Hello. I see a very strange behaviour when I try to send my calls out to PSTN E1. My call flow is CUCM --sip trunk-- GW1 --sip -- GW2 --E1 -- PSTN. I clearly see that call reached gateway2 and than routed to E1, sometimes I can see 1 second ringing on the destination phone, but then call is CANCELLED  by GW2 or Pstn (Do not know exactly where it cancelled but maximum wait time is 4 seconds). I see this in debug isdn 002653: *Sep 16 12:47:41.215: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8  callref = 0x00E8         Cause i = 0x80E6 - Recovery on timer expiry 002654: *Sep 16 12:47:41.275: ISDN Se0/0/0:15 Q931: RX <- RELEASE_COMP pd = 8  callref = 0x80E8         Cause i = 0x87E4 - Invalid information element contents          My best suggestion is T310 timer, but I have already increased it on CUCM/GW2  to 15 and 60 seconds and I see no difference. Do not know  what 'Invalid information element contents' error means. Please help me to point out this problem. I have included partial show run and full debug isdn q931. Thank you ... View more

Hello, guys.   Sorry if this topic is a duplicate, but I really haven't found clear solution for my question.   I want to setup load balancing to UC application pairs. Including CUIC, CUCM, Finesse, CUP, CUC.   I have set Haproxy solution with pass-through mode. It works fine but I need to accept server ssl certificate every new session. To avoid this I think I have to implement SSL termination on my load balancer. To do that I believe I have to import certificates/private key pairs to load balancer.   I have found the way to export certificate from Cisco Unified OS Administration > Security > Certificate management, but how can I get a private key? Is it possible without breaking the OS?   Maybe you can suggest other possible ways to implement load balancing and reliability between these applications. Any help appreciated. ... View more

Hi. We have some calls disconnected during the conversation. These failures happen random on different phones and users several times a day. The phones are SIP Ip communicators. We have CUCM cluster 10.13.81.6(pub) 10.13.81.7(sub1 - the server was shut down) 10.13.81.16(sub2 - the only call-processing node ) I found this error messages at the log file(call from 934760460 to 8267 at 15:17 ) Thu Feb 12 15:17:10.447 :   DET : (  4984) -VM| main|cip.app.bv:? - reservationGranted - render ( "From 934760460" ) Thu Feb 12 15:17:10.447 :    EE : (  4984) main|cip.app.bv:? - reservationGranted - render ( "From 934760460" ) Thu Feb 12 15:17:12.467 :    EE : (  3308) -SFB| --> CSoftFireBoxWnd::onKeyUp() Thu Feb 12 15:17:12.467 :  SPCL : (  3308) -SFB|  &S CSoftFireBoxWnd::onKeyUp(): wParam_ = <44>, lParam_ = <0xC1370001> Thu Feb 12 15:17:12.467 :    EE : (  3308) -SFB| <-- CSoftFireBoxWnd::onKeyUp() Thu Feb 12 15:17:21.697 : ERROR : (  4116) SIP : sip_tcp_create_connection : socket connect failed errno: 34 ipaddr: 10.13.81.7, port: 5060 Thu Feb 12 15:17:21.697 : ERROR : (  4116) SIP : sip_tcp_detach_socket : Max TCP connections reached. Thu Feb 12 15:17:21.697 :   DET : (  4116) SIPCC-SIP_TCP_MSG: sip_tcp_purge_entry: Socket fd: 1396 closed for connid 1 with address: 1, remote port: 168644871 Thu Feb 12 15:17:21.697 : ERROR : (  4116) sip_regmgr_ev_tmr_ack_retry: Error: sip_platform_tcp_channel_create(server addr=10.13.81.7, server port=5060) failed. Thu Feb 12 15:17:21.697 :   DET : (  4116) SIPCC-SIP_FALLBACK: 60/1, sip_regmgr_retry_timer_start: Starting fallback timer (120 sec) Does CUCM have tcp connections limit? is it configurable? Or maybe it is windows machine's error? Attached full log ... View more

Hi all. I see following SIP options call-flow for sip options ping: 17907581: Jan 16 08:11:31.057: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: OPTIONS sip:10.250.20.25:5060 SIP/2.0 Via: SIP/2.0/UDP 10.11.3.17:9256;branch=z9hG4bKabuf3p5e3es57i595u9viaf77 Call-ID: c7vb43afudub9c4f944edcf4a937p4s3@SBC.com From: <sip:SBC@10.11.3.17>;tag=sbc0800uf7upc43 To: <sip:10.250.20.25> CSeq: 1 OPTIONS Max-Forwards: 70 Content-Length: 0 17907583: Jan 16 08:11:31.057: //28111388/1CCF073F9ED3/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.11.3.17:9256;branch=z9hG4bKabuf3p5e3es57i595u9viaf77 From: <sip:SBC@10.11.3.17>;tag=sbc0800uf7upc43 To: <sip:10.250.20.25>;tag=BC08FBDC-1F16 Date: Fri, 16 Jan 2015 08:11:31 GMT Call-ID: c7vb43afudub9c4f944edcf4a937p4s3@SBC.com Server: Cisco-SIPGateway/IOS-15.4.20141104.060737. CSeq: 1 OPTIONS Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Accept: application/sdp Supported: timer,resource-priority,replaces,sdp-anat Content-Type: application/sdp Content-Length: 363 v=0 o=CiscoSystemsSIP-GW-UserAgent 1616 9170 IN IP4 10.13.4.43 s=SIP Call c=IN IP4 10.13.4.43 t=0 0 m=audio 0 RTP/AVP 18 0 8 9 4 2 15 c=IN IP4 10.13.4.43 m=image 0 udptl t38 c=IN IP4 10.13.4.43 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:320 a=T38FaxUdpEC:t38UDPRedundancy   I know that this is normal behavior for voice gateway(3925 in our case). Is it possible to disable SDP in 200 OK message going out to ITSP. They said that their PBX can't recognize 200 ok with SDP as reply to sip Options message. ... View more