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UC540

divine007
Level 1
Level 1

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

3 Accepted Solutions

Accepted Solutions

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.

View solution in original post

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".

View solution in original post

Good, thank you for the nice rating and good luck!

View solution in original post

10 Replies 10

paolo bevilacqua
Hall of Fame
Hall of Fame

As requested before, you need to include "debug ccsip message" taken with "term mon".

divine007
Level 1
Level 1

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: ;vnd.pimg.port=29;tag=252232463135364104081DF7

Allow-Events: telephone-event

Supported: replaces

Remote-Party-ID: "808" <808>;party=called;screen=no;privacy=off

Content-Length: 0

To: <808>;tag=12CF7C4-21E3

Contact: <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

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.

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

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".

great is ok now

the problem was the NAT given that i have a inter device that Nat

Thanks

Good, thank you for the nice rating and good luck!

divine007
Level 1
Level 1

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

That was answered before already, wasn't ?

Just a new problem with another UC540 of a customer

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