[cisco-voip] Very new to cisco VoIP products have a few questions/problem.

Mon Oct 13 09:31:50 EDT 2008

Sure, I sanitized everything below.

sip-ua
 no remote-party-id
 retry invite 2
 retry register 10
 timers connect 100
 registrar ipv4:10.2.3.59 expires 3600
 sip-server ipv4:10.2.3.59
  host-registrar

Oct 13 13:24:47.236: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentGTD: No GTD foun
d in inbound container
Oct 13 13:24:47.236: //3664/1F5499CCA5D8/SIP/Transport/sipSPISendAck: Sending AC
K to the transport layer
Oct 13 13:24:47.236: //3664/1F5499CCA5D8/SIP/Transport/sipSPIGetSwitchTransportF
lag: Return the Dial peer configuration, Switch Transport is FALSE
Oct 13 13:24:47.236: //3664/1F5499CCA5D8/SIP/Transport/sipSPITransportSendMessag
e: msg=0x8679644C, addr=10.2.3.59, port=5060, sentBy_port=5060, is_req=0, tra
nsport=1, switch=0, callBack=0x00000000
Oct 13 13:24:47.236: //3664/1F5499CCA5D8/SIP/Transport/sipSPITransportSendMessag
e: Proceedable for sending msg immediately
Oct 13 13:24:47.236: //3664/1F5499CCA5D8/SIP/Transport/sipTransportLogicSendMsg:
 switch transport is 0
Oct 13 13:24:47.236: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage
: Posting send for msg=0x8679644C, addr=10.2.3.59, port=5060, connId=0 for UD
P
Oct 13 13:24:47.236: //3664/1F5499CCA5D8/SIP/Info/act_recdproc_new_message_respo
nse: Received a 4/5/6xx message with StatusCode: 503
Oct 13 13:24:47.236: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_set_release_source_for_pee
r: ownCallId[3664], src[4]
Oct 13 13:24:47.236: //3664/1F5499CCA5D8/SIP/Info/sipSPIInitiateDisconnect: Init
iate call disconnect(63) for outgoing call
Oct 13 13:24:47.240: //3664/1F5499CCA5D8/SIP/State/sipSPIChangeState: 0x86F75D9C
 : State change from (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)  to
 (STATE_DISCONNECTING, SUBSTATE_NONE)
Oct 13 13:24:47.240: //3664/1F5499CCA5D8/SIP/Info/ccsip_call_statistics: Request
ing stats for callid=3664
Oct 13 13:24:47.240: //3664/1F5499CCA5D8/SIP/Info/ccsip_call_statistics: Stats r
equest failed for callid=3664, dstCallID=-1, rc=-7
Oct 13 13:24:47.240: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event f
rom SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
Oct 13 13:24:47.240: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event
: ccsip_spi_get_msg_type returned: 3 for event 7
Oct 13 13:24:47.240: //3664/1F5499CCA5D8/SIP/Info/sipSPIIcpifUpdate: CallState:
2 Playout: 0 DiscTime:6848233 ConnTime 0
Oct 13 13:24:47.240: //3664/1F5499CCA5D8/SIP/State/sipSPIChangeState: 0x86F75D9C
 : State change from (STATE_DISCONNECTING, SUBSTATE_NONE)  to (STATE_DEAD, SUBST
ATE_NONE)
Oct 13 13:24:47.240: //3664/1F5499CCA5D8/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x86F75D9C
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 6144234xxx
Called Number            : 53713xx
Source IP Address (Sig  ): 10.1.0.22
Destn SIP Req Addr:Port  : 10.2.3.59:5060
Destn SIP Resp Addr:Port : 10.2.3.59:5060
Destination Name         : 10.2.3.59

Oct 13 13:24:47.240: //3664/1F5499CCA5D8/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): 10.1.0.22
Source IP Port    (Media): 17116
Destn  IP Address (Media): 0.0.0.0
Destn  IP Port    (Media): 0
Orig Destn IP Address:Port (Media): 0.0.0.0:0

Oct 13 13:24:47.240: //3664/1F5499CCA5D8/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 63
Disconnect Cause (SIP)   : 503

Oct 13 13:24:47.240: //3664/1F5499CCA5D8/SIP/Info/sipSPIUdeleteccCallIdFromTable
: Removing call id E50
Oct 13 13:24:47.244: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:5371369 at 10.2.3.59:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.22:5060;branch=z9hG4bK5A5209A
From: "Drew Weaver" <sip:6144234xxx at 10.2.3.59>;tag=414F434-1F76
To: <sip:5371369 at 10.2.3.59>;tag=3432893595-30855
Date: Mon, 13 Oct 2008 13:24:47 GMT
Call-ID: 23C10A3F-986111DD-A5E5D7B8-B954D02E at voice.localdomain
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

From: Ryan West [mailto:rwest at zyedge.com]
Sent: Monday, October 13, 2008 9:24 AM
To: Drew Weaver; 'cisco-voip at puck.nether.net'
Subject: RE: [cisco-voip] Very new to cisco VoIP products have a few questions/problem.

Drew,

Can you post a ‘deb ccsip messages’ output and also a sanitized version of your sip-ua section?

-ryan

From: Drew Weaver [mailto:drew.weaver at thenap.com]
Sent: Monday, October 13, 2008 9:04 AM
To: 'cisco-voip at puck.nether.net'
Subject: RE: [cisco-voip] Very new to cisco VoIP products have a few questions/problem.

Outbound dial-peer:

dial-peer voice 1001 voip
 description ** Outgoing call to SIP trunk (Generic SIP Trunk Provider) **
 translation-profile outgoing PSTN_Outgoing
 destination-pattern 9T
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 ip qos dscp cs5 media
 ip qos dscp cs4 signaling
 no vad

voice translation-profile PSTN_Outgoing
 translate calling 1111
 translate called 1112
 translate redirect-target 410
 translate redirect-called 410

voice class codec 1
 codec preference 1 g711ulaw

From: Ryan West [mailto:rwest at zyedge.com]
Sent: Monday, October 13, 2008 8:47 AM
To: Ed Leatherman; Drew Weaver
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Very new to cisco VoIP products have a few questions/problem.

To build on what Ed has already added.  When you talk with your SIP provider, ask them what type of authentication they are expecting.  Some providers will do source IP with no authentication, others support a username and password on dial-peers, and others require the hidden ‘credential username password realm’ under sip-ua.  Try running ‘show sip-ua reg stat’ and see what shows as registered.  A lot of providers won’t like the extra registrations that CME / voice gateways send, so if you see 1 registration with a yes and a bunch that say no, you can trim all those down with ‘no sip-register’ under each of the offending dial-peers.

-ryan

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ed Leatherman
Sent: Monday, October 13, 2008 8:37 AM
To: Drew Weaver
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Very new to cisco VoIP products have a few questions/problem.

Sounds like the SIP messages are going back and forth OK but your audio stream isn't. Perhaps ACL or firewall causing the problem?

We've got a UC520 setup at a small office across town, but no SIP trunk so i'm not sure what exactly would be the cause.

CME is running as part of the IOS on the box, if it weren't running your phones wouldn't work at all. I would double check the dialpeers that CCA generated for you. From my limited experience CCA was OK for a basic configuration but as soon as we had to setup something special I got fed up with it and just started using the CLI instead.

Barring that you might want to check with your SIP provider to see if they can help with the outbound call problems.. maybe the call is getting rejected by them?

Ed
On Sun, Oct 12, 2008 at 9:04 PM, Drew Weaver <drew.weaver at thenap.com<mailto:drew.weaver at thenap.com>> wrote:

                Hi there,

--
Ed Leatherman
Assistant Director, Voice Services
West Virginia University
Telecommunications and Network Operations
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