Solved: Re: can make only one outbound call and then busy tone - CME using FXO - Page 2

rajkamath · ‎10-17-2022

Good day,

Trying to get a outbound call working through a single line connected to fxo port. first call always works, ( have tested with international, national and mobile ). i hang up and try calling again, it gives a busy tone till i reload the router. Anything to check or try to get that sorted out. Given below is the debug logs that showed up on the router when dialing out. Voice ports are showing on hook . sh run attached.

Router#sh voice port su

Router#sh voice port summary

IN OUT

PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC

=============== == ============ ===== ==== ======== ======== ==

0/1/0 -- fxo-ls up dorm idle on-hook y

0/1/1 -- fxo-ls up dorm idle on-hook y

0/2/0 -- fxo-ls up dorm idle on-hook y

0/2/1 -- fxo-ls up dorm idle on-hook y

0/2/2 -- fxo-ls up dorm idle on-hook y

0/2/3 -- fxo-ls up dorm idle on-hook y

PWR FAILOVER PORT PSTN FAILOVER PORT

================= ==================

Router#

445: *Oct 17 13:23:48.853: //50/C47E15DC8068/

------------------ Cover Buffer ---------------

Search-key = 801:0508363235:50

Timestamp = *Oct 17 13:23:48.251

CallID = 50

Peer-CallID = NA

Correlator = NA

Called-Number = 0508363235

Calling-Number = 801

SIP CallID = 00df1d88-6a200007-40dbbd29-2451d9cc@10.90.80.3

SIP SessionID =

GUID = C47E15DC8068

-----------------------------------------------

416: *Oct 17 13:23:48.251: //50/C47E15DC8068/CUBE_VT/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:0508363235@10.90.80.1;user=phone SIP/2.0

Via: SIP/2.0/UDP 10.90.80.3:5060;branch=z9hG4bK47f93dda

From: "Basel Thaher " <sip:801@10.90.80.1>;tag=00df1d886a20001c31ce4143-3531f195

To: <sip:0508363235@10.90.80.1>

Call-ID: 00df1d88-6a200007-40dbbd29-2451d9cc@10.90.80.3

Max-Forwards: 70

Session-ID: 4dfbd3e200105000a00000df1d886a20;remote=0000000000000000000000000000 0000

Date: Mon, 17 Oct 2022 13:23:47 GMT

CSeq: 101 INVITE

User-Agent: Cisco-CP7811/14.1.1

Contact: <sip:1A694-BD@10.90.80.3:5060;transport=udp>;+u.sip!devicename.ccm.cisc o.com="SEP00DF1D886A20"

Expires: 180

Accept: application/sdp

Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO

Remote-Party-ID: "Basel Thaher " <sip:801@10.90.80.1>;party=calling;id-type=subs criber;privacy=off;screen=yes

Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X- cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X- cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0,X-cisco-xsi- 8.5.1

Allow-Events: kpml,dialog

Recv-Info: conference

Recv-Info: x-cisco-conference

Authorization: Digest username="801",realm="",uri="sip:0508363235@10.90.80.1;use r=phone",response="d3cc0a64befe157d795b079ad6492ec0",nonce="17175C020025C2A1",cn once="23389a8e",qop=auth,nc=00000002,algorithm=MD5

Content-Length: 345

Content-Type: application/sdp

Content-Disposition: session;handling=optional

v=0

o=Cisco-SIPUA 28297 0 IN IP4 10.90.80.3

s=SIP Call

b=AS:4064

t=0 0

m=audio 24366 RTP/AVP 0 8 116 18 101

c=IN IP4 10.90.80.3

b=TIAS:64000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:116 iLBC/8000

a=fmtp:116 mode=20

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

420: *Oct 17 13:23:48.251: //50/C47E15DC8068/CUBE_VT/SIP/FSM/SPI-State-Change: C urrent State = STATE_NONE, Next State = STATE_IDLE, Current Sub-State = STATE_NO NE, Next Sub-State = STATE_NONE

421: *Oct 17 13:23:48.252: //50/C47E15DC8068/CUBE_VT/SIP/MISC/Matched Dialpeer: Dir:Inbound, Peer-Tag: 40001

422: *Oct 17 13:23:48.252: //50/C47E15DC8068/CUBE_VT/SIP/FSM/Offer-Answer: Event = E_SIP_INVITE_SDP_RCVD, Current State = S_SIP_EARLY_DIALOG_IDLE, Next State = S_SIP_EARLY_DIALOG_OFFER_RCVD

423: *Oct 17 13:23:48.253: //50/C47E15DC8068/CUBE_VT/SIP/FSM/IWF: Event = E_SIP_ IWF_EV_RCVD_SDP, Current State = S_SIP_IWF_SDP_IDLE, Next State = S_SIP_IWF_SDP_ RCVD_AWAIT_PEER_EVENT

424: *Oct 17 13:23:48.253: //50/C47E15DC8068/CUBE_VT/SIP/MISC/Media Stream Param eters: Stream Type = voice-only, Stream State = STREAM_ADDING Negotiated Codec = g711ulaw, Negotiated DTMF Type = inband-voice, Stream Index = 1

425: *Oct 17 13:23:48.253: //50/C47E15DC8068/CUBE_VT/SIP/API: cc_api_update_inte rface_cac_resource (0)

426: *Oct 17 13:23:48.253: //50/C47E15DC8068/CUBE_VT/SIP/API: voip_rtp_allocate_ port (8024)

427: *Oct 17 13:23:48.253: //50/C47E15DC8068/CUBE_VT/SIP/MISC/Media Stream Param eters: Stream Type = voice-only, Stream State = STREAM_ADDING Negotiated Codec = g711ulaw, Negotiated DTMF Type = inband-voice, Stream Index = 1

428: *Oct 17 13:23:48.253: //50/C47E15DC8068/CUBE_VT/SIP/API: cc_api_call_setup_ ind_with_callID (0)

429: *Oct 17 13:23:48.253: //50/C47E15DC8068/CUBE_VT/SIP/FSM/SPI-State-Change: C urrent State = STATE_IDLE, Next State = STATE_RECD_INVITE, Current Sub-State = S TATE_NONE, Next Sub-State = STATE_NONE

430: *Oct 17 13:23:48.255: //50/C47E15DC8068/CUBE_VT/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 10.90.80.3:5060;branch=z9hG4bK47f93dda

From: "Basel Thaher " <sip:801@10.90.80.1>;tag=00df1d886a20001c31ce4143-3531f195

To: <sip:0508363235@10.90.80.1>

Date: Mon, 17 Oct 2022 13:23:48 GMT

Call-ID: 00df1d88-6a200007-40dbbd29-2451d9cc@10.90.80.3

CSeq: 101 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-17.3.5

Session-ID: 00000000000000000000000000000000;remote=4dfbd3e200105000a00000df1d88 6a20

Content-Length: 0

431: *Oct 17 13:23:48.255: //50/C47E15DC8068/CUBE_VT/SIP/MISC/Call Disconnect: I nitiated at: 0x260070A, Originated at:0x260070B, Cause Code = 28

432: *Oct 17 13:23:48.255: //50/C47E15DC8068/CUBE_VT/SIP/API: cc_api_update_inte rface_cac_resource (0)

433: *Oct 17 13:23:48.255: //50/C47E15DC8068/CUBE_VT/SIP/FSM/Event-Action: Event = SIPSPI_EV_CC_CALL_DISCONNECT, Current State = STATE_RECD_INVITE

434: *Oct 17 13:23:48.255: //50/C47E15DC8068/CUBE_VT/SIP/FSM/SPI-State-Change: C urrent State = STATE_RECD_INVITE, Next State = STATE_DISCONNECTING, Current Sub- State = STATE_NONE, Next Sub-State = STATE_NONE

435: *Oct 17 13:23:48.256: //50/C47E15DC8068/CUBE_VT/SIP/FSM/SPI-State-Change: C urrent State = STATE_DISCONNECTING, Next State = STATE_DISCONNECTING, Current Su b-State = STATE_NONE, Next Sub-State = STATE_NONE

436: *Oct 17 13:23:48.256: //50/C47E15DC8068/CUBE_VT/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 484 Address Incomplete

Via: SIP/2.0/UDP 10.90.80.3:5060;branch=z9hG4bK47f93dda

From: "Basel Thaher " <sip:801@10.90.80.1>;tag=00df1d886a20001c31ce4143-3531f195

To: <sip:0508363235@10.90.80.1>;tag=18523C8-EEF

Date: Mon, 17 Oct 2022 13:23:48 GMT

Call-ID: 00df1d88-6a200007-40dbbd29-2451d9cc@10.90.80.3

CSeq: 101 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-17.3.5

Reason: Q.850;cause=28

Session-ID: 4dfbd3e200105000a00000df1d886a20;remote=0cb9ec974c0e5853b09e6ceb62c6 3c7c

Content-Length: 0

438: *Oct 17 13:23:48.356: //50/C47E15DC8068/CUBE_VT/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:0508363235@10.90.80.1;user=phone SIP/2.0

Via: SIP/2.0/UDP 10.90.80.3:5060;branch=z9hG4bK47f93dda

From: "Basel Thaher " <sip:801@10.90.80.1>;tag=00df1d886a20001c31ce4143-3531f195

To: <sip:0508363235@10.90.80.1>;tag=18523C8-EEF

Call-ID: 00df1d88-6a200007-40dbbd29-2451d9cc@10.90.80.3

Session-ID: 4dfbd3e200105000a00000df1d886a20;remote=4dfbd3e200105000a00000df1d88 6a20

Max-Forwards: 70

Date: Mon, 17 Oct 2022 13:23:47 GMT

CSeq: 101 ACK

Content-Length: 0

439: *Oct 17 13:23:48.356: //50/C47E15DC8068/CUBE_VT/SIP/FSM/Event-Action: Event = SIPSPI_EV_NEW_MESSAGE, Current State = STATE_DISCONNECTING

440: *Oct 17 13:23:48.357: //50/C47E15DC8068/CUBE_VT/SIP/API: voip_rtp_release_p ort (8024)

441: *Oct 17 13:23:48.357: //50/C47E15DC8068/CUBE_VT/SIP/API: cc_api_call_discon nect_done (0)

442: *Oct 17 13:23:48.357: //50/C47E15DC8068/CUBE_VT/SIP/FSM/SPI-State-Change: C urrent State = STATE_DISCONNECTING, Next State = STATE_DEAD, Current Sub-State = STATE_NONE, Next Sub-State = STATE_NONE

443: *Oct 17 13:23:48.357: //50/C47E15DC8068/CUBE_VT/SIP/MISC/Error: sipSPIFlush DeferredQueue: Invalid deferredQueue

444: *Oct 17 13:23:48.357: //50/C47E15DC8068/CUBE_VT/SIP/API: voip_rtp_release_p ort (8024)

Router#

Router#sh voice

Router#sh voice po

Router#sh voice port su

Router#sh voice port summary

IN OUT

PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC

=============== == ============ ===== ==== ======== ======== ==

0/1/0 -- fxo-ls up dorm idle on-hook y

0/1/1 -- fxo-ls up dorm idle on-hook y

0/2/0 -- fxo-ls up dorm idle on-hook y

0/2/1 -- fxo-ls up dorm idle on-hook y

0/2/2 -- fxo-ls up dorm idle on-hook y

0/2/3 -- fxo-ls up dorm idle on-hook y

PWR FAILOVER PORT PSTN FAILOVER PORT

================= ==================

Roger Kallberg · ‎10-17-2022

With a single line there is really no point in having a trunk group, especially not with multiple FXO ports added to it. If you do want to keep the trunk group you should remove the other FXO ports from it, leaving only the operational FXO port in the trunk group. That would in essence be the same as having one single FXO port on the dial peers.

Roger Kallberg · ‎10-17-2022

With the trunk group configuration you have the call would flip between your FXO ports depending on how long they have been idle. Meaning that the call would not go to the operational FXO port on the second call as that then is the least idle as it just handled a call. As I and @TechLvr stated start by using a single port, easiest would be to use the port command on your dial peers or if you do not want to be easy remote all, but the one port, from your trunk group.

rajkamath · ‎10-17-2022

that worked. been under the impression that since they are not being utilized, it doesn't matter. just added single port to the trunkgroup and calls are flowing through without any issues.

Thanks again.

Roger Kallberg · ‎10-17-2022

Glad to hear that.

can make only one outbound call and then busy tone - CME using FXO por