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</o:shapelayout></xml><![endif]--></head><body lang=IT link="#0563C1" vlink="#954F72"><div class=WordSection1><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><span lang=EN-GB>Hi All,<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB>I’ve this problem:<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB><o:p> </o:p></span></p><p style='mso-margin-top-alt:0cm;margin-right:0cm;margin-bottom:7.5pt;margin-left:0cm;line-height:15.0pt;background:white'><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri",sans-serif;mso-fareast-language:EN-US'>all SIP phone are impacted from my issue.<o:p></o:p></span></p><p style='mso-margin-top-alt:0cm;margin-right:0cm;margin-bottom:7.5pt;margin-left:0cm;line-height:15.0pt;background:white;outline: none;orphans: auto;text-align:start;widows: auto;-webkit-text-stroke-width: 0px;word-spacing:0px'><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri",sans-serif;mso-fareast-language:EN-US'> <o:p></o:p></span></p><p style='mso-margin-top-alt:0cm;margin-right:0cm;margin-bottom:7.5pt;margin-left:0cm;line-height:15.0pt;background:white;outline: none;orphans: auto;text-align:start;widows: auto;-webkit-text-stroke-width: 0px;word-spacing:0px'><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri",sans-serif;mso-fareast-language:EN-US'>So, when connection to the CUCM is broken, all SIP phone correctly register on the SRST gateway version 15.4(3)M2 at the Branch Site, but when communication with CUCM go up again no one phone reconnect to the CUCM. To force that I need to remove "voice register global" command from the router, and wait about 2 minutes phones come back to the CUCM.<o:p></o:p></span></p><p style='mso-margin-top-alt:0cm;margin-right:0cm;margin-bottom:7.5pt;margin-left:0cm;line-height:15.0pt;background:white;outline: none;orphans: auto;text-align:start;widows: auto;-webkit-text-stroke-width: 0px;word-spacing:0px'><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri",sans-serif;mso-fareast-language:EN-US'>Only SIP phone are affected from this issue. Phone used in my environment are:<o:p></o:p></span></p><p style='mso-margin-top-alt:0cm;margin-right:0cm;margin-bottom:7.5pt;margin-left:0cm;line-height:15.0pt;background:white;outline: none;orphans: auto;text-align:start;widows: auto;-webkit-text-stroke-width: 0px;word-spacing:0px'><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri",sans-serif;mso-fareast-language:EN-US'>7821 (sip78xx.10-2-1-12SR1-4)<o:p></o:p></span></p><p style='mso-margin-top-alt:0cm;margin-right:0cm;margin-bottom:7.5pt;margin-left:0cm;line-height:15.0pt;background:white;outline: none;orphans: auto;text-align:start;widows: auto;-webkit-text-stroke-width: 0px;word-spacing:0px'><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri",sans-serif;mso-fareast-language:EN-US'>9951 (sip9951.9-4-2-13)<o:p></o:p></span></p><p style='mso-margin-top-alt:0cm;margin-right:0cm;margin-bottom:7.5pt;margin-left:0cm;line-height:15.0pt;background:white;outline: none;orphans: auto;text-align:start;widows: auto;-webkit-text-stroke-width: 0px;word-spacing:0px'><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri",sans-serif;mso-fareast-language:EN-US'>ATA190 (1.1.2 (005) Feb 6 2015)<o:p></o:p></span></p><p style='mso-margin-top-alt:0cm;margin-right:0cm;margin-bottom:7.5pt;margin-left:0cm;line-height:15.0pt;background:white;outline: none;orphans: auto;text-align:start;widows: auto;-webkit-text-stroke-width: 0px;word-spacing:0px'><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri",sans-serif;mso-fareast-language:EN-US'>No Keepalive issue, because phone after removing voice register global connect again to the CUCM. <o:p></o:p></span></p><p style='mso-margin-top-alt:0cm;margin-right:0cm;margin-bottom:7.5pt;margin-left:0cm;line-height:15.0pt;background:white'><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri",sans-serif;mso-fareast-language:EN-US'><o:p> </o:p></span></p><p style='mso-margin-top-alt:0cm;margin-right:0cm;margin-bottom:7.5pt;margin-left:0cm;line-height:15.0pt;background:white;outline: none;orphans: auto;text-align:start;widows: auto;-webkit-text-stroke-width: 0px;word-spacing:0px'><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri",sans-serif;mso-fareast-language:EN-US'>Configuration of the SRST gateway are:<o:p></o:p></span></p><p style='mso-margin-top-alt:0cm;margin-right:0cm;margin-bottom:7.5pt;margin-left:0cm;line-height:15.0pt;background:white;outline: none;orphans: auto;text-align:start;widows: auto;-webkit-text-stroke-width: 0px;word-spacing:0px'><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri",sans-serif;mso-fareast-language:EN-US'>!<br>voice service voip<br> allow-connections h323 to h323<br> allow-connections h323 to sip<br> allow-connections sip to h323<br> allow-connections sip to sip<br> supplementary-service h450.12<br> fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none<br> modem passthrough nse codec g711ulaw<br> h323<br> sip<br> bind control source-interface GigabitEthernet0/1.20<br> bind media source-interface GigabitEthernet0/1.20<br> registrar server expires max 600 min 60<br> no silent-discard untrusted<o:p></o:p></span></p><p style='mso-margin-top-alt:0cm;margin-right:0cm;margin-bottom:7.5pt;margin-left:0cm;line-height:15.0pt;background:white;outline: none;orphans: auto;text-align:start;widows: auto;-webkit-text-stroke-width: 0px;word-spacing:0px'><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri",sans-serif;mso-fareast-language:EN-US'>!<o:p></o:p></span></p><p style='mso-margin-top-alt:0cm;margin-right:0cm;margin-bottom:7.5pt;margin-left:0cm;line-height:15.0pt;background:white;outline: none;orphans: auto;text-align:start;widows: auto;-webkit-text-stroke-width: 0px;word-spacing:0px'><span lang=EN-GB style='font-size:11.0pt;font-family:"Calibri",sans-serif;mso-fareast-language:EN-US'>!<br>voice register global<br> mode srst<br> timeouts interdigit 7<br> system message Systema SRST Attivo<br> max-dn 116<br> max-pool 58<br>!<br>voice register pool 1<br> registration-timer max 120 min 60<br> id network 192.168.101.0 mask 255.255.255.0<br> dtmf-relay rtp-nte sip-notify<br> voice-class codec 1<br>!<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB>Can you help me?<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-GB>Thank you regards<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-GB><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-GB>Alessandro Bertacco<o:p></o:p></span></p></div></body></html>