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

CLAUDIO RIVAS
Level 1
Level 1

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,

2 Accepted Solutions

Accepted Solutions

Hi Claudio

Just some thoughts

its better to have (incoming called number .) for calls coming from alcatel through h323 (h323 dial peer)and also the same for calls coming from sip trunk(sip dial peer)

1)Are you able to ping the isp ip

2) incoming calls are working?

3) so the provider is using a calling number athentication(or no authentication at all)

4) The calling number is what isp expecting?

5) did you ask isp to be involed on that?Help with trace etc?

Could you pls answer the above

We will be waiting for the debug ccsip messages

Regards

chrysostomos

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

View solution in original post

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?

Please rate all useful posts

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

Please rate all useful posts

View solution in original post

29 Replies 29

Hi

I am wondering if the sip trunk is register with the isp

show sip-ua registration status

debug ccsip messages

One question

calls from alcatel are g711 or g729

what protocol have the alcatel phones(sip?)

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

hi, calls from alcatel are g711alaw and the alcatel phones are analog connected to the pbx, which send the calls to the voice gateway through h323 protocol.

No sip-ua registration because everything is on dial-peers configuration, carrirer is not having authentication.

I noticed there is also a SIP Cause code 481 also, but i think this could be caused depending if the people on the pstn phone answers when the phone rings, if this is distracting lets focus on 487.

2012-09-14 09:57:52          Local7.Notice          172.16.4.2          5825009: *Sep 14 09:38:42: %SYS-5-CONFIG_I: Configured from console by sistlajo on vty0 (172.16.1.43)

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:

2012-09-14 10:41:51          Local7.Debug          172.16.4.2          5825041: *Sep 14 10:22:41: //2141792/80EFAF09D636/SIP/Call/sipSPICallInfo:

2012-09-14 10:41:51          Local7.Debug          172.16.4.2          5825042: The Call Setup Information is:

2012-09-14 10:41:51          Local7.Debug          172.16.4.2          5825043: Call Control Block (CCB) : 0x9AA6D58

2012-09-14 10:41:51          Local7.Debug          172.16.4.2          5825044: State of The Call        : STATE_DEAD

2012-09-14 10:41:51          Local7.Debug          172.16.4.2          5825045: TCP Sockets Used         : NO

2012-09-14 10:41:51          Local7.Debug          172.16.4.2          5825046: Calling Number           : 3324004900

2012-09-14 10:41:51          Local7.Debug          172.16.4.2          5825047: Called Number            : 37700028

2012-09-14 10:41:51          Local7.Debug          172.16.4.2          5825048: Source IP Address (Sig  ): 10.15.0.22

2012-09-14 10:41:51          Local7.Debug          172.16.4.2          5825049: Destn SIP Req Addr:Port  : 10.255.252.134:5060

2012-09-14 10:41:51          Local7.Debug          172.16.4.2          5825050: Destn SIP Resp Addr:Port : 10.255.252.134:5060

2012-09-14 10:41:51          Local7.Debug          172.16.4.2          5825051: Destination Name         : 10.255.252.134

2012-09-14 10:41:51          Local7.Debug          172.16.4.2          5825052:

2012-09-14 10:41:51          Local7.Debug          172.16.4.2          5825053: *Sep 14 10:22:41: //2141792/80EFAF09D636/SIP/Call/sipSPICallInfo:

2012-09-14 10:41:51          Local7.Debug          172.16.4.2          5825054: Disconnect Cause (CC)    : 16

2012-09-14 10:41:51          Local7.Debug          172.16.4.2          5825055: Disconnect Cause (SIP)   : 487

2012-09-14 10:41:51          Local7.Debug          172.16.4.2          5825056:

Thank you for your quick response and regards,

Claudio.

Hi

Two questions

What is the ip:172.16.250.29

What is the ip :10.255.252.134

Pls send debug ccsip messages, debug ccsip errors in VG

what is the version of your ios?

Regards

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

Thank you Chrys,

Version 15.0(1)M4

The ip address 172.16.250.29 - The RTP is passed by an rtp forwarder on the carrier side. Seems they handle the RTP on another equipment. I don't think this is an impact, i have a CUCM also using this gateway and sip trunk but on this scenario doesn't have problems.

Currently to place the test again i need to modify the scenario from the Alcatel side and this impacts the user, I'll try to get the debugs as soon as posible.

Regards,

Claudio.

Hi Claudio

Just some thoughts

its better to have (incoming called number .) for calls coming from alcatel through h323 (h323 dial peer)and also the same for calls coming from sip trunk(sip dial peer)

1)Are you able to ping the isp ip

2) incoming calls are working?

3) so the provider is using a calling number athentication(or no authentication at all)

4) The calling number is what isp expecting?

5) did you ask isp to be involed on that?Help with trace etc?

Could you pls answer the above

We will be waiting for the debug ccsip messages

Regards

chrysostomos

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

Thank you Chrys,

1.- im able to ping the isp

2.- incomming calls are on another trunk, this is just for outbound

3.- no authentication on the carrier side for sip

4.- yes, they requested us to change the 4 digit extension format (ej. 3811@ip address for a 10 digit number (ej 3312000122@ipaddress)

5.- the carrier is involved

I'm preparing the scenario to collect the debugs.

Regards,

Hi,

I taked the debugs from the gateway, thank you for your patience, the callflow is:

alcatel->voice gateway->sip trunk

The caller keeps listening ringback even if the call.

The pstn phone listen silence for a while when the call is answered, but suddently listen fast busy as if the call where disconnected.

Best regards,

Claudio.

Hi Chris, i've noticed about a response from you this morning about changing a parameter on the Alcatel, but now that i'm on site i can't get your post to retrieve the value that you suggested, can you repeat me the value that you suggested on your last/deleted post?

Regards,

Hi Claudio

Config Alcatel for Fast Start on the H323 connection.

Sent from Cisco Technical Support iPhone App

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

Hi,

Configured alcatel for Destination: Gateway Fast Start on the Alcatel OmniPCX

Also tryied configured on gateway the following lines:

voice service voip

h323

  call start fast

but the result is the same.

On the destination number there is just silence.

Regards,

Claudio.

Hi claudio

Did you check without the fast start enable in the cube?

Can you send another one debug?

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

Hi Chrys,

Please find attached the logs on this post, thank you for your patience, here is the log file that i retrieved.

You will identify the calls by the dialed number 37700028.

Best regards,

Claudio.

Hi

First test:

ip route 10.15.0.22  255.255.255.255  10.15.0.254

Test one call with debug

Second test:

Remove the bind commans into sip

Test one call with debug

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

Hi Claudio,

I have looked at your traces and here is what I see..

1. Your CUBE sends an invite with delayed offer, that means you do not send any SDP in your invite

2. In this scenario, you cant enable fast start on your h323 trunk becasue cube does not support delayed offer to fast start support.

3. Something is not quite wright with your sip integration to alcatel..

Alcaltel does not send the required sip messages such as "trying", 183 ringing..Insteand alcatel sends 2oo ok without any SDP parameter...then sends 183 with media change, and sends it multiple times because cube does not send any ACK to it. I belive the reason is because cube does not understand what the message is for..

So here is my advice

1. Try and enable early offer on your cube

voice service voip

sip

early-offer forced.

Then test again..If this doesnt work, you will need to check CCO for how to integrate your version of Alcatel with CUCM and CUBE

Please rate all useful posts

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

Please rate all useful posts
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