01-08-2012 06:44 AM - edited 03-16-2019 08:52 AM
Hello ALL,
an using uc540 i have config cisco ip phones 504 among others and using a dialogic gateway to attribute numbers to the cisco phones they have 5004-5007they number attribution enable calls entering from GSM to reach this phones 5004-5007
but upto now they ring but on the gsm number it doesnt ring and when i capture traffic at the level of the dialgic i see that there is invitefrom dialogic but there is no ack
locally they communicate to together but when calls are coming outside that when it not having ACK.
when call from GSM the cisco phone ringo and shows the GSM number, but on the GSM piece it doesnt Ring and even when the cisco phone user picks up the call there no sign to show that the is on hook
Solved! Go to Solution.
01-08-2012 07:05 AM
You should include the complete trace, as the above is not complete.
However, is seem that the GSM box is on a public IP, while the UC500 is on a private IP behind a NAT and/or firewall router, likely not a Cisco device..
In that case, it's normal that doesn't work. You need to have them both on the same local network.
Otherwise, you will have major problems to make it work.
01-08-2012 07:27 AM
Actually, the trace shows that both devices have private address, and are behind two different NAT routers.
Likely, both non-Cisco NAT routers have no support for SIP ALG, or supports it incorrectly.
So as explained above,. "IP communication" is not enough. There is no "other cause".
01-09-2012 03:18 AM
Good, thank you for the nice rating and good luck!
01-08-2012 06:46 AM
As requested before, you need to include "debug ccsip message" taken with "term mon".
01-08-2012 06:46 AM
this is what i have with debug ccsip message
001481: Jan 8 14:36:02.574: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Reason: Q.850;cause=102
Date: Sun, 08 Jan 2012 14:36:02 GMT
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
From:
Allow-Events: telephone-event
Supported: replaces
Remote-Party-ID: "808" <808>;party=called;screen=no;privacy=off808>
Content-Length: 0
To: <808>;tag=12CF7C4-21E3808>
Contact: <808>808>
Call-ID: 01B28F943B81400000005C62@pbxgw.default.com
Via: SIP/2.0/UDP 192.168.5.254:5060;branch=z9hG4bK9EF45718FDD90989C081A9D4333EA589;received=41.190.234.172
CSeq: 1 INVITE
Server: Cisco-SIPGateway/IOS-12.x
01-08-2012 07:05 AM
You should include the complete trace, as the above is not complete.
However, is seem that the GSM box is on a public IP, while the UC500 is on a private IP behind a NAT and/or firewall router, likely not a Cisco device..
In that case, it's normal that doesn't work. You need to have them both on the same local network.
Otherwise, you will have major problems to make it work.
01-08-2012 07:20 AM
the private ip address is my GSM gateway and the public is my UC540
there is ip communication between the 2
will love to have more causes on this
01-08-2012 07:27 AM
Actually, the trace shows that both devices have private address, and are behind two different NAT routers.
Likely, both non-Cisco NAT routers have no support for SIP ALG, or supports it incorrectly.
So as explained above,. "IP communication" is not enough. There is no "other cause".
01-09-2012 01:43 AM
great is ok now
the problem was the NAT given that i have a inter device that Nat
Thanks
01-09-2012 03:18 AM
Good, thank you for the nice rating and good luck!
07-31-2012 12:21 AM
Hello All
I have a uc540, that users can, make outgoing calls to gsm, but cant get a return voice
Here is the log i have gathered by doing show log
000280: Jul 30 16:44:40.729: //101/E9A6F0698127/SIP/Info/sipSPICheckResponseExt: INVITE response with no RSEQ - disable IS_REL1XX
000281: Jul 30 16:44:40.729: //101/E9A6F0698127/SIP/Error/sipSPICheckReliableProvStringtag: Unable to access supported header values
000282: Jul 30 16:44:40.729: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentGTD: No GTD found in inbound container
000283: Jul 30 16:44:40.729: //101/E9A6F0698127/SIP/Info/sipSPIDoMediaNegotiation: Number of m-lines = 1
SIP: Attribute mid, level 1 instance 1 not found.
000284: Jul 30 16:44:40.729: //101/E9A6F0698127/SIP/Info/resolve_media_ip_address_to_bind: Media already bound, use existing source_media_ip_addr
000285: Jul 30 16:44:40.729: //101/E9A6F0698127/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 192.168.30.1
000286: Jul 30 16:44:40.729: //101/E9A6F0698127/SIP/Info/sipSPIDoAudioNegotiation: Codec (g711ulaw) Negotiation Successful on Static Payload for m-line 1
000287: Jul 30 16:44:40.729: //101/E9A6F0698127/SIP/Info/sipSPIDoPtimeNegotiation: One ptime attribute found - value:20
000288: Jul 30 16:44:40.729: //-1/xxxxxxxxxxxx/SIP/Info/convert_ptime_to_codec_bytes: Values :Codec: g711ulaw ptime :20, codecbytes: 160
000289: Jul 30 16:44:40.729: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711ulaw codecbytes :160, ptime: 20
000290: Jul 30 16:44:40.729: //101/E9A6F0698127/SIP/Media/sipSPIDoPtimeNegotiation: Offered ptime:20, Negotiated ptime:20 Negotiated codec bytes: 160 for codec g711ulaw
000291: Jul 30 16:44:40.729: //101/E9A6F0698127/SIP/Info/sipSPIDoDTMFRelayNegotiation: m-line index 1
000292: Jul 30 16:44:40.729: //101/E9A6F0698127/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(101) could not be reserved.
000293: Jul 30 16:44:40.729: //101/E9A6F0698127/SIP/Info/sipSPIDoDTMFRelayNegotiation: RTP-NTE DTMF relay option
000294: Jul 30 16:44:40.729: //101/E9A6F0698127/SIP/Info/sipSPIDoDTMFRelayNegotiation: Case of full named event(NE) match in fmtp list of events.
000295: Jul 30 16:44:40.729: //-1/xxxxxxxxxxxx/SIP/Info/sip_sdp_get_modem_relay_cap_params: NSE payload from X-cap = 0
000296: Jul 30 16:44:40.729: //101/E9A6F0698127/SIP/Info/sip_select_modem_relay_params: X-tmr not present in SDP. Disable modem relay
000297: Jul 30 16:44:40.729: //101/E9A6F0698127/SIP/Info/sipSPIGetSDPDirectionAttribute: No direction attribute present or multiple direction attributes that can't be handled for m-line:1 and num-a-lines:0
000298: Jul 30 16:44:40.729: //101/E9A6F0698127/SIP/Info/sipSPIDoAudioNegotiation: Codec negotiation successful for media line 1
payload_type=0, codec_bytes=160, codec=g711ulaw, dtmf_relay=rtp-nte
stream_type=voice+dtmf (1), dest_ip_address=41.190.224.227, dest_port=13000
000299: Jul 30 16:44:40.729: //101/E9A6F0698127/SIP/State/sipSPIChangeStreamState: Stream (callid = -1) State changed from (STREAM_DEAD) to (STREAM_ADDING)
000300: Jul 30 16:44:40.733: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISipSdpFree:
000301: Jul 30 16:44:40.733: //101/E9A6F0698127/SIP/Media/sipSPIUpdCallWithSdpInfo:
Preferred Codec : g711ulaw, bytes :160
Preferred DTMF relay : rtp-nte
Preferred NTE payload : 101
Early Media : No
Delayed Media : No
Bridge Done : No
New Media : No
DSP DNLD Reqd : No
000302: Jul 30 16:44:40.733: //101/E9A6F0698127/SIP/Info/resolve_media_ip_address_to_bind: Media already bound, use existing source_media_ip_addr
000303: Jul 30 16:44:40.733: //101/E9A6F0698127/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 192.168.30.1
000304: Jul 30 16:44:40.733: //101/E9A6F0698127/SIP/Media/sipSPIUpdCallWithSdpInfo:
Stream type : voice+dtmf
Media line : 1
State : STREAM_ADDING (2)
Stream address type : 1
Callid : 101
Negotiated Codec : g711ulaw, bytes :160
Nego. Codec payload : 0 (tx), 0 (rx)
Negotiated DTMF relay : rtp-nte
Negotiated NTE payload : 101 (tx), 101 (rx)
Negotiated CN payload : 0
Media Srce Addr/Port : [192.168.30.1]:18806
Media Dest Addr/Port : [41.190.224.227]:13000
000305: Jul 30 16:44:40.733: //101/E9A6F0698127/SIP/Info/sipSPIProcessHistoryInfoHeader: No HI headers recvd from app container
000306: Jul 30 16:44:40.733: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentQSIG: No QSIG Body found in inbound container
000307: Jul 30 16:44:40.733: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentQ931: No RawMsg Body found in inbound container
000308: Jul 30 16:44:40.733: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICreateNewRawMsg: No Data to form The Raw Message
000309: Jul 30 16:44:40.733: //101/E9A6F0698127/SIP/Info/HandleSIP1xxSessionProgress: ccsip_api_call_cut_progress returned: SIP_SUCCESS
000310: Jul 30 16:44:40.733: //101/E9A6F0698127/SIP/State/sipSPIChangeState: 0x8786E308 : State change from (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING) to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)
000311: Jul 30 16:44:40.733: //101/E9A6F0698127/SIP/Info/HandleSIP1xxSessionProgress: Transaction Complete. Lock on Facilities released.
000312: Jul 30 16:44:40.733: //101/E9A6F0698127/SIP/Info/ccsip_bridge: confID = 21, srcCallID = 101, dstCallID = 100
000313: Jul 30 16:44:40.733: //101/E9A6F0698127/SIP/Info/sipSPIUupdateCcCallIds: Old src/dest ccCallids: -1/-1, new src/dest ccCallids: 101/100
000314: Jul 30 16:44:40.733: //101/E9A6F0698127/SIP/Info/sipSPIUupdateCcCallIds: Old streamcallid=101, new streamcallid=101
000315: Jul 30 16:44:40.733: //101/E9A6F0698127/SIP/Info/ccsip_gw_set_sipspi_mode: Setting SPI mode to SIP-TDM
000316: Jul 30 16:44:40.733: //101/E9A6F0698127/SIP/Info/ccsip_bridge: xcoder_attached = 0, xmitFunc = -2141347968, ccb xmitFunc = -2141347968
000317: Jul 30 16:44:40.733: //101/E9A6F0698127/SIP/Media/sipSPIProcessRtpSessions: sipSPIProcessRtpSessions
000318: Jul 30 16:44:40.733: //101/E9A6F0698127/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 101) to the VOIP RTP library
000319: Jul 30 16:44:40.733: //101/E9A6F0698127/SIP/Info/resolve_media_ip_address_to_bind: Media already bound, use existing source_media_ip_addr
000320: Jul 30 16:44:40.733: //101/E9A6F0698127/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 192.168.30.1
000321: Jul 30 16:44:40.733: //101/E9A6F0698127/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1
000322: Jul 30 16:44:40.733: //101/E9A6F0698127/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info
laddr = 192.168.30.1, lport = 18806, raddr = 41.190.224.227, rport=13000, do_rtcp=TRUE
src_callid = 101, dest_callid = 100, stream type = voice+dtmf, stream direction = SENDRECV
media_ip_addr = 41.190.224.227, vrf tableid = 0 media_addr_type = 1
000323: Jul 30 16:44:40.733: //101/E9A6F0698127/SIP/Media/sipSPIUpdateRtcpSession: RTP session already created - update
000324: Jul 30 16:44:40.733: //101/E9A6F0698127/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:88027A8C
000325: Jul 30 16:44:40.737: //101/E9A6F0698127/SIP/Info/sipSPIUpdateRtcpSession:
DTMF inb/oob iwf enabled 0
000326: Jul 30 16:44:40.737: //101/E9A6F0698127/SIP/Media/sipSPIGetNewLocalMediaDirection:
New Remote Media Direction = SENDRECV
Present Local Media Direction = SENDRECV
New Local Media Direction = SENDRECV
retVal = 0
000327: Jul 30 16:44:40.737: //101/E9A6F0698127/SIP/State/sipSPIChangeStreamState: Stream (callid = 101) State changed from (STREAM_ADDING) to (STREAM_ACTIVE)
000328: Jul 30 16:44:40.737: //101/E9A6F0698127/SIP/Info/ccsip_bridge:
DTMF inb/oob iwf enabled 0
000329: Jul 30 16:44:40.737: //101/E9A6F0698127/SIP/Info/ccsip_caps_ind: Entry
000330: Jul 30 16:44:40.737: //101/E9A6F0698127/SIP/Info/ccsip_get_rtcp_session_parameters: CURRENT VALUES: stream_callid=101, current_seq_num=0x178C
000331: Jul 30 16:44:40.737: //101/E9A6F0698127/SIP/Info/ccsip_get_rtcp_session_parameters: NEW VALUES: stream_callid=101, current_seq_num=0x1255
000332: Jul 30 16:44:40.737: //101/E9A6F0698127/SIP/Info/ccsip_caps_ind: Load DSP with negotiated codec: g711ulaw, Bytes=160
000333: Jul 30 16:44:40.737: //101/E9A6F0698127/SIP/Info/ccsip_caps_ind: Set forking flag to 0x0
000334: Jul 30 16:44:40.737: //101/E9A6F0698127/SIP/Info/sipSPISetDTMFRelayMode: Set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_NTE_AND_OOB with rx payload = 101, tx payload = 101
000335: Jul 30 16:44:40.737: //101/E9A6F0698127/SIP/Info/sip_set_modem_caps: Preferred (or the one that came from DSM) modem relay=0, from CLI config=0
000336: Jul 30 16:44:40.737: //101/E9A6F0698127/SIP/Info/sip_set_modem_caps: Disabling Modem Relay...
000337: Jul 30 16:44:40.737: //101/E9A6F0698127/SIP/Info/sip_set_modem_caps: Negotiation already Done. Set negotiated Modem caps and generate SDP Xcap list
000338: Jul 30 16:44:40.737: //101/E9A6F0698127/SIP/Info/sip_set_modem_caps: Modem Relay & Passthru both disabled
000339: Jul 30 16:44:40.737: //101/E9A6F0698127/SIP/Info/sip_set_modem_caps: nse payload = 0, ptru mode = 0, ptru-codec=0, redundancy=0, xid=0, relay=0, sprt-retry=12, latecncy=200, compres-dir=3, dict=1024, strnlen=32
000340: Jul 30 16:44:40.737: //101/E9A6F0698127/SIP/Media/sipSPISetStreamInfo: 1 Active Streams
000341: Jul 30 16:44:40.737: //101/E9A6F0698127/SIP/Media/sipSPISetStreamInfo: Adding stream type (voice+dtmf) from media
line 1 codec g711ulaw
000342: Jul 30 16:44:40.737: //101/E9A6F0698127/SIP/Media/sipSPISetStreamInfo:
caps.stream_count=1,caps.stream[0].stream_type=0x3, caps.stream_list.xmitFunc=
000343: Jul 30 16:44:40.737: //101/E9A6F0698127/SIP/Media/sipSPISetStreamInfo: voip_rtp_xmit, caps.stream_list.context=
000344: Jul 30 16:44:40.737: //101/E9A6F0698127/SIP/Media/sipSPISetStreamInfo: 0x8A3D3F00 (gccb)
000345: Jul 30 16:44:40.737: //101/E9A6F0698127/SIP/Info/ccsip_caps_ind: Load DSP with codec : g711ulaw, Bytes=160, payload = 0
000346: Jul 30 16:44:40.737: //101/E9A6F0698127/SIP/Info/ccsip_caps_ind: ccsip_caps_ind: ccb->pld.flags_ipip = 0x2201
000347: Jul 30 16:44:40.737: //101/E9A6F0698127/SIP/Info/sipSPISrtpTranscoder:
Checking if transcoder is needed for SRTP-RTP
000348: Jul 30 16:44:40.737: //101/E9A6F0698127/SIP/Info/ccsip_caps_ind: Calling cc_api_caps_ack()
000349: Jul 30 16:44:40.737: //101/E9A6F0698127/SIP/Info/ccsip_caps_ack: Set forking flag to 0x0
000350: Jul 30 16:44:43.529: //101/E9A6F0698127/SIP/Info/ccsip_indicate_rt_packet_stats: Processing stats for callid=101, proc_id=9
000351: Jul 30 16:44:46.357: //101/E9A6F0698127/SIP/Info/ccsip_indicate_rt_packet_stats: Processing stats for callid=101, proc_id=9
000352: Jul 30 16:44:52.321: //101/E9A6F0698127/SIP/Info/ccsip_indicate_rt_packet_stats: Processing stats for callid=101, proc_id=9
000353: Jul 30 16:44:56.605: //101/E9A6F0698127/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:88027A8C
000354: Jul 30 16:44:56.605: //101/E9A6F0698127/SIP/Info/ccsip_call_statistics: Requesting stats for callid=101
000355: Jul 30 16:44:56.605: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
000356: Jul 30 16:44:56.609: //101/E9A6F0698127/SIP/Info/ccsip_indicate_rt_packet_stats: Processing stats for callid=101, proc_id=1
000357: Jul 30 16:44:56.609: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 7
000358: Jul 30 16:44:56.609: //101/E9A6F0698127/SIP/Info/sipSPISendCancel: Associated container=0x885EFD7C to Cancel
000359: Jul 30 16:44:56.609: //101/E9A6F0698127/SIP/Transport/sipSPISendCancel: Sending CANCEL to the transport layer
000360: Jul 30 16:44:56.609: //101/E9A6F0698127/SIP/Transport/sipSPITransportSendMessage: msg=0x88027468, addr=41.190.224.226, port=5060, sentBy_port=0, is_req=1, transport=1, switch=0, callBack=0x80EAE1FC
000361: Jul 30 16:44:56.609: //101/E9A6F0698127/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
000362: Jul 30 16:44:56.609: //101/E9A6F0698127/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
000363: Jul 30 16:44:56.609: //101/E9A6F0698127/SIP/Transport/sipTransportLogicSendMsg: Set to send the msg=0x88027468
07-31-2012 02:09 AM
That was answered before already, wasn't ?
07-31-2012 02:14 AM
Just a new problem with another UC540 of a customer
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