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Hi preface: I will try to setup CUBE to work with SIP ITSP CUCM_6.1(old) ->CUBE(sip-to-sip)->ITSP I doesn't want to use MTP required on SIP trunk !!! ITSP requires EO ( INVITE w/SDP ) . ITSP doesn't answer with 100/Trying in INVITE ...
Hi!Our client said when he reloads the subscriber he cannot get access to CUCM publisher web access for a 5-10 min.After sub reboots pub web admin becomes accessible again.He said that issue can be reproduced.This is not typical problem. I look thru ...
Hi!Is it possible to debug DTMF event ( when gataway receive/transmit DTMF tone over ISDN PRI ) on 28XX/29XX/5350XM gateways?On analog FXO ports there is a 'debug vpm signal' commandI looking similar coommand for PRI tone debugging .../lexxa
HelloI've got 871 routerThe was c870-advsecurityk9-mz.123-8.YI2.bin inside. After I upgraded IOS to c870-advsecurityk9-mz.124-9.T7.bin and reload, 871 dorp to ROMMON with strange error_________________________________________________________________...
HelloThere 3 types of MTP1) CCM software ( Media Streaming App)2) IOS software3) IOS hardwareI have some clients who use SIP trunk (w req. MTP) from CUCM to Service Provider and MGCP controlled FXS portshint: Service provider uses NON CISCO equpment...
I didn't try to do 'request reiNVITE sdp-header session-info add' because the detination/call controlled by one dial-peer and sip profile 1) INVITE w/SDP2) REINVITE w/SDP ( fixed by sip profile) - HOLD3) REINVITE without/SDP ( fixed by sip profile) ...
Hi Ayodeji When call puts on hold CUCM sends REINVITE with IP=0.0.0.0 & a=inactive which can be fixed by "voice-class sip profile"When call unhold CUCM sends REINVITE without SDP and CUBE also sends REINVITE without SDP to ITSP this is the key I t...
Hi, Kirill Thanks for you reply>> What problem do u have with this DO ReInvite? Yes, call doesn't come back . Because ITSP doesn't reply with 100/Trying. So CUBE retrys to send REINVITEs and after timer expires CUCM send BYE. The real question why t...
Thanks for right direction debug vtsp dsp: seems like working on analog portsbut debug voip vtsp session ( on 5350XM) shows digits during mid-callJun 6 13:56:50.521: //5004160/xxxxxxxxxxxx/VTSP:(3/1:D):30:0:0/vtsp_dsm_digit_begin_cb: Be...
Hi aokanlawon, thanks for you replybut 'debug voip rtp session named-event' shows that gateway send/recieve dtmf event on VOIP call leg as NTE event ( when configured dtmf relay rtp-nte )I need a debug on POTS leg, when gataway recieve a tone in B-c...