip2ip

fax protocol pass-through g711alaw

h323

modem passthrough nse codec g711alaw

sip

  bind control source-interface FastEthernet0/0

  session transport tcp

  registrar server expires max 3600 min 3600

  redirect contact order best-match

  localhost dns:sipgate.de

!

voice class codec 1

codec preference 1 g711alaw

!

!

!

!

voice translation-rule 3

rule 1 /^0/ /+49/

!

voice translation-rule 4

rule 1 /^1001/ /004930129854765/

!

voice translation-rule 11

rule 1 /^/ /0/ type unknown unknown

rule 2 /^/ /00/ type national national

rule 3 /^/ /000/ type international international

!

voice translation-rule 12

rule 1 /^004930129854765/ /1001/

!

!

voice translation-profile SIP-Incoming

translate calling 11

translate called 12

!

voice translation-profile SIP-Outgoing

translate calling 4

translate called 3

!

!

!

dial-peer voice 11 voip

description **Incoming Call**

translation-profile incoming SIP-Incoming

session protocol sipv2

session target dns:sipgate.de

incoming called-number .%

voice-class codec 1

voice-class sip dtmf-relay force rtp-nte

dtmf-relay rtp-nte

no vad

!

dial-peer voice 10 voip

description **Outgoing Call**

translation-profile outgoing SIP-Outgoing

destination-pattern 0T

session protocol sipv2

session target dns:sipgate.de

voice-class codec 1

voice-class sip dtmf-relay force rtp-nte

dtmf-relay rtp-nte

no vad

!

!

sip-ua

credentials username 004930129854765 password 7 <password> realm sipgate.de

authentication username 004930129854765@sipgate.de password 7 <password>

no remote-party-id

retry invite 2

retry register 10

retry options 1

timers connect 100

registrar dns:sipgate.de expires 3600

sip-server dns:sipgate.de

host-registrar

!

!

!

telephony-service

no auto-reg-ephone

max-ephones 20

max-dn 40

ip source-address 192.168.5.99 port 2000

load 7960-7940 P00308010200

max-conferences 8 gain -6

web admin system name cisco secret 5 <password>

dn-webedit

time-webedit

transfer-system full-consult

create cnf-files version-stamp Jan 01 2002 00:00:00

!

!

ephone-dn  1  dual-line

number 1001 secondary 004930129854765 no-reg primary

!

!

**DEBUG INCOMING**

*Nov  2 23:05:09.759: //8/xxxxxxxxxxxx/CCAPI/cc_api_caps_ind:

   Call Entry Is Not Found

*Nov  2 23:05:09.763: //-1/F2512023800F/CCAPI/cc_api_display_ie_subfields:

   cc_api_call_setup_ind_common:

   cisco-username=+4917661022799

   ----- ccCallInfo IE subfields -----

   cisco-ani=+4917975322866

   cisco-anitype=0

   cisco-aniplan=0

   cisco-anipi=0

   cisco-anisi=1

   dest=004930129854765

   cisco-desttype=0

   cisco-destplan=0

   cisco-rdie=FFFFFFFF

   cisco-rdn=

   cisco-lastrdn=

   cisco-rdntype=0

   cisco-rdnplan=0

   cisco-rdnpi=-1

   cisco-rdnsi=-1

   cisco-redirectreason=-1   fwd_final_type =0

   final_redirectNumber =

   hunt_group_timeout =0

*Nov  2 23:05:09.763: //-1/F2512023800F/CCAPI/cc_api_call_setup_ind_common:

   Interface=0x49E2F7A4, Call Info(

   Calling Number=+4917975322866,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),

   Called Number=004930129854765(TON=Unknown, NPI=Unknown),

   Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,

  Incoming Dial-peer=11, Progress Indication=NULL(0), Calling IE Present=TRUE,

   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=8

   Guid=F2512023-04DD-11E1-800F-807480744F27, Outgoing Dial-peer=20001

*Nov  2 23:05:09.775: //8/F2512023800F/CCAPI/cc_api_display_ie_subfields:

   ccCallSetupRequest:

   cisco-username=+4917975322866

   ----- ccCallInfo IE subfields -----

   cisco-ani=+4917975322866

   cisco-anitype=0

   cisco-aniplan=0

   cisco-anipi=0

   cisco-anisi=1

   dest=1001

   cisco-desttype=0

   cisco-destplan=0

   cisco-rdie=FFFFFFFF

   cisco-rdn=

   cisco-lastrdn=

   cisco-rdntype=0

   cisco-rdnplan=0

   cisco-rdnpi=-1

   cisco-rdnsi=-1

   cisco-redirectreason=-1   fwd_final_type =0

   final_redirectNumber =

   hunt_group_timeout =0

*Nov  2 23:05:09.779: //8/F2512023800F/CCAPI/ccIFCallSetupRequestPrivate:

   Interface=0x4BAD9944, Interface Type=6, Destination=, Mode=0x0,

   Call Params(Calling Number=+4917975322866,(Calling Name=+4917975322866 )(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),

   Called Number=1001(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,

   Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=20001, Call Count On=FALSE,

   Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)

The Call Setup Information is:

Call Control Block (CCB) : 0x4AB78130

State of The Call        : STATE_DEAD

TCP Sockets Used         : NO

Calling Number           : +4917975322866

Called Number            : 004930129854765

Source IP Address (Sig  ): 192.168.5.99

Destn SIP Req Addr:Port  : 217.10.79.9:5060

Destn SIP Resp Addr:Port : 217.10.79.9:5060

Destination Name         : 217.10.79.9

*Nov  2 23:05:14.575: //8/F2512023800F/SIP/Call/sipSPIMediaCallInfo:

Number of Media Streams: 1

Media Stream             : 1

Negotiated Codec         : g711alaw

Negotiated Codec Bytes   : 160

Nego. Codec payload      : 8 (tx), 8 (rx)

Negotiated Dtmf-relay    : 6

Dtmf-relay Payload       : 101 (tx), 101 (rx)

Source IP Address (Media): 192.168.5.99

Source IP Port    (Media): 17066

Destn  IP Address (Media): 217.10.79.9

Destn  IP Port    (Media): 40144

Orig Destn IP Address:Port (Media): [ - ]:0

*Nov  2 23:05:14.575: //8/F2512023800F/SIP/Call/sipSPICallInfo:

Disconnect Cause (CC)    : 16

Disconnect Cause (SIP)   : 487

**DEBUG OUTGOING**

   Guid=A9C0A698-04DB-11E1-801B-F8B7FAD2A1EA, Outgoing Dial-peer=10

Nov  2 22:48:51.650: //12/A9C0A698801B/CCAPI/cc_api_display_ie_subfields:

   ccCallSetupRequest:

   cisco-username=

   ----- ccCallInfo IE subfields -----

   cisco-ani=004930129854765

   cisco-anitype=0

   cisco-aniplan=0

   cisco-anipi=0

   cisco-anisi=0

   dest=+4917975322866

   cisco-desttype=0

   cisco-destplan=0

   cisco-rdie=FFFFFFFF

   cisco-rdn=

   cisco-lastrdn=

   cisco-rdntype=0

   cisco-rdnplan=0

   cisco-rdnpi=0

   cisco-rdnsi=0

   cisco-redirectreason=0   fwd_final_type =0

   final_redirectNumber =

   hunt_group_timeout =0

Nov  2 22:48:51.650: //12/A9C0A698801B/CCAPI/ccIFCallSetupRequestPrivate:

   Interface=0x4A07C064, Interface Type=3, Destination=, Mode=0x0,

   Call Params(Calling Number=004930129854765,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),

   Called Number=+4917975322866(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,

   Subscriber Type Str=RegularLine, FinalDestinationFlag=FALSE, Outgoing Dial-peer=10, Call Count On=FALSE,

   Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)

The Call Setup Information is:

Call Control Block (CCB) : 0x4ADCE9F0

State of The Call        : STATE_DEAD

TCP Sockets Used         : YES

Calling Number           : 004930129854765

Called Number            : +4917975322866

Source IP Address (Sig  ): 192.168.5.99

Destn SIP Req Addr:Port  : 217.10.79.9:5060

Destn SIP Resp Addr:Port : 217.10.79.9:5060

Destination Name         : sipgate.de

Nov  2 22:48:56.826: //13/A9C0A698801B/SIP/Call/sipSPIMediaCallInfo:

Number of Media Streams: 1

Media Stream             : 1

Negotiated Codec         : No Codec  

Negotiated Codec Bytes   : 0

Nego. Codec payload      : 255 (tx), 255 (rx)

Negotiated Dtmf-relay    : 0

Dtmf-relay Payload       : 0 (tx), 0 (rx)

Source IP Address (Media): 192.168.5.99

Source IP Port    (Media): 18588

Destn  IP Address (Media):  -

Destn  IP Port    (Media): 0

Orig Destn IP Address:Port (Media): [ - ]:0

Nov  2 22:48:56.826: //13/A9C0A698801B/SIP/Call/sipSPICallInfo:

Disconnect Cause (CC)    : 38

Disconnect Cause (SIP)   : 503

1 ACCEPTED SOLUTION

Accepted Solutions
New Member

Re: Outgoing Calls SIP Provider (CME)

Hm, the debug you provided before is showing a 503 disconnect cause.

Is your uploaded config still accurate (since it's from march '10)? Your latest debugs reveal a different ITSP (telefonica instead of sipgate).

Are you sure the provider is working with SIP over TCP? Is the CME able to do a DNS-lookup for the record in the dialpeer?

Test both with:

dial-peer 10

session transport udp

end

ping sipgate.de

6 REPLIES
New Member

Outgoing Calls SIP Provider (CME)

C`mon Voice Gurus. :-)

Any advice please.

Whats going wrong here ?

Regards

Jason

New Member

Outgoing Calls SIP Provider (CME)

Can you post a 'debug ccsip messages' of both calls as well?

New Member

Re: Outgoing Calls SIP Provider (CME)

Yes, of course. But if i dial out. No debug appear. Why? See attched the incoming debug

New Member

Re: Outgoing Calls SIP Provider (CME)

Hm, the debug you provided before is showing a 503 disconnect cause.

Is your uploaded config still accurate (since it's from march '10)? Your latest debugs reveal a different ITSP (telefonica instead of sipgate).

Are you sure the provider is working with SIP over TCP? Is the CME able to do a DNS-lookup for the record in the dialpeer?

Test both with:

dial-peer 10

session transport udp

end

ping sipgate.de

New Member

Re: Outgoing Calls SIP Provider (CME)

Hm, the debug you provided before is showing a 503 disconnect cause.

No, i have sent you the incoming debug. The other debug with  the error 503 was for outgoing. And for "debug ccsip messages" i didn`t get any output if i try to dial out

Is your uploaded config still accurate (since it's from march '10)?

yes, its the same. Where do you see March?

Your latest debugs reveal a different ITSP (telefonica instead of sipgate). 

Sorry, i modified the config a little bit for the public.

Are you sure the provider is working with SIP over TCP? Is the CME able to do a DNS-lookup for the record in the dialpeer?

See next answer .

Test both with:

dial-peer 10

session transport udp

end

You are the man. That was the issue. Now it works. . Great. THANKS,

ping sipgate.de

CME#ping sipgate.de

Type escape sequence to abort.

Sending 5, 100-byte ICMP Echos to 217.10.79.9, timeout is 2 seconds:

!!!!!

Success rate is 100 percent (5/5), round-trip min/avg/max = 36/41/56 ms

CME#

New Member

Re: Outgoing Calls SIP Provider (CME)

I was referring to the debug of the outgoiing (failing) call. It show SIP 503 disconnect cause and CC (call control) cause 38. This indicates networking issue e.g. no DNS/ip routing.

You better should set the global transport option to UDP, since that's the default anyway and used by most of the ITSP:

voice service voip

sip

no transport tcp

In this case you should see errors in some kind of TCP-debug because you're trying to establish TCP-sessions with a non-responsive host.

ps the creation date of your profile (march '10) confused me with the post creation date.

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