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SIP Phone outgoing call is not working

Hi Experts,

 

I have 7841 SIP IP phones with CUCME, the outgoing call is not working. The other party phone rings but I cant hear the ring back tone and after few second call disconnected when answered the call.

Incoming call is working perfectly.

I have 7965 SCCP phones where I am not facing any issue. My call flow is given below.

 

7841 IP Phone >>> CUCME >>>> H323>>>>. CUCM 7.1 >>>>> MGCP gateway

I have added CUCME router as H323 gateway in CUCM as SIP trunk was not working even for internal calls to CUCM.

My configuration is given below.

 

voice-card 0
 dsp services dspfarm
!
!
voice call send-alert
voice rtp send-recv
!
voice service voip
 ip address trusted list
  ipv4 172.16.0.0 255.255.0.0
 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
 sip
  bind control source-interface GigabitEthernet0/0
  bind media source-interface GigabitEthernet0/0
  registrar server expires max 3600 min 3600
!
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g711alaw
 codec preference 3 g729r8
!
!
voice register pool-type  7841
 phoneload-support
 telnet-support
 num-lines 6
 description 7841 Phone
 reference-pooltype 6941
!
voice register global
 mode  cme
 source-address 172.16.241.1 port 5060
 max-dn 300
 max-pool 100
 timezone 35
 time-format 24
 hold-alert
 phone-redirect-limit 20
 tftp-path flash:
 file text
 create profile sync 0000523242199018
 camera
 video
!
voice register dn  4
 number 1804
 name Salim Obaid Salim
 label Salim Obaid Salim
!
voice register template  1
 softkeys hold  Resume Newcall iDivert
 softkeys idle  Newcall Redial Pickup Gpickup Cfwdall DND
 softkeys ringIn  Answer DND iDivert
 softkeys connected  Endcall Hold Park Trnsfer iDivert Confrn
 conference admin
!
voice register pool  4
 busy-trigger-per-button 1
 id mac 5067.AEE2.0151
 type 7841
 number 1 dn 4
 template 1

 

sccp local GigabitEthernet0/0
sccp ccm 172.16.241.1 identifier 1 version 7.0
sccp
!
sccp ccm group 1
 bind interface GigabitEthernet0/0
 associate ccm 1 priority 1
 associate profile 1 register conf-dsp-1
 keepalive retries 5
!
!
!
dspfarm profile 1 conference 
 codec g729br8
 codec g729r8
 codec g729abr8
 codec g729ar8
 codec g711alaw
 codec g711ulaw
 maximum sessions 8
 associate application SCCP
!
dial-peer cor custom
 name local
 name mobile
 name NWD
 name ISD
!
!
dial-peer cor list outgoing-local
 member local
!
dial-peer cor list outgoing-mobile
 member mobile
!
dial-peer cor list outgoing-NWD
 member NWD
!
dial-peer cor list outgoing-ISD
 member ISD
!
dial-peer cor list Employee
 member local
!
dial-peer cor list Manager
 member local
 member mobile
 member NWD
!
dial-peer cor list Executive
 member local
 member mobile
 member NWD
 member ISD
!
!
dial-peer voice 201 voip
 corlist outgoing Employee
 destination-pattern 9[2-8]......$
 session target ipv4:172.16.14.5
 incoming called-number .
 dtmf-relay h245-alphanumeric
!
dial-peer voice 202 voip
 corlist outgoing Employee
 destination-pattern 9[19]..$
 session target ipv4:172.16.14.5
 dtmf-relay h245-alphanumeric
 codec g711ulaw
!
dial-peer voice 203 voip
 corlist outgoing Employee
 destination-pattern 9[68]00T
 session target ipv4:172.16.14.5
 dtmf-relay h245-alphanumeric
 codec g711ulaw
!
dial-peer voice 204 voip
 corlist outgoing Manager
 destination-pattern 905........$
 session target ipv4:172.16.14.5
 dtmf-relay h245-alphanumeric
 codec g711ulaw
!

 

!
telephony-service
 sdspfarm units 4
 sdspfarm tag 1 conf-dsp-1
 conference hardware
 no auto-reg-ephone
 em keep-history
 max-ephones 150
 max-dn 200
 ip source-address 172.16.241.1 port 2000
 system message FEWA DHAID
 time-zone 35
 time-format 24
 date-format dd-mm-yy
 max-conferences 8 gain -6
 call-park system redirect
 web admin system name gbm password fewac1sc0
 dn-webedit
 time-webedit
 transfer-system full-consult
 create cnf-files version-stamp Jan 01 2002 00:00:00
!

ephone-dn  1  dual-line
 number 1800
 label Abdalla Naser Alowais
 name Abdalla Naser Alowais

ephone  1
 mac-address 381C.1ABA.9ACF
 type 7965
 button  1:1

I tried to enable MTP at gateway but problem was not resolved. Any idea.

 

Some logs are given below for outgoing call. I also tried with G711ulaw but problem is still persist.

 

DHAID-VG#
*Oct  9 16:00:36.589 UAE: //-1/xxxxxxxxxxxx/SIP/Error/httpish_msg_free:
 Freeing NULL pointer!
*Oct  9 16:00:36.589 UAE: //-1/B6E372EE82E5/SIP/Media/sipSPICopyStunConfigFromPeerToCCB: Firewall traversal is not enabled
SIP: (319) Attribute mid, level 1 instance 1 not found.
*Oct  9 16:00:36.589 UAE: //319/B6E372EE82E5/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 172.16.241.1
*Oct  9 16:00:36.589 UAE: //319/B6E372EE82E5/SIP/Media/sipSPISelectCodecVersion: g729r8 flavor of g729 codec will be used
*Oct  9 16:00:36.589 UAE: //319/B6E372EE82E5/SIP/Media/sipSPIUpdCallWithSdpInfo:
        Preferred Codec        : g729r8, bytes :20
        Preferred  DTMF relay  : inband-voice
        Preferred NTE payload  : 98
        Early Media            : No
        Delayed Media          : No
        Bridge Done            : No
        New Media              : No
        DSP DNLD Reqd          : No

*Oct  9 16:00:36.589 UAE: //319/B6E372EE82E5/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 172.16.241.1
SIP: (319) setup attribute, level 1 instance 1 not found.
SIP: (319) connection attribute, level 1 instance 1 not found.
SIP: (319) Attribute label, level 1 instance 1 not found.
SIP: (319) a=framerate attribute, level 1 instance 1 not found.
*Oct  9 16:00:36.589 UAE: //319/B6E372EE82E5/SIP/Media/sipSPISelectCodecVersion: g729r8 flavor of g729 codec will be used
*Oct  9 16:00:36.589 UAE: //319/B6E372EE82E5/SIP/Media/sipSPIDisplayStreamInfo:
          Stream type            : voice-only
          Media line             : 1
          State                  : STREAM_ADDING (2)
          Stream address type    : 1
          Callid                 : 319
          Peer Callid            : -1
          RTP/SRTP Negotiated     : 8
          Negotiated Codec       : g729r8, bytes :20
          Nego. Codec payload    : 18 (tx), 18 (rx)
          Negotiated DTMF relay  : inband-voice
          Negotiated NTE payload : 0 (tx), 0 (rx)
          Negotiated CN payload  : 0
          Media Srce Addr/Port   : [172.16.241.1]:0
          Media Dest Addr/Port   : [172.16.241.7]:19718

*Oct  9 16:00:36.589 UAE: //319/B6E372EE82E5/SIP/Media/sipSPIUpdCallWithSdpInfo:
          Stream type            : voice-only
          Media line             : 1
          State                  : STREAM_ADDING (2)
          Stream address type    : 1
          Callid                 : 319
          Negotiated Codec       : g729r8, bytes :20
          Nego. Codec payload    : 18 (tx), 18 (rx)
          Negotiated DTMF relay  : inband-voice
          Negotiated NTE payload : 0 (tx), 0 (rx)
          Negotiated CN payload  : 0
          Media Srce Addr/Port   : [172.16.241.1]:0
          Media Dest Addr/Port   : [172.16.241.7]:19718

*Oct  9 16:00:36.589 UAE: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 16780 for stream 1
*Oct  9 16:00:36.589 UAE: //319/B6E372EE82E5/SIP/Media/sipSPIDisplayStreamInfo:
          Stream type            : voice-only
          Media line             : 1
          State                  : STREAM_ADDING (2)
          Stream address type    : 1
          Callid                 : 319
          Peer Callid            : -1
          RTP/SRTP Negotiated     : 8
          Negotiated Codec       : g729r8, bytes :20
          Nego. Codec payload    : 18 (tx), 18 (rx)
          Negotiated DTMF relay  : inband-voice
          Negotiated NTE payload : 0 (tx), 0 (rx)
          Negotiated CN payload  : 0
          Media Srce Addr/Port   : [172.16.241.1]:16780
          Media Dest Addr/Port   : [172.16.241.7]:19718

*Oct  9 16:00:39.607 UAE: //319/B6E372EE82E5/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_ADDING
*Oct  9 16:00:39.607 UAE: //319/B6E372EE82E5/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice-only (callid 319) to the VOIP RTP library
*Oct  9 16:00:39.607 UAE: //319/B6E372EE82E5/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 172.16.241.1
*Oct  9 16:00:39.607 UAE: //319/B6E372EE82E5/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1
*Oct  9 16:00:39.607 UAE: //319/B6E372EE82E5/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info
        laddr = 172.16.241.1, lport = 16780, raddr = 172.16.241.7, rport=19718, do_rtcp=TRUE
        src_callid = 319, dest_callid = 320, stream type = voice-only, stream direction = SENDRECV
        media_ip_addr = 172.16.241.7, vrf tableid = 0 media_addr_type = 1       negotiated_bandwidth (kbps) = 0 srtp_services = 0 nat_flag = 0
*Oct  9 16:00:39.607 UAE: //319/B6E372EE82E5/SIP/Media/sipSPIUpdateRtcpSession: No rtp session, creating a new one
*Oct  9 16:00:39.607 UAE: //319/B6E372EE82E5/SIP/Media/sipSPICreateRtpSession: stun is disabled
*Oct  9 16:00:39.607 UAE: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: Voice quality monitoring is not enabled for this RTP session due to sdp passthru enabled
*Oct  9 16:00:39.607 UAE: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: VQM: gccb=0x0, gccb->callId=0, ccb->ccCallID=319
*Oct  9 16:00:39.607 UAE: //319/B6E372EE82E5/SIP/Media/sipSPIGetNewLocalMediaDirection:
        New Remote Media Direction = SENDRECV
        Present Local Media Direction = SENDRECV
        New Local Media Direction = SENDRECV
        retVal = 0

*Oct  9 16:00:39.607 UAE: //319/B6E372EE82E5/SIP/Error/sipSPI_ipip_set_history_info_header:
 Not SIP2SIP mode
*Oct  9 16:00:43.009 UAE: //319/B6E372EE82E5/SIP/Error/sipSPI_ipip_set_history_info_header:
 Not SIP2SIP mode
*Oct  9 16:00:51.617 UAE: //319/B6E372EE82E5/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:1FCFD46C
*Oct  9 16:00:51.625 UAE: //319/B6E372EE82E5/SIP/Media/sipSPIHandleDestroyRtpSession: stream:1FCFD46C
*Oct  9 16:01:42.951 UAE: //322/8043CB300700/SIP/Error/sipSPIGetCallServerGroupTargets:
 No server group configured
*Oct  9 16:01:42.951 UAE: //322/8043CB300700/SIP/Media/sipSPICopyStunConfigFromPeerToCCB: Firewall traversal is not enabled
SIP: Attribute mid, level 1 instance 1 not found.
*Oct  9 16:01:45.745 UAE: //322/8043CB300700/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 172.16.241.1
*Oct  9 16:01:45.745 UAE: //322/8043CB300700/SIP/Media/sipSPISelectCodecVersion: g729r8 flavor of g729 codec will be used
*Oct  9 16:01:45.745 UAE: //322/8043CB300700/SIP/Media/sipSPIUpdCallWithSdpInfo:
        Preferred Codec        : g729r8, bytes :20
        Preferred  DTMF relay  : inband-voice
        Preferred NTE payload  : 98
        Early Media            : No
        Delayed Media          : Yes
        Bridge Done            : No
        New Media              : No
        DSP DNLD Reqd          : No

*Oct  9 16:01:45.745 UAE: //322/8043CB300700/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 172.16.241.1
*Oct  9 16:01:45.745 UAE: //322/8043CB300700/SIP/Media/sipSPISelectCodecVersion: g729r8 flavor of g729 codec will be used
*Oct  9 16:01:45.745 UAE: //322/8043CB300700/SIP/Media/sipSPISetStreamInfo: 0 Active Streams
*Oct  9 16:01:45.745 UAE: //322/8043CB300700/SIP/Media/sipSPISetStreamInfo: Number of active streams is zero (0)!
*Oct  9 16:01:45.745 UAE: //322/8043CB300700/SIP/Media/sipSPISetStreamInfo:
caps.stream_count=0,caps.stream[0].stream_type=0xFFFF, caps.stream_list.xmitFunc=
*Oct  9 16:01:45.745 UAE: //322/8043CB300700/SIP/Media/sipSPISetStreamInfo: ??unknown??, caps.stream_list.context=
*Oct  9 16:01:45.745 UAE: //322/8043CB300700/SIP/Media/sipSPISetStreamInfo: 0x0 (gccb)
*Oct  9 16:01:45.745 UAE: //322/8043CB300700/SIP/Media/sipSPIUpdCallWithSdpInfo:
          Stream type            : voice-only
          Media line             : 1
          State                  : STREAM_ADDING (2)
          Stream address type    : 1
          Callid                 : -1
          Negotiated Codec       : g729r8, bytes :20
          Nego. Codec payload    : 18 (tx), 18 (rx)
          Negotiated DTMF relay  : inband-voice
          Negotiated NTE payload : 0 (tx), 0 (rx)
          Negotiated CN payload  : 0
          Media Srce Addr/Port   : [172.16.241.1]:0
          Media Dest Addr/Port   : [172.16.241.7]:30700

*Oct  9 16:01:45.745 UAE: //322/8043CB300700/SIP/Error/sipSPICheckAndClearSrcSRTPSdp:
 CCB SDP source pointer NULL
*Oct  9 16:01:45.763 UAE: //322/8043CB300700/SIP/Media/sipSPISetStreamInfo: 0 Active Streams
*Oct  9 16:01:45.763 UAE: //322/8043CB300700/SIP/Media/sipSPISetStreamInfo: Number of active streams is zero (0)!
*Oct  9 16:01:45.763 UAE: //322/8043CB300700/SIP/Media/sipSPISetStreamInfo:
caps.stream_count=0,caps.stream[0].stream_type=0xFFFF, caps.stream_list.xmitFunc=
*Oct  9 16:01:45.763 UAE: //322/8043CB300700/SIP/Media/sipSPISetStreamInfo: ??unknown??, caps.stream_list.context=
*Oct  9 16:01:45.763 UAE: //322/8043CB300700/SIP/Media/sipSPISetStreamInfo: 0x0 (gccb)
*Oct  9 16:01:45.787 UAE: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 16786 for stream 1
*Oct  9 16:01:45.787 UAE: //322/8043CB300700/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_ADDING
*Oct  9 16:01:45.789 UAE: //322/8043CB300700/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice-only (callid 322) to the VOIP RTP library
*Oct  9 16:01:45.789 UAE: //322/8043CB300700/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 172.16.241.1
*Oct  9 16:01:45.789 UAE: //322/8043CB300700/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1
*Oct  9 16:01:45.789 UAE: //322/8043CB300700/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info
        laddr = 172.16.241.1, lport = 16786, raddr = 172.16.241.7, rport=30700, do_rtcp=TRUE
        src_callid = 322, dest_callid = 321, stream type = voice-only, stream direction = SENDRECV
        media_ip_addr = 172.16.241.7, vrf tableid = 0 media_addr_type = 1       negotiated_bandwidth (kbps) = 0 srtp_services = 0 nat_flag = 0
*Oct  9 16:01:45.789 UAE: //322/8043CB300700/SIP/Media/sipSPIUpdateRtcpSession: No rtp session, creating a new one
*Oct  9 16:01:45.789 UAE: //322/8043CB300700/SIP/Media/sipSPICreateRtpSession: stun is disabled
*Oct  9 16:01:45.789 UAE: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: Voice quality monitoring is not enabled for this RTP session due to sdp passthru enabled
*Oct  9 16:01:45.789 UAE: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: VQM: gccb=0x0, gccb->callId=0, ccb->ccCallID=322
*Oct  9 16:01:45.789 UAE: //322/8043CB300700/SIP/Media/sipSPIGetNewLocalMediaDirection:
        New Remote Media Direction = SENDRECV
        Present Local Media Direction = SENDRECV
        New Local Media Direction = SENDRECV
        retVal = 0

*Oct  9 16:01:45.789 UAE: //322/8043CB300700/SIP/Error/sipSPIHandleSDPOwnerVersionIDChange:
 Pointers are NULL..
*Oct  9 16:01:54.827 UAE: //322/8043CB300700/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:20900924
*Oct  9 16:01:54.919 UAE: //322/8043CB300700/SIP/Media/sipSPIHandleDestroyRtpSession: stream:20900924

 

3 REPLIES
Cisco Employee

voice register pool  4add--

voice register pool  4

add-->codec g711ulaw

voice register global

no create pro

create prof

and enable below debugs and send the logs

deb ccsip mess

deb voip ccapi inou

deb h225 asn1

deb h245 asn1

 

New Member

Hi I tried with G711ulaw but

Hi I tried with G711ulaw but problem was same.

any other idea.

New Member

Hi I just delete the SIP

Hi I just delete the SIP trunk on CUCM and recreate it and it started working.

Very amazed.

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