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

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.

View solution in original post

Roger Kallberg · ‎10-17-2022

From what I can see your call is not successful as you get Cause Code = 28, this equates to invalid number format (address incomplete).

Roger Kallberg · ‎10-17-2022

As on your other post where you wrote that you have attached the running configuration there are no attached file on this post either. Please share your running configuration.

Maren Mahoney · ‎10-17-2022

Can you post your dial-peers and any digit manipulation (like voice translation profiles or sip profiles)? I see the SIP side of the call, but not the analog side. So it is hard to tell how the gateway is egressing the call.

Maren

rajkamath · ‎10-17-2022

These are the dial peers i added. I havent used any digit manipulation. Anything else to try on the system .

dial-peer voice 1 pots
trunkgroup FXO
destination-pattern 90[234679].......
prefix 0
dial-peer voice 2 pots
trunkgroup FXO
destination-pattern 9[2-8][0-9][1-9]....
forward-digits 7
dial-peer voice 5 pots
trunkgroup FXO
destination-pattern 900T
prefix 00
dial-peer voice 7 pots
trunkgroup FXO
destination-pattern 9800T
prefix 800
dial-peer voice 6 pots
trunkgroup FXO
destination-pattern 9600T
prefix 600
dial-peer voice 3 pots
trunkgroup FXO
destination-pattern 905[0456].......
prefix 05

rajkamath · ‎10-17-2022

sh run attached. There is no translation rule applied. is that required ? The first call, local , or mobile or even international rings without any issues. Once you hung up , then the next call is possible is only if you reload the router.

Roger Kallberg · ‎10-17-2022

As you use a trunk group on your dial peers for your FXO ports do you know if all of your ports in the trunk group are operational?

Apart from this what do you mean by this “call working through a single line connected to fxo port” as you have grouped your FXO ports into a trunk group?

If you only have one operational FXO port you should remove the other, non operational, ports from the trunk group or use a single port on your dial peers.

TechLvr · ‎10-17-2022

@rajkamath If you dial 0508363235, it finds no match in your voice gateway.

But if you dial the number as 90508363235, then it would match dial peer 3 shown below.

dial-peer voice 3 pots
trunkgroup FXO
destination-pattern 905[0456].......
prefix 05

The resulting dialed number that leaves the FXO port would then be 0508363235.

rajkamath · ‎10-17-2022

The number should be 0508363235. It worked on the first call that the phone does. So confused. Will certainly give that a shot and post back.

rajkamath · ‎10-17-2022

That didnt help. also tried taking the trunk group off and using port numbers one by one. still gives busy tone. reload the box and dial out. works properly for a single call and back to square one again.

TechLvr · ‎10-17-2022

That may be an IOS issue then. Have you done a bug search on your IOS version?

Roger Kallberg · ‎10-17-2022

One small correction on your reply. The resulting number would be 0508363235 as the 905 would be consumed on the dial peer as they are explicitly matched. This is because a POTS dial peer will per default remove anything that is specifically matched on the dial peer and 905 is, but 0456 is not as they are within square brackets. In essence what the dial peer does is to drop the leading 9 as it prefix’s 05 back to the called number to result in 0508363235.

TechLvr · ‎10-17-2022

Roger, Thanks for the correction. You are right. I totally forgot about how that worked until you reminded me. I went ahead and edited my response.

rajkamath · ‎10-17-2022

testing it with a single analogue line at the moment . should i take that trunk group off since its a single line. first timer. so its been a mix of reading and seeking guidance on how and what to do and what not to .thanks again.

why does it work on the first instance and then it goes down.

TechLvr · ‎10-17-2022

Try taking the trunk group off and test with one port at a time. Instead of "trunk-group FXO" just use "port 0/1/0" and test again.

If that doesn't work, then try "port 0/1/1" and so on.