Solved: sip Disconnect Cause (CC): 16 Disconnect Cause (SIP): 487 call k - Page 2

CLAUDIO RIVAS · ‎09-26-2012

Hi,

Currently i have an scenario with

alcatel (h323) -> 2921 -> SIP Trunk -> Carrier Softswitch

When i call to a PSTN number form an alcatel extension by going through the SIP Trunk from the alcatel extension i herd ringback, and on the PSTN phone rings, when the phone is answered then there is a silence on the call and then sudently drop the call.

I collected the sip debg info

2012-09-14 10:08:43 Local7.Notice 172.16.4.2 5825010: *Sep 14 09:49:34: %SYS-5-CONFIG_I: Configured from console by sistlajo on vty0 (172.16.1.43)

2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825011: *Sep 14 10:06:59: //2141100/00809F331317/SIP/Call/sipSPICallInfo:

2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825012: The Call Setup Information is:

2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825013: Call Control Block (CCB) : 0x9AA6D58

2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825014: State of The Call : STATE_DEAD

2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825015: TCP Sockets Used : NO

2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825016: Calling Number : 3324004900

2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825017: Called Number : 37700028

2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825018: Source IP Address (Sig ): 10.15.0.22

2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825019: Destn SIP Req Addr:Port : 10.255.252.134:5060

2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825020: Destn SIP Resp Addr:Port : 10.255.252.134:5060

2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825021: Destination Name : 10.255.252.134

2012-09-14 10:26:08 Local7.Debug 172.16.4.2 5825022:

2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825023: *Sep 14 10:06:59: //2141100/00809F331317/SIP/Call/sipSPIMediaCallInfo:

2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825024: Number of Media Streams: 1

2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825025: Media Stream : 1

2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825026: Negotiated Codec : g711alaw

2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825027: Negotiated Codec Bytes : 160

2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825028: Nego. Codec payload : 8 (tx), 8 (rx)

2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825029: Negotiated Dtmf-relay : 6

2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825030: Dtmf-relay Payload : 101 (tx), 101 (rx)

2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825031: Source IP Address (Media): 10.15.0.22

2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825032: Source IP Port (Media): 0

2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825033: Destn IP Address (Media): 172.16.250.29

2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825034: Destn IP Port (Media): 31100

2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825035: Orig Destn IP Address:Port (Media): [ - ]:0

2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825036:

2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825037: *Sep 14 10:06:59: //2141100/00809F331317/SIP/Call/sipSPICallInfo:

2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825038: Disconnect Cause (CC) : 16

2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825039: Disconnect Cause (SIP) : 481

2012-09-14 10:26:09 Local7.Debug 172.16.4.2 5825040:

Here is the meaningful part of my configuration:

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

redirect ip2ip

fax protocol pass-through g711ulaw

h323

call preserve

modem passthrough nse codec g711ulaw

--More-- sip

bind control source-interface GigabitEthernet0/1

bind media source-interface GigabitEthernet0/1

!

voice class codec 10

codec preference 1 g711ulaw

codec preference 2 g711alaw

codec preference 3 g729r8

codec preference 4 g729br8

!

voice class codec 11

codec preference 1 g729r8

codec preference 2 g711alaw

codec preference 3 g711ulaw

codec preference 4 g729br8

!

voice class codec 12

codec preference 1 g711alaw

codec preference 2 g711ulaw

codec preference 3 g729r8

codec preference 4 g729br8

!

voice class h323 1

--More-- h225 timeout tcp establish 3

!

voice translation-rule 10

rule 1 /$[^9].*$/ /9\1/

!

voice translation-rule 20

rule 1 /^9/ //

!

voice translation-rule 21

rule 1 /..../ /3324004900/

!

interface GigabitEthernet0/0

description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$

ip address 172.16.4.2 255.255.255.0

duplex auto

speed auto

h323-gateway voip interface

h323-gateway voip bind srcaddr 172.16.4.2

!

interface GigabitEthernet0/1

ip address 10.15.0.22 255.255.255.0

duplex auto

--More-- speed auto

media-type rj45

no cdp enable

!

ip route 0.0.0.0 0.0.0.0 172.16.4.1

ip route 10.255.252.134 255.255.255.255 10.15.0.254

ip route 172.16.250.0 255.255.255.0 10.15.0.254

ip route 172.16.253.0 255.255.255.0 10.15.0.254

!

dial-peer voice 4011 voip

description ~-~-~-~-~-~-~-Dir 937700028~-~-~-~-~-~-~-

translation-profile outgoing SalidaSIP2

preference 2

destination-pattern 937700028

session protocol sipv2

session target ipv4:10.255.252.134

session transport udp

voice-class codec 12

dtmf-relay rtp-nte h245-alphanumeric h245-signal cisco-rtp sip-kpml sip-notify

no vad

as you can notice i hardcoded the phone number i'm trying to reach,

Hope to find help soon, on advice thanks!

Regards,

CLAUDIO RIVAS · ‎10-02-2012

Hi Aok,

Thank you for your response, i placed the early-offer forced under the sip configuration, but stills the same result.

Sorry about the question but, On CCO you mean Cisco website?

Regards,

chrysostomos1980 · ‎10-02-2012

Hi

Aok means to login into cisco.com with your cco(Cisco account) and check configuration guide for your case

Personal i believe that the issue is with the ISP , but its better to check it again that your config is correct , between alcatel and CUBE

Please rate all useful posts Regards Chrysostomos ""The Most Successful People Are Those Who Are Good At Plan B""

CLAUDIO RIVAS · ‎10-02-2012

For me seems an issue with Alcatel and CUBE, because currently i'm ussing the same trunk and even same gateway and dial-peers for dial out from a CallManager on the same network without problems, but at this point i'm short of ideas.

Best regards,

chrysostomos1980 · ‎10-02-2012

Claudio

Do you have cco in cisco?

If no tell me your alcatel model and i will try to find the confguration guide

Also ISP must be able to give you some ideas about the issue i think and structure you what you have to change regarding their traces

One question:

Did you create a dial peer for calls coming to cube from alcatel?

dial-peer voice 100 voip

incoming called-number .

voice-class h323 1

voice-class codec 12

dtmf-relay h245-alphanumeric

fax rate disable

no vad

Please rate all useful posts Regards Chrysostomos ""The Most Successful People Are Those Who Are Good At Plan B""

CLAUDIO RIVAS · ‎10-02-2012

Hi Chrys,

Yes i got one, but cant find a document for Alcatel OmniPCX Office, just for the 4400 .

Regarding the dial-peer I have this configuration

dial-peer voice 2 voip

incoming called-number .

voice-class codec 11

dtmf-relay h245-alphanumeric

and about the voice class codec i have this configuration:

voice class codec 10

codec preference 1 g711ulaw

codec preference 2 g711alaw

codec preference 3 g729r8

codec preference 4 g729br8

!

voice class codec 11

codec preference 1 g729r8

codec preference 2 g711alaw

codec preference 3 g711ulaw

codec preference 4 g729br8

!

voice class h323 1

h225 timeout tcp establish 3

!

So i think this could be a codec missconfiguration, as this kind of calls uses first the g729r8 but the call from alcatel to voice gateway is suposed to be stablished by g711alaw and the calls from the voice gateway to the sip trunk uses the same codec (g711alaw).

My next try will be to use your configuration once i get on site tomorrow.

I would like to know whats your deep thinking about this, for me seems interesting and logicaly correct what you are suggesting but i need to understand a little bit more about this.

Regards,

Claudio.

CLAUDIO RIVAS · ‎10-04-2012

Hi Chrys,

I've tryied with the configuration for the dial-peer as you suggested:

dial-peer voice 2 voip

incoming called-number .

voice-class h323 1

voice-class codec 12

dtmf-relay h245-alphanumeric

fax rate disable

no vad

The main result is the same, the caller listen ringing until fast busy, and the pstn phone listen nothing but silence until the call drops. The difference is that the caller and callee have the fast busy faster than the other times, i suppose that this is because the order of the codec is first on the g711alaw for the voice class 12.

I attached some logs, which are the same for Aok.

Regards,

Claudio.

Ayodeji Okanlawon · ‎10-02-2012

Claudio,

Please send your CUBE traces after enabling early offer..Let me know the called and calling number

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

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CLAUDIO RIVAS · ‎10-02-2012

Hi Aok,

Here is the log file, i think it's not complete, tomorrow i will place same test again. Do you think i need any particular debugs?

Regards,

Ayodeji Okanlawon · ‎10-02-2012

Is this the called number 37980899 ?

Can you explain the set up better...Is this diagram correct?

Alcatel---h323--->CUBE--sip trunk-->-ITSP

Where are you making the call from..Is it from alcatel to Carried? or carrier to alcatel?

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CLAUDIO RIVAS · ‎10-02-2012

Hi Aok,

Sorry about the correct answer, was an error, the call flow is correct, the called number was 37700028.

Regarding the call, yes is starting on alcatel (extensions 38XX) and going through the SIP trunk to the carrier.

Regards,

Ayodeji Okanlawon · ‎10-02-2012

Ok. The called number is not in the trace you sent..

Can you send your full config of the cube..

can you also do another test call and send

debug voip ccapi inout

debug ccsip messages

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

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Arturo Alejandro Mendoza Rodriguez · ‎10-02-2012

Hi Claudio,

Who is your ITSP?, for TELUM without Auth, here is my outbound config

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none

h323

sip

g729 annexb-all

!

voice class codec 1

codec preference 1 g729br8 bytes 30

codec preference 2 g729r8 bytes 30

codec preference 3 g711ulaw

!

voice translation-rule 3

rule 1 /^.*/ /8122223333/

!

voice translation-rule 110

rule 1 /^9/ //

!

voice translation-profile TELUM

translate calling 3

translate called 110

!

dial-peer voice 2002 voip

description [][][] Llamadas Nacionales to TELUM [][][]

translation-profile outgoing TELUM

destination-pattern 901..........

session protocol sipv2

session target ipv4:<>:5060

incoming called-number .T

voice-class codec 1

voice-class sip early-offer forced

dtmf-relay rtp-nte

!

best regards!

CLAUDIO RIVAS · ‎10-02-2012

HI Aok,

Thanks for your quick answer, i'll get debugs tomorrow morning. pls find on the doc the information requested.

Regards,

CLAUDIO RIVAS · ‎10-04-2012

Hi Aok,

I attached on this post the debugs you requested, the call is from 37700028 so you can find it.

Regards,

Claudio.

Ayodeji Okanlawon · ‎10-04-2012

Claudio,

In the trace I can see that the gateway used dial-peer 4011. However i do not see the configuration for that dial-peer. I also do not see the early offer forced config I suggested earlier. Is this config you sent an old config. Can you send the correct config.

ct 4 11:54:45: //2601943/00809F33070D/CCAPI/ccSaveDialpeerTag:

2012-10-04 12:16:31,local7.debug,172.16.4.2, 27821960: Outgoing Dial-peer=4011

ct 4 11:54:45: //2601943/00809F33070D/CCAPI/ccSaveDialpeerTag:

2012-10-04 12:16:31,local7.debug,172.16.4.2, 27821960: Outgoing Dial-peer=4011

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

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sip Disconnect Cause (CC): 16 Disconnect Cause (SIP): 487 call keep ringing