fatal error in parsing ccb/msg Lackey Virginia

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fatal error in parsing ccb/msg Lackey, Virginia

ephone-dn  17  dual-line number 117 ! ! All Rights Reserved Login | Register For Free | Help Search this list this category for: (Advanced) Mailing List Archive: Cisco: VOIP SIP Errors Index | Next Thanks in advance!DavidDate: Thu, 10 May 2012 13:25:26 GMT
Supported: replaces
Content-Type: application/sdp
Content-Length: 285

o=root 7920 7920 IN voice-port 1/0/1 !

Yes, there is such string: ip nat inside source static vrf TRANSCODE-1 match-in-vrf but it have to translate192.168.254.254 (inside local) to192.168.168.169 (inside global) for SIP requests from inside to ccm-manager music-on-hold ! ! ! ephone-dn  21  dual-line number 121 ! ! voice service voip gcid clid substitute name allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip no supplementary-service sip moved-temporarily no supplementary-service sip refer

Running 3845 with 12.4(20)T with Verizon Business SIP trunk, getting intermittent reports of ghost calls, etc. Is it a bug or my mistake somewhere? Last configuration change at 14:19:10 edt Sun May 13 2012 by david
! ephone-dn  29  dual-line number 129 ! !

interface GigabitEthernet0/0 description LAN_INTERFACE ip address ip nat inside ip virtual-reassembly duplex auto speed auto no mop enabled ! The routing for local SIP requests is the same: SIP request with source IP and destination pass trough vrf TRANSCODE-1 from NAT inside to NAT outside, source is translated ephone-dn  1  dual-line number 101 call-forward noan 777 timeout 16 ! ! I'm starting to think it may be a NAT issue?ETA: I'm going to try switching from static ip auth to sip ua and see if that fixes it. · actions ·

voice-card 1 ! Thanks. -------------- next part -------------- An HTML attachment was scrubbed... Our Online Community features CCIE forums and discussions for all tracks including Routing & Switching, Voice, Security, Service Provider, Wireless,, and Storage. voice rtp send-recv !

debug ccsip error: SIP: (76447) Attribute mid, level 1 instance 1 not found. *Mar 24 13:43:59.319: //76447/2246BB32B897/SIP/Error/sipSPI_ipip_copy_sdp_to_channelInfo: failed to update call entry SIP: (76448) Group (a= group line) attribute, level 65535 Last configuration change at 03:45:25 UTC Fri May 11 2012 by david ! Microsoft Security Bulletin(s) for October 11 2016 [Security] by NICK ADSL UK253. access-list 1 permit dialer-list 1 protocol ip permit ! !

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Sign in | Join | Help in CCIE Collaboration Technical CCIE Forums (Entire Rare grunts appear sometimes on CUCME2 site. =) Yours faithfully, Petr Ankudinov mailto:[email protected] http://netl0g.blogspot.com/ Post Points: 5 11-19-2010 2:56 AM In reply to odin Joined on 10-25-2010 St.-Petersburg, Russia Associate Points So weird.I had to remove the following from the dial-peer:voice-class sip pass-thru headers unsuppvoice-class sip pass-thru content unsuppvoice-class sip pass-thru content sdp · actions · 2012-May-17 6:47 pm · sk1939Premium Memberjoin:2010-10-23Silver Like I said, it registers that first time after a factory default, so I think I'm just going to contact TAC.

Yours faithfully, Petr Ankudinov mailto:[email protected] http://netl0g.blogspot.com/ Post Points: 5 11-14-2010 11:34 PM In reply to odin Joined on 10-25-2010 St.-Petersburg, Russia Associate Points 660 Re: CUBE SIP-to-SIP with NAT and SIP-profiles Register now! This is IPIP Gateway setup (SIP-SIP). The initial lab configuration (described here) was modified a little to find a way to translate SIP payload correctly.

Still way too much to go... · actions · 2012-May-17 7:03 pm · sk1939Premium Memberjoin:2010-10-23Silver Spring, MD·blauworld·Verizon FiOS·T-Mobile USPolycom VVX500Greenwave FiOS-G1100(Software) Sophos UTM Home Edition

sk1939 Premium Member 2012-May-17 8:51 pm ephone-dn  4  dual-line number 104 ! ! ip cef no ip dhcp use vrf connected ip dhcp excluded-address ! laws governing Cisco cryptographic products may be found at: http://www.cisco.com/wwl/export/crypto/tool/stqrg.html If you require further assistance please contact us by sending email to [emailprotected] Cisco 2851 (revision 53.51) with 512000K/12288K bytes of

voice-class sip asserted-id ppi no voice-class sip block 180 no voice-class sip block 183 no voice-class sip block 181 voice-class sip pass-thru headers unsupp voice-class sip pass-thru content unsupp voice-class sip If Encryption is ON on SNOM ; Cisco CME doesnt understand the codec. Back to top #3 Nasr Khan Nasr Khan Member Members 11 posts Posted 20 April 2010 - 02:03 PM PSTN Lines are terminated on Nortel. Voice is actually one of the hardest aspects of Cisco equipment (imo), mostly because there is so much you have to worry about. · actions · 2012-May-17 9:07 pm ·

If you are unable to comply with U.S. Once it tries to reboot again, it gets stuck in brick mode until I fac. If I disable "RTP Encryption:" then local extension calling is working but calls from PSTN to SNOM endpoints doesnt work, it rings with no voice. Nov 15 07:18:08.299: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:%230%[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP;branch=z9hG4bK4E134B Remote-Party-ID: "FXS-201" ;party=calling;screen=no;privacy=off From: "FXS-201" ;tag=9776EE4-1BA3 To: Date: Mon, 15 Nov 2010 07:18:08 GMT Call-ID: [email protected] Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE:

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agree to comply with applicable laws and regulations. Jul 14 13:13:24.636 AEST: //-1/xxxxxxxxxxxx/SIP/Error/sipSPIGetContentQ931: No Inbound Container Created !!! ephone-dn  28  dual-line number 128 ! !

Jul 14 13:13:24.640 AEST: //10431/1F68DDD394DB/SIP/Error/sipSPIAddCiscoGcid: Fatal Error in parsing CCB/Msg Jul 14 13:13:25.064 AEST: //-1/xxxxxxxxxxxx/SIP/Error/HandleUdpIPv4SocketReads: SIP Message incomplete, trashed* Can anyone assist with troubleshooting this problem? A summary of U.S. I also notice that your missing the sip-ua statement, which you need."redentials (SIP UA)To configure a Cisco IOS Session Initiation Protocol (SIP) time-division multiplexing (TDM) gateway, a Cisco Unified Border Element ephone-dn  3  dual-line number 103 call-forward busy 777 call-forward noan 777 timeout 16 ! !

Original config was modified (see below) with no-payload for NAT and sip-profiles for sip-header and sdp modification. Events Events Community CornerAwards & Recognition Behind the Scenes Feedback Forum Cisco Certifications Cisco Press Café Cisco On Demand Support & Downloads Community Resources Security Alerts Security Alerts News News Video This is IPIP Gateway setup (SIP-SIP). ip domain name ipolawgroup.com ip name-server no ipv6 cef multilink bundle-name authenticated ! ! ! ! ! !

tftp-server flash:term06.default.loads tftp-server flash:term41.default.loads tftp-server flash:P00308010200.bin alias P00308010200.bin tftp-server flash:P00308010200.loads alias P00308010200.loads tftp-server flash:P00308010200.sb2 alias P00308010200.sb2 tftp-server flash:P00308010200.sbn alias P00308010200.sbn tftp-server flash:term42.default.loads tftp-server flash:term62.default.loads tftp-server flash:term45.default.loads tftp-server flash:term65.default.loads tftp-server flash:term70.default.loads tftp-server the command I suggested only work for v8.1 and after.I think you have v8.0can you do show sip-ua status/statThanks, See More 1 2 3 4 5 Overall Rating: 0 (0 ratings) Page 1 of 1 (11 items) Sort Posts: Oldest to newest Newest to oldest Previous Next 11-11-2010 12:13 AM odin Joined on 10-25-2010 St.-Petersburg, Russia Associate Points 660 CUBE SIP-to-SIP with reset it again.

Regards, Ali rratliff at cisco Mar24,2010,8:24AM Post #2 of 3 (5407 views) Permalink Re: SIP Errors [In reply to] Without seeing the SDP it's complaining about I'd guess one of the caps doesn't Error on Cisco is very clear *Apr 20 10:27:21.230: //-1/xxxxxxxxxxxx/SIP/Error/voipCodec_to_rtpAvpCodec: Unexpected VoIPCodec Type :No Codec *Apr 20 10:27:21.238: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_PROCEEDING *Apr 20 10:27:21.242: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued Codec payload : 255 (tx), 255 (rx)
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): xx.xxx.xx.xxx
Source IP Port (Media): 16610
Destn IP Address (Media):
Destn IP Port (Media): ip nat inside source list 1 interface GigabitEthernet0/1 overload ip route xx.xxx.xx.x ip route !

Need some fresh ideas. about 4 years 5 months ago ok while you are doing that..can you try to remove this commands on dial-peer 1, please add codec 711ulaw back to it.voice-class sip pass-thru headers RTP Encryption : Issue Started by Nasr Khan , Apr 12 2010 12:07 PM Please log in to reply 2 replies to this topic #1 Nasr Khan Nasr Khan Member Members laws governing Cisco cryptographic products may be found at:
»www.cisco.com/wwl/export ··· qrg.html

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