Solved: Re: Cisco Webex calling LGW - outbound calls failing

touma.kayal · ‎01-29-2024

Dears,

I have configured a new sandbox for Webex calling, and i have enabled LGW on my Local ISR 4k (4321) router.

the router shows as online on the control hub, and i am as well able to make inbound calls from ITSP to the Webex calling extension

but making outbound calls through the same ITSP failed with the below error log on the router

*Jan 29 12:55:39.020: %SYS-6-TTY_EXPIRE_TIMER: (exec timer expired, tty 0 (0.0.0.0)), user
1022: *Jan 29 12:59:24.117: //446/0F0B38B28198/
------------------ Cover Buffer ---------------
Search-key = +96181279344:+9613359722:446
Timestamp = *Jan 29 12:59:23.287
CallID = 446
Peer-CallID = NA
Correlator = NA
Called-Number = +9613359722
Calling-Number = +96181279344
SIP CallID = SSE1259228292901241978982203@139.177.67.11
SIP SessionID =
GUID = 0F0B38B28198
Tenant = 0
Cause-code = bearer capability not implemented (65)
-----------------------------------------------
1000: *Jan 29 12:59:23.288: //446/0F0B38B28198/CUBE_VT/SIP/Msg/ccsipDisplayMsg:
Received: SIP TLS message from 139.177.67.11:8934 to 192.168.0.113:5061
INVITE sip:+9613359722@192.168.0.113:5061;transport=tls;dtg=lgwtrunk9796_lgu SIP/2.0
Via:SIP/2.0/TLS 139.177.67.11:8934;branch=z9hG4bKBroadworksSSE.-213.204.114.213V42437-0-100-366463071-1706533162829-
From:"tkayal tkayal"<sip:+96181279344@139.177.67.11;user=phone>;tag=366463071-1706533162829-
To:<sip:+9613359722@98303641.us10.bcld.webex.com;user=phone>
Call-ID:SSE1259228292901241978982203@139.177.67.11
CSeq:100 INVITE
Contact:<sip:139.177.67.11:8934;transport=tls>
P-Asserted-Identity:"tkayal tkayal"<sip:+96181279344@10.71.100.200;user=phone>
Privacy:none
Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Recv-Info:x-broadworks-client-session-info,x-cisco-mute-status
X-Cisco-Location-Info:077a38c7-a316-489a-a886-91f11da7410b;country=LB
X-BroadWorks-Correlation-Info:5d405603-adc4-41b1-9572-eafbe32b2dd3
Accept:application/dtmf-relay,application/media_control+xml,application/sdp,multipart/mixed
Supported:
Max-Forwards:69
Session-ID:1998f32001205000800051e3bfb5a4c2;remote=00000000000000000000000000000000
Content-Type:application/sdp
Content-Length:1433

v=0
o=BroadWorks 173799819 1706533162826 IN IP4 139.177.67.15
s=-
c=IN IP4 139.177.67.15
t=0 0
a=cisco-mari:v1
a=cisco-mari-rate
a=cisco-mari-rtx:v0
a=cisco-mari-hybrid-resilience:v0
m=audio 20774 RTP/SAVP 99 9 8 0 18 102 101 111 112
a=sendrecv
a=rtpmap:99 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 iLBC/8000
a=fmtp:102 mode=30
a=rtpmap:101 telephone-event/8000
a=rtpmap:111 x-ulpfecuc/8000
a=fmtp:111 max_esel=1400;max_n=255;m=8;multi_ssrc=1;FEC_ORDER=FEC_SRTP;non_seq=1;feedback=0
a=rtpmap:112 mari-rtx/90000
a=fmtp:112 RTX_ORDER=RTX_SRTP;rtx-time=180
a=extmap:4/sendrecv http://protocols.cisco.com/timestamp#100us
a=extmap:9/sendrecv http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=ice-ufrag:BW6N
a=ice-pwd:IBLbzzmRWOn+Y3OcuuLgaraOqAZPXnHi
a=candidate:1 1 UDP 2113935615 192.168.43.227 8574 typ host
a=candidate:1 2 UDP 2113935614 192.168.43.227 8575 typ host
a=candidate:3 1 UDP 1677727999 91.232.101.128 32478 typ srflx raddr 192.168.43.227 rport 8574
a=candidate:3 2 UDP 1677727998 91.232.101.128 33961 typ srflx raddr 192.168.43.227 rport 8575
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=candidate:mse 1 UDP 16777215 139.177.67.15 20774 typ relay
a=candidate:mse 2 UDP 16777214 139.177.67.15 20775 typ relay

1006: *Jan 29 12:59:23.288: //446/0F0B38B28198/CUBE_VT/SIP/FSM/SPI-State-Change: Current State: STATE_NONE, Next State: STATE_IDLE, Current Sub-State: STATE_NONE, Next Sub-State: STATE_NONE
1007: *Jan 29 12:59:23.289: //446/0F0B38B28198/CUBE_VT/SIP/MISC/Matched Dialpeer: Dir: Inbound, Peer-Tag: 0
1008: *Jan 29 12:59:23.289: //446/0F0B38B28198/CUBE_VT/SIP/MISC/Error: ccsip_ipip_media_forking_anchor_leg_config: MF: Dial-peer is absent..
1009: *Jan 29 12:59:23.289: //446/0F0B38B28198/CUBE_VT/SIP/MISC/Error: ccsip_ipip_media_forking_intra_frame_request_config: MF:video profile Dial-peer is absent..
1010: *Jan 29 12:59:23.290: //446/0F0B38B28198/CUBE_VT/SIP/MISC/Error: sipSPIDoMediaNegotiation: Failed to negotiate main stream. Main stream dead
1011: *Jan 29 12:59:23.290: //446/0F0B38B28198/CUBE_VT/SIP/MISC/Call Disconnect: Initiated at: 0x1700862, Originated at: 0x4103D66, Cause Code: 65
1012: *Jan 29 12:59:23.290: //446/0F0B38B28198/CUBE_VT/SIP/FSM/SPI-State-Change: Current State: STATE_IDLE, Next State: STATE_DISCONNECTING, Current Sub-State: STATE_NONE, Next Sub-State: STATE_NONE
1013: *Jan 29 12:59:23.290: //446/0F0B38B28198/CUBE_VT/SIP/Msg/ccsipDisplayMsg:
Sent: SIP TLS message from 192.168.0.113:5061 to 139.177.67.11:8934
SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/TLS 139.177.67.11:8934;branch=z9hG4bKBroadworksSSE.-213.204.114.213V42437-0-100-366463071-1706533162829-
From: "tkayal tkayal"<sip:+96181279344@139.177.67.11;user=phone>;tag=366463071-1706533162829-
To: <sip:+9613359722@98303641.us10.bcld.webex.com;user=phone>;tag=8335C3-912
Date: Mon, 29 Jan 2024 12:59:23 GMT
Call-ID: SSE1259228292901241978982203@139.177.67.11
CSeq: 100 INVITE
Allow-Events: telephone-event
Warning: 399 192.168.0.113 "SRTP Offer/Answer not acceptable.RTP configured on dialpeer"
Reason: Q.850;cause=65
Server: Cisco-SIPGateway/IOS-17.12.2
Session-ID: 1998f32001205000800051e3bfb5a4c2;remote=0f6fae05b79c5a97985082c2b5e6b375
Content-Length: 0

1015: *Jan 29 12:59:23.360: //446/0F0B38B28198/CUBE_VT/SIP/Msg/ccsipDisplayMsg:
Received: SIP TLS message from 139.177.67.11:8934 to 192.168.0.113:5061
ACK sip:+9613359722@192.168.0.113:5061;transport=tls;dtg=lgwtrunk9796_lgu SIP/2.0
Via:SIP/2.0/TLS 139.177.67.11:8934;branch=z9hG4bKBroadworksSSE.-213.204.114.213V42437-0-100-366463071-1706533162829-
From:"tkayal tkayal"<sip:+96181279344@139.177.67.11;user=phone>;tag=366463071-1706533162829-
To:<sip:+9613359722@98303641.us10.bcld.webex.com;user=phone>;tag=8335C3-912
Call-ID:SSE1259228292901241978982203@139.177.67.11
CSeq:100 ACK
Session-ID:0f6fae05b79c5a97985082c2b5e6b375;remote=1998f32001205000800051e3bfb5a4c2
Max-Forwards:10
Content-Length:0

1017: *Jan 29 12:59:23.360: //446/0F0B38B28198/CUBE_VT/SIP/FSM/Event-Action: Event: SIPSPI_EV_NEW_MESSAGE, Current State: STATE_DISCONNECTING
1018: *Jan 29 12:59:23.360: //446/0F0B38B28198/CUBE_VT/SIP/FSM/SPI-State-Change: Current State: STATE_DISCONNECTING, Next State: STATE_DEAD, Current Sub-State: STATE_NONE, Next Sub-State: STATE_NONE
1019: *Jan 29 12:59:23.360: //446/0F0B38B28198/CUBE_VT/SIP/MISC/Error: sipSPIFlushDeferredQueue: Invalid deferredQueue
1020: *Jan 29 12:59:23.360: //446/0F0B38B28198/CUBE_VT/SIP/MISC/Error: ws_call_fork_cleanup: ws_info is NULL
1021: *Jan 29 12:59:23.360: //446/0F0B38B28198/CUBE_VT/SIP/API: voip_rtp_release_port (0)

within the log file i can see "SRTP Offer/Answer not acceptable.RTP configured on dialpeer",

but the router is ISR 4321 and SRTP to RTP interworking is available by default.

To note that the same router with the same configuration was working with a client Webex organization, but stopped working when i moved it to the sandbox tenant

NB: inbound calls from ITSP to service provider is working fine (it is an RTP to SRTP connection as well)

Any suggestions here ?

thank you a lot

b.winter · ‎01-31-2024

You are matching the wrong dial-peer:
539: *Jan 31 10:46:34.502: //444/D61A51FB8180/CUBE_VT/SIP/MISC/Matched Dialpeer: Dir: Inbound, Peer-Tag: 0

So, somewhere in your config, there must be an error.

Please post the full config of the "voice class uri 200 sip"

View solution in original post

Shalid Kurunnan Chalil · ‎01-29-2024

Hi there,

Can you cross-check check your Webex calling dial peer (assuming using the same DP for both inbound and outbound) does have the "srtp" configured?

if it is there and still experiencing the issue, could you please share your dial-peer config here?

Regards

Shalid

b.winter · ‎01-29-2024

As @Shalid Kurunnan Chalil indicated, please post the full config (without any usernamen, passwords) to get the complete overview of your config.

touma.kayal · ‎01-31-2024

Hello @b.winter and @Shalid Kurunnan Chalil ,

attached you can find the config

to note that inbound from ITSP to Webex calling is working fine

but from Webex calling extension to ITSP is not working

b.winter · ‎01-31-2024

Please also post the output of following debugs for an outgoing call. My guess is, that you are matching the wrong incoming dial-peer.
debug ccsip messages
debug voice ccapi ind 1
debug voice ccapi ind 2
debug voice ccapi ind 74

And please don't put the debug output here in the message again. It makes the thread completely unreadable.
Save it as a text file and upload the file. Like you did for the config.

Furthermore:
You don't need 2 different tenants for the connection towards the provider. The template from Cisco is not 100% ideal, but as a technician, you shouldn't just copy/paste everything.
You can delete tenant 300 and use tenant 100 in dial-peer 100.

touma.kayal · ‎01-31-2024

hello @b.winter ,

That is absolutely true regarding the dial peer and the SIP tenant , but as i was facing that issue, i went through the document again and replicate the configuration suggested by cisco. in anyway, find attached below the requested logs:

=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2024.01.31 12:46:20 =~=~=~=~=~=~=~=~=~=~=~=

*Jan 31 10:46:15.304: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:192.168.5.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.150:5060;branch=z9hG4bK55808d61;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.5.150>;tag=as5b115a58
To: <sip:192.168.5.2>
Contact: <sip:Unknown@192.168.5.150>
Call-ID: 0c5e5dbb2fbe59420689939f3b592189@192.168.5.150
CSeq: 102 OPTIONS
Date: Thu, 01 Jan 1970 03:10:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

*Jan 31 10:46:15.306: //442/CAA93EA9817E/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.150:5060;branch=z9hG4bK55808d61;rport
From: "Unknown" <sip:Unknown@192.168.5.150>;tag=as5b115a58
To: <sip:192.168.5.2>;tag=AC3459-100C
Date: Wed, 31 Jan 2024 10:46:15 GMT
Call-ID: 0c5e5dbb2fbe59420689939f3b592189@192.168.5.150
Server: Cisco-SIPGateway/IOS-17.12.2
CSeq: 102 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces
Content-Type: application/sdp
Content-Length: 369

v=0
o=CiscoSystemsSIP-GW-UserAgent 9391 2656 IN IP4 192.168.5.2
s=SIP Call
c=IN IP4 192.168.5.2
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
c=IN IP4 192.168.5.2
m=image 0 udptl t38
c=IN IP4 192.168.5.2
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy

Router#
Router#
Router#
*Jan 31 10:46:27.059: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:192.168.0.113:5061;transport=tls SIP/2.0
Via:SIP/2.0/TLS 139.177.67.11:8934;branch=z9hG4bKBroadworksSSE.-213.204.114.213V15477-0-100-1719610156-1706697985730-
From:<sip:139.177.67.11>;tag=1719610156-1706697985730-
To:<sip:192.168.0.113>
Call-ID:SSE104625730310124-1076159296@139.177.67.11
CSeq:100 OPTIONS
Max-Forwards:0
Content-Length:0

*Jan 31 10:46:27.061: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TLS 139.177.67.11:8934;branch=z9hG4bKBroadworksSSE.-213.204.114.213V15477-0-100-1719610156-1706697985730-
From: <sip:139.177.67.11>;tag=1719610156-1706697985730-
To: <sip:192.168.0.113>;tag=AC6244-1F7A
Date: Wed, 31 Jan 2024 10:46:27 GMT
Call-ID: SSE104625730310124-1076159296@139.177.67.11
Server: Cisco-SIPGateway/IOS-17.12.2
CSeq: 100 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event

Router#
Router#
Router#Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces
Content-Type: application/sdp
Content-Length: 377

v=0
o=CiscoSystemsSIP-GW-UserAgent 9754 8938 IN IP4 192.168.0.113
s=SIP Call
c=IN IP4 192.168.0.113
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
c=IN IP4 192.168.0.113
m=image 0 udptl t38
c=IN IP4 192.168.0.113
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy

Router#
Router#
Router#
Router#
Router#
*Jan 31 10:46:34.501: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:+9613803367@192.168.0.113:5061;transport=tls;dtg=lgwtrunk9796_lgu SIP/2.0
Via:SIP/2.0/TLS 139.177.67.11:8934;branch=z9hG4bKBroadworksSSE.-213.204.114.213V15477-0-100-761608332-1706697993172-
From:"tkayal tkayal"<sip:+96170501112@139.177.67.11;user=phone>;tag=761608332-1706697993172-
To:<sip:+9613803367@98303641.us10.bcld.webex.com;user=phone>
Call-ID:SSE104633172310124281027114@139.177.67.11
CSeq:100 INVITE
Contact:<sip:139.177.67.11:8934;transport=tls>
P-Asserted-Identity:"tkayal tkayal"<sip:+96170501112@10.71.100.200;user=phone>
Privacy:none
Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Recv-Info:x-broadworks-client-session-info,x-cisco-mute-status
X-Cisco-Location-Info:077a38c7-a316-489a-a886-91f11da7410b;country=LB
X-BroadWorks-Correlation-Info:e2794c1e-3bd6-4d28-84fa-3f348f69ea50
Accept:application/dtmf-relay,application/media_control+xml,application/sdp,multipart/mixed
Supported:
Max-Forwards:69
Session-ID:37e4f2f30120500080006860bf2e2989;remote=00000000000000000000000000000000
Content-Type:application/sdp
Content-Length:1441

v=0
o=BroadWorks 313121556 1706697993169 IN IP4 139.177.67.19
s=-
c=IN IP4 139.177.67.19
t=0 0
a=cisco-mari:v1
a=cisco-mari-rate
a=cisco-mari-rtx:v0
a=cisco-mari-hybrid-resilience:v0
m=audio 41254 RTP/SAVP 99 9 8 0 18 102 101 111 112
a=sendrecv
a=rtpmap:99 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 iLBC/8000
a=fmtp:102 mode=30
a=rtpmap:101 telephone-event/8000
a=rtpmap:111 x-ulpfecuc/8000
a=fmtp:111 max_esel=1400;max_n=255;m=8;multi_ssrc=1;FEC_ORDER=FEC_SRTP;non_seq=1;feedback=0
a=rtpmap:112 mari-rtx/90000
a=fmtp:112 RTX_ORDER=RTX_SRTP;rtx-time=180
a=extmap:4/sendrecv http://protocols.cisco.com/timestamp#100us
a=extmap:9/sendrecv http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time
a=rtcp-mux
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=ice-ufrag:+tKl
a=ice-pwd:GkveEtDL6GNm/6i5sNPKuwAKDl5PZD89
a=candidate:1 1 UDP 2113935615 192.168.0.103 8548 typ host
a=candidate:1 2 UDP 2113935614 192.168.0.103 8549 typ host
a=candidate:3 1 UDP 1677727999 213.204.114.213 8548 typ srflx raddr 192.168.0.103 rport 8548
a=candidate:3 2 UDP 1677727998 213.204.114.213 8549 typ srflx raddr 192.168.0.103 rport 8549
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=candidate:mse 1 UDP 16777215 139.177.67.19 41254 typ relay
a=candidate:mse 2 UDP 16777214 139.177.67.19 41255 typ relay

*Jan 31 10:46:34.504: //444/D61A51FB8180/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/TLS 139.177.67.11:8934;branch=z9hG4bKBroadworksSSE.-213.204.114.213V15477-0-100-761608332-1706697993172-
From: "tkayal tkayal"<sip:+96170501112@139.177.67.11;user=phone>;tag=761608332-1706697993172-
To: <sip:+9613803367@98303641.us10.bcld.webex.com;user=phone>;tag=AC7F58-26CA
Date: Wed, 31 Jan 2024 10:46:34 GMT
Call-ID: SSE104633172310124281027114@139.177.67.11
CSeq: 100 INVITE
Allow-Events: telephone-event
Warning: 399 192.168.0.113 "SRTP Offer/Answer not acceptable.RTP configured on dialpeer"
Reason: Q.850;cause=65
Server: Cisco-SIPGateway/IOS-17.12.2
Session-ID: 37e4f2f30120500080006860bf2e2989;remote=5b02e48e83645de2b612640a7cbe6fc1
Content-Length: 0

*Jan 31 10:46:34.570: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:+9613803367@192.168.0.113:5061;transport=tls;dtg=lgwtrunk9796_lgu SIP/2.0
Via:SIP/2.0/TLS 139.177.67.11:8934;branch=z9hG4bKBroadworksSSE.-213.204.114.213V15477-0-100-761608332-1706697993172-
From:"tkayal tkayal"<sip:+96170501112@139.177.67.11;user=phone>;tag=761608332-1706697993172-
To:<sip:+9613803367@98303641.us10.bcld.webex.com;user=phone>;tag=AC7F58-26CA
Call-ID:SSE104633172310124281027114@139.177.67.11
CSeq:100 ACK
Session-ID:5b02e48e83645de2b612640a7cbe6fc1;remote=37e4f2f30120500080006860bf2e2989
Max-Forwards:10
Content-Length:0

554: *Jan 31 10:46:34.994: //444/D61A51FB8180/
------------------ Cover Buffer ---------------
Search-key = +96170501112:+9613803367:444
Timestamp = *Jan 31 10:46:34.502
CallID = 444
Peer-CallID = NA
Correlator = NA
Called-Number = +9613803367
Calling-Number = +96170501112
SIP CallID = SSE104633172310124281027114@139.177.67.11
SIP SessionID =
GUID = D61A51FB8180
Tenant = 0
Cause-code = bearer capability not implemented (65)
-----------------------------------------------
532: *Jan 31 10:46:34.502: //444/D61A51FB8180/CUBE_VT/SIP/Msg/ccsipDisplayMsg:
Received: SIP TLS message from 139.177.67.11:8934 to 192.168.0.113:5061
INVITE sip:+9613803367@192.168.0.113:5061;transport=tls;dtg=lgwtrunk9796_lgu SIP/2.0
Via:SIP/2.0/TLS 139.177.67.11:8934;branch=z9hG4bKBroadworksSSE.-213.204.114.213V15477-0-100-761608332-1706697993172-
From:"tkayal tkayal"<sip:+96170501112@139.177.67.11;user=phone>;tag=761608332-1706697993172-
To:<sip:+9613803367@98303641.us10.bcld.webex.com;user=phone>
Call-ID:SSE104633172310124281027114@139.177.67.11
CSeq:100 INVITE
Contact:<sip:139.177.67.11:8934;transport=tls>
P-Asserted-Identity:"tkayal tkayal"<sip:+96170501112@10.71.100.200;user=phone>
Privacy:none
Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Recv-Info:x-broadworks-client-session-info,x-cisco-mute-status
X-Cisco-Location-Info:077a38c7-a316-489a-a886-91f11da7410b;country=LB
X-BroadWorks-Correlation-Info:e2794c1e-3bd6-4d28-84fa-3f348f69ea50
Accept:application/dtmf-relay,application/media_control+xml,application/sdp,multipart/mixed
Supported:
Max-Forwards:69
Session-ID:37e4f2f30120500080006860bf2e2989;remote=00000000000000000000000000000000
Content-Type:application/sdp
Content-Length:1441

v=0
o=BroadWorks 313121556 1706697993169 IN IP4 139.177.67.19
s=-
c=IN IP4 139.177.67.19
t=0 0
a=cisco-mari:v1
a=cisco-mari-rate
a=cisco-mari-rtx:v0
a=cisco-mari-hybrid-resilience:v0
m=audio 41254 RTP/SAVP 99 9 8 0 18 102 101 111 112
a=sendrecv
a=rtpmap:99 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 iLBC/8000
a=fmtp:102 mode=30
a=rtpmap:101 telephone-event/8000
a=rtpmap:111 x-ulpfecuc/8000
a=fmtp:111 max_esel=1400;max_n=255;m=8;multi_ssrc=1;FEC_ORDER=FEC_SRTP;non_seq=1;feedback=0
a=rtpmap:112 mari-rtx/90000
a=fmtp:112 RTX_ORDER=RTX_SRTP;rtx-time=180
a=extmap:4/sendrecv http://protocols.cisco.com/timestamp#100us
a=extmap:9/sendrecv http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time
a=rtcp-mux
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=ice-ufrag:+tKl
a=ice-pwd:GkveEtDL6GNm/6i5sNPKuwAKDl5PZD89
a=candidate:1 1 UDP 2113935615 192.168.0.103 8548 typ host
a=candidate:1 2 UDP 2113935614 192.168.0.103 8549 typ host
a=candidate:3 1 UDP 1677727999 213.204.114.213 8548 typ srflx raddr 192.168.0.103 rport 8548
a=candidate:3 2 UDP 1677727998 213.204.114.213 8549 typ srflx raddr 192.168.0.103 rport 8549
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=candidate:mse 1 UDP 16777215 139.177.67.19 41254 typ relay
a=candidate:mse 2 UDP 16777214 139.177.67.19 41255 typ relay

538: *Jan 31 10:46:34.501: //444/D61A51FB8180/CUBE_VT/SIP/FSM/SPI-State-Change: Current State: STATE_NONE, Next State: STATE_IDLE, Current Sub-State: STATE_NONE, Next Sub-State: STATE_NONE
539: *Jan 31 10:46:34.502: //444/D61A51FB8180/CUBE_VT/SIP/MISC/Matched Dialpeer: Dir: Inbound, Peer-Tag: 0
540: *Jan 31 10:46:34.502: //444/D61A51FB8180/CUBE_VT/SIP/MISC/Error: ccsip_ipip_media_forking_anchor_leg_config: MF: Dial-peer is absent..
541: *Jan 31 10:46:34.502: //444/D61A51FB8180/CUBE_VT/SIP/MISC/Error: ccsip_ipip_media_forking_intra_frame_request_config: MF:video profile Dial-peer is absent..
542: *Jan 31 10:46:34.503: //444/D61A51FB8180/CUBE_VT/SIP/MISC/Error: sipSPIDoMediaNegotiation: Failed to negotiate main stream. Main stream dead
543: *Jan 31 10:46:34.503: //444/D61A51FB8180/CUBE_VT/SIP/MISC/Call Disconnect: Initiated at: 0x1700862, Originated at: 0x4103D66, Cause Code: 65
544: *Jan 31 10:46:34.503: //444/D61A51FB8180/CUBE_VT/SIP/FSM/SPI-State-Change: Current State: STATE_IDLE, Next State: STATE_DISCONNECTING, Current Sub-State: STATE_NONE, Next Sub-State: STATE_NONE
545: *Jan 31 10:46:34.504: //444/D61A51FB8180/CUBE_VT/SIP/Msg/ccsipDisplayMsg:
Sent: SIP TLS message from 192.168.0.113:5061 to 139.177.67.11:8934
SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/TLS 139.177.67.11:8934;branch=z9hG4bKBroadworksSSE.-213.204.114.213V15477-0-100-761608332-1706697993172-
From: "tkayal tkayal"<sip:+96170501112@139.177.67.11;user=phone>;tag=761608332-1706697993172-
To: <sip:+9613803367@98303641.us10.bcld.webex.com;user=phone>;tag=AC7F58-26CA
Date: Wed, 31 Jan 2024 10:46:34 GMT
Call-ID: SSE104633172310124281027114@139.177.67.11
CSeq: 100 INVITE
Allow-Events: telephone-event
Warning: 399 192.168.0.113 "SRTP Offer/Answer not acceptable.RTP configured on dialpeer"
Reason: Q.850;cause=65
Server: Cisco-SIPGateway/IOS-17.12.2
Session-ID: 37e4f2f30120500080006860bf2e2989;remote=5b02e48e83645de2b612640a7cbe6fc1
Content-Length: 0

547: *Jan 31 10:46:34.571: //444/D61A51FB8180/CUBE_VT/SIP/Msg/ccsipDisplayMsg:
Received: SIP TLS message from 139.177.67.11:8934 to 192.168.0.113:5061
ACK sip:+9613803367@192.168.0.113:5061;transport=tls;dtg=lgwtrunk9796_lgu SIP/2.0
Via:SIP/2.0/TLS 139.177.67.11:8934;branch=z9hG4bKBroadworksSSE.-213.204.114.213V15477-0-100-761608332-1706697993172-
From:"tkayal tkayal"<sip:+96170501112@139.177.67.11;user=phone>;tag=761608332-1706697993172-
To:<sip:+9613803367@98303641.us10.bcld.webex.com;user=phone>;tag=AC7F58-26CA
Call-ID:SSE104633172310124281027114@139.177.67.11
CSeq:100 ACK
Session-ID:5b02e48e83645de2b612640a7cbe6fc1;remote=37e4f2f30120500080006860bf2e2989
Max-Forwards:10
Content-Length:0

549: *Jan 31 10:46:34.571: //444/D61A51FB8180/CUBE_VT/SIP/FSM/Event-Action: Event: SIPSPI_EV_NEW_MESSAGE, Current State: STATE_DISCONNECTING
550: *Jan 31 10:46:34.571: //444/D61A51FB8180/CUBE_VT/SIP/FSM/SPI-State-Change: Current State: STATE_DISCONNECTING, Next State: STATE_DEAD, Current Sub-State: STATE_NONE, Next Sub-State: STATE_NONE
551: *Jan 31 10:46:34.572: //444/D61A51FB8180/CUBE_VT/SIP/MISC/Error: sipSPIFlushDeferredQueue: Invalid deferredQueue
552: *Jan 31 10:46:34.572: //444/D61A51FB8180/CUBE_VT/SIP/MISC/Error: ws_call_fork_cleanup: ws_info is NULL
553: *Jan 31 10:46:34.572: //444/D61A51FB8180/CUBE_VT/SIP/API: voip_rtp_release_port (0)
Router#
Router#
Router#un all
All possible debugging has been turned off
Router#

b.winter · ‎01-31-2024

Pleeeease don't spam the thread with endlessly long debug outputs. Save them in a file and upload the file ...

Where is the output of the ALL the debug commands? You should enable all 4 commands and not just 1

touma.kayal · ‎01-31-2024

i tried the upload the file but it kept throwing errors about the file content... although i have uploaded the configuration the same way, but for the debug messages i was not able to upload them.

yes that way all the output of the debug messages for an outbound call

touma.kayal · ‎01-31-2024

b.winter · ‎01-31-2024

You are matching the wrong dial-peer:
539: *Jan 31 10:46:34.502: //444/D61A51FB8180/CUBE_VT/SIP/MISC/Matched Dialpeer: Dir: Inbound, Peer-Tag: 0

So, somewhere in your config, there must be an error.

Please post the full config of the "voice class uri 200 sip"

touma.kayal · ‎01-31-2024

below is the config:

voice class uri 200 sip
pattern dtg=lgwtrunk9796.lgu

b.winter · ‎01-31-2024

I see, that you accepted on message as a solution.
But maybe for others: What did you change? Or what was not correct?

touma.kayal · ‎01-31-2024

@b.winter ,

i had the LGW in caps lock for lgwtrunk9796.lgw, once i changed it to small letters it works

thx you a lot for your great support

much appreciated!!