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

Answered Question
Sep 26th, 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,

I have this problem too.
0 votes
Correct Answer by Ayodeji oladipo... about 1 year 6 months ago

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

Correct Answer by chrysostomos1980 about 1 year 6 months ago

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

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Average Rating: 5 (3 ratings)
chrysostomos1980 Thu, 09/27/2012 - 00:19

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?)

claudio_rivas Thu, 09/27/2012 - 06:45

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.

chrysostomos1980 Thu, 09/27/2012 - 07:01

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

claudio_rivas Thu, 09/27/2012 - 11:53

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.

Correct Answer
chrysostomos1980 Thu, 09/27/2012 - 12:29

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

claudio_rivas Thu, 09/27/2012 - 12:53

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,

claudio_rivas Thu, 09/27/2012 - 14:44

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.

claudio_rivas Fri, 09/28/2012 - 13:01

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,

chrysostomos1980 Fri, 09/28/2012 - 23:23

Hi Claudio

Config Alcatel for Fast Start on the H323 connection.

Sent from Cisco Technical Support iPhone App

claudio_rivas Mon, 10/01/2012 - 10:21

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.

chrysostomos1980 Mon, 10/01/2012 - 10:59

Hi claudio

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

Can you send another one debug?

chrysostomos1980 Mon, 10/01/2012 - 14:04

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

Ayodeji oladipo... Mon, 10/01/2012 - 14:08

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

claudio_rivas Tue, 10/02/2012 - 11:26

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 Tue, 10/02/2012 - 11:31

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

claudio_rivas Tue, 10/02/2012 - 11:40

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 Tue, 10/02/2012 - 11:45

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

claudio_rivas Tue, 10/02/2012 - 12:53

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 Thu, 10/04/2012 - 10:59

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 oladipo... Tue, 10/02/2012 - 11:56

Claudio,

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

Please rate all useful posts

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

Correct Answer
Ayodeji oladipo... Tue, 10/02/2012 - 13:08

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 Tue, 10/02/2012 - 13:28

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 oladipo... Tue, 10/02/2012 - 13:46

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

amendozar Tue, 10/02/2012 - 13:52

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!

Ayodeji oladipo... Thu, 10/04/2012 - 12:43

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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Posted September 26, 2012 at 11:56 PM
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