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This is IPIP Gateway setup (SIP-SIP). Not SIP-SIP or SET_MODE is not done.SIP: (672) Attribute mid, level 1 instance 1 not found.Nov 24 22:12:25.031: //672/DC45A4DD87BA/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:failed to update call entryNov 24 22:12:25.031: //672/DC45A4DD87BA/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:failed to update call entryNov 24 Top Profile Reply with quote Display posts from previous: All posts1 day7 days2 weeks1 month3 months6 months1 yearSort by AuthorPost timeSubject AscendingDescending Post new topic Reply to topic Page Next message: [cisco-voip] One-way audio Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] Running 3845 with 12.4(20)T with Verizon Business SIP trunk, getting intermittent http://qaisoftware.com/failed-to/failed-to-fix-dc-dns-entry.html

Audiocodes gw are affected from this problem?The log from vg is in attachment.ThanksPost by GmbHi,a sipx v.3.11.12-015003 in HA configuration connected using a SIPTrunk with a Cisco voice gatewayused for connecting Na chwilę obecną pamiętam że: Idzie SIP/2.0 487 Request Cancelled z Cisco do Proxy. Czy sprawdzałeś czy DISCONNECT dla calla wystawiany jest do ISDN przez ASa czy przychodzi z PSTN? As I'm not using NAT, everything on the same Subnet.

Channel Count Is Not Set At This Point. Not Sip-sip Or Set_mode Is Not Done.

Try it for free! Open the Windows Start Menu, type firefox and right-click on the "Mozilla Firefox" entry that appears. I preferred it when everything was bound to an internal loopback but this does resolve the issue. i think that call is incorrectly routed. 172.16.10.2 is interface of CME.

I'm complete newbie in VoIP network, but my goal is to learn more. mozilla Ask a question Sign In English Search Home Firefox Fix slowness, crashing, error ... Select Run as Administrator and Continue if the User Account Control dialog comes up. Mar 23 04:57:52.916: //1796/0D90B5378366/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo: failed to update call entry Mar 23 04:57:52.916: //1796/0D90B5378366/SIP/Error/sipSPI_ipip_update_call_entry: failed to update call entry Mar 23 04:57:52.916: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_ipip_set_channel_count: Unable to set CHANNEL_COUNT for callid 1796 Mar 23

duplex auto speed auto!ip forward-protocol ndip http serverip http authentication localip http secure-server!! !!!!!!tftp-server flash:spa50x_30x_cz_v753.xml alias spa50x_30x_cz_v753.xmltftp-server flash:/cs-be-sccp.jar alias user_define_1/be-sccp.jartftp-server flash:/649EF376D356.xml alias 210.xmltftp-server flash:211.xml!control-plane!call threshold global Group (a= Group Line) Attribute, Level 65535 Instance 1 Not Found Contact Us Privacy Policy Legal Notices Report Trademark Abuse Source Code Twitter Facebook Firefox Friends Switch to mobile site [email protected] Discussion: Problem with Directed call pickup and Cisco Vg (too old Jeśli nie możesz pokazać outputu z Code:debug ccsip allz ASa to chociaż pokaż całą sygnalizację połączenia z proxy, uwzględniającą porty RTP itd. - wiadomości UPDATE wykorzystywane są m.in. find more And found this post http://forums.whirlpool.net.au/archive/1718978 HTH If you like you can read up more on the command(s) here: http://www.cisco.com/en/US/docs/ios/12_3/sip/configuration/guide/chapter7.html I'm not really sure of the security implications of this to be

Not SIP-SIP or SET_MODE is not done. Codec payload : 18 (tx), 18 (rx)Negotiated Dtmf-relay : 6Dtmf-relay Payload : 101 (tx), 101 (rx)Source IP Address (Media): XXXXXXXXX1Source IP Port (Media): 19274Destn IP Cisco Support Community Directory Network Infrastructure WAN, Routing and Switching LAN, Switching and Routing Network Management Remote Access Optical Networking Getting Started with LANs IPv6 Integration and Transition EEM Scripting Other Join the community of 500,000 technology professionals and ask your questions.

  • Top Profile Reply with quote brth Post subject: Post #4 Posted: 01 Sep 2009, 21:17 Offline wannabe Joined: 14 Mar 2009, 15:42 Posts: 171 Output który pokazałeś świadczy o
  • Dial-peer now looks like this: dial-peer voice 1 voip description Outbound destination-pattern .T session protocol sipv2 session target ipv4:10.5.5.1 session transport udp no vadWhen i try call to outside number, sip
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  • Zdebugowałem sygnalizacje, wprawdzie nie na samym Cisco bo dosyć spory mam tam ruch, ale na SIP Proxy (bezpośrednie połaczenie Cisco i Sip Proxy).
  • Code:Sep 7 13:40:51.142: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpSocketReads: Msg enqueued for SPI with IP addr: XXXXXXXXX:5060Sep 7 13:40:51.142: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1Sep 7 13:40:51.142: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000Sep 7 13:40:51.142: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received:INVITE sip:[email protected]:5060 SIP/2.0Record-Route: Record-Route:
  • Poniżej zrzut INVITE (wychodzący z Proxy do Cisco) i UPDATE (wychodzący z Cisco do Proxy).
  • Issue number 2 with one way calling with nodephone was resolved to using the following: voice service voip sip bind media source-interface Dialer0 (or whatever your external interface is) end I
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  • Not SIP-SIP or SET_MODE is not done.SIP: (670) Attribute mid, level 1 instance 1 not found.Nov 24 22:12:17.027: //670/D77FB29787B0/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:failed to update call entryNov 24 22:12:17.027: //670/D77FB29787B0/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:failed to update call entryNov 24

Group (a= Group Line) Attribute, Level 65535 Instance 1 Not Found

Thanks Gmb ha scritto: Hi, i've deployed this scenario: a sipx v.3.11.12-015003 in HA configuration connected using a SIP Trunk with a Cisco voice gateway used for connecting another traditional pbx, learn this here now First exception on row 0 with id a02R000000INf53IAD; first error: SELF_REFERENCE_FROM_TRIGGER, Object (id = a02R000000INf53) is currently in trigger Decide_Which_Address_to_Use, therefore it cannot recursively update itself: [] Trigger.Update_SD_Member_Enrollment_Progress_Status: line 19, column Channel Count Is Not Set At This Point. Not Sip-sip Or Set_mode Is Not Done. If the issue is not resolved, close Firefox and restart the computer again but this time, open the program folder where Firefox is installed (e.g., open the C:\Program Files\Mozilla Firefox folder Top Profile Reply with quote valid Post subject: Post #8 Posted: 11 Sep 2009, 10:26 Offline fresh Joined: 20 Aug 2009, 14:02 Posts: 5 Location: Warszawa brth wrote:Z ostatniego

Disable modem relaySep 7 13:40:51.146: //5701830/E453EA89A7CC/SIP/Info/sipSPIGetSDPDirectionAttribute: No direction attribute present or multiple direction attributes that can't be handled for m-line:1 and num-a-lines:0Sep 7 13:40:51.146: //5701830/E453EA89A7CC/SIP/Info/sipSPIDoAudioNegotiation: Codec negotiation successful for media line this contact form Top Profile Reply with quote valid Post subject: Post #3 Posted: 01 Sep 2009, 17:06 Offline fresh Joined: 20 Aug 2009, 14:02 Posts: 5 Location: Warszawa Dzięki za info. Is there anything crucial missing in my config? In routing table, i have static route saying: Everythig going to SIP server (ip address 88.103.xxx.xx) shoud go over 10.5.5.1.

Nie zaszkodzi również jeśli pokażesz cały konfig ASa (oczywiście powycinaj dane wrażliwe). Sep 17 16:36:48.447: //366/5D90487B83AE/SIP/Error/sipSPI_ipip_copy_sdp_to_channelInfo: Unable to acquire event mask for rfc2833 dtmf relay Sep 17 16:36:48.451: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_PROCEEDING Sep 17 16:36:48.451: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from Sep 17 16:36:48.447: //366/5D90487B83AE/SIP/Error/sipSPI_ipip_copy_sdp_to_channelInfo: failed to update call entry Sep 17 16:36:48.447: //366/5D90487B83AE/SIP/Error/sipSPI_ipip_copy_sdp_to_channelInfo: Unable to find the proper instance for FMTP SIP: (366) fmtp attribute, level 1 instance 0 not found. http://qaisoftware.com/failed-to/failed-to-change-download-properties-of-call.html This behavior isn't understandable for me.

From: ;tag=280EF2AC-21D9..To: "test";tag =aeb38f16..Date: Tue, 18 Aug 2009 10:32:19 GMT..Call-ID: NzdkYWI1NzM2NTk5MGQwY2FiOWNhZmIwODgzODZmYjE...User-Agen t: Cisco-SIPGateway/IOS-12.x..Max-Forwards: 70..Route: ,,..Timestamp: 12505915 44..CSeq: 101 UPDATE..Contact: ..Remote-Party-ID: ; party=calling;screen=no;privacy=off..Content-Length: 0....Część konfiguracji:Code:sip-uano redirectionretry invite 3retry response 3retry bye Privacy Policy Support Terms of Use Login | Register For Free | Help Search this list this category for: (Advanced) Mailing List Archive: Cisco: VOIP SIP Errors If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer.

Thank you for your precious time, Standa See More 1 2 3 4 5 Overall Rating: 0 (0 ratings) Log in or register to post comments Ayodeji oladipo...

As near as I can tell all this is doing is changing the source IP address of the RTP packets from internal to external calls and internal to internal call packets Codec payload : 18 (tx), 18 (rx) Negotiated DTMF relay : rtp-nte Negotiated NTE payload : 101 (tx), 101 (rx) ccb=0x6B0CAAE4Sep 7 13:41:05.306: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received:SIP/2.0 408 Request TimeoutVia: SIP/2.0/UDP XXXXXXXXX1:5060;branch=z9hG4bK25DD224E1From: ;tag=8FBAA2F8-11D3To: "Robertoo";tag=d8fc530eCall-ID: MTE4YWZmMTI5YmMyODQ0YmY2ODBhMTRkOWY0NzMxYzA.CSeq: 101 UPDATEServer: Sip EXpress router (0.8.14 (i386/linux))Content-Length: 0Sep 7 13:41:05.306: //5701830/E453EA89A7CC/SIP/Transport/sipSPISendErrorResponse: Sending ERROR Response to the transport layerSep 7 Rozumiem że ze strony ASa?

Dzięki. Unfortunatly i've noticed that directed call pickup doesn't work when i try to pick up from a sip phones a call from traditional phones to sip phones, pick up between sip Czy jeśli w czasie tych 4 dzwonków strona B odbierze połączenie to dochodzi do skutku?Ab. Check This Out Sep 17 16:36:48.447: //366/5D90487B83AE/SIP/Error/sipSPI_ipip_copy_sdp_to_channelInfo: Unable to acquire event mask for rfc2833 dtmf relay Sep 17 16:36:48.451: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_PROCEEDING Sep 17 16:36:48.451: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from

Na numerze ujętym w debugu dzieje się to notorycznie. U abonenta B (PSTN), telefon dzwoni, ale po dokładnie 4 zwrotnych sygnałach dzwonienia połączenie jest zrywane. Internal calls are working properly but Call from outside to inside is impossible. To również Cisco terminuje połączenie w kierunku PSTN.