Re: [asterisk-users] linksys PAP2t and asterisk
Hi: yes i think this is it ,but what is it and how can i remove it ? Date: Sat, 14 Feb 2009 14:23:27 -0700From: floj...@gmail.comto: asterisk-us...@lists.digium.comsubject: Re: [asterisk-users] linksys PAP2t and asteriskMan, as the CLI says: SIP/us-092acb78 is ringing (here it gives me a fake ring) It's the channel SIP/us/something, which is generating ring signalling. 2009/2/14 wassim Darwish this post is attached to the prevoius post, this is what i have on CLI when i call from Linksys pap2t to asterisk and then asterisk bridge the call to a sip provider:-- Executing [88017736288...@direct:1] Dial("SIP/490115-092bacc8", "SIP/us/88017736288155") in new stack-- Called us/88017736288155-- Call on SIP/us-092acb78 left from hold-- SIP/us-092acb78 is making progress passing it to SIP/490115-092bacc8-- SIP/us-092acb78 is ringing (here it gives me a fake ring) how can i disable this ringing . From: wassim...@hotmail.comto: asterisk-us...@lists.digium.comdate: Fri, 13 Feb 2009 20:08:20 +0000Subject: [asterisk-users] linksys PAP2t and asterisk Hi all:when i make a call from linksys pap2t to an asterisk server a fake ring is heard some times ,but when sending calls between 2 asterisk servers through sip no fake ring is heard but real one. any suggestions please. Windows Live™: E-mail. Chat. Share. Get more ways to connect. Check it out. Windows Live™: E-mail. Chat. Share. Get more ways to connect. Check it out.___-- Bandwidth and Colocation Provided by http://www.api-digital.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Jose Flores Galicia<>BriefCode && Code Based Training _ Windows Live™: Keep your life in sync. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_t1_allup_explore_022009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] linksys PAP2t and asterisk
Man, as the CLI says: SIP/us-092acb78 is ringing (here it gives me a fake ring) It's the channel SIP/us/something, which is generating ring signalling. 2009/2/14 wassim Darwish > this post is attached to the prevoius post, this is what i have on CLI > when i call from Linksys pap2t to asterisk and then asterisk bridge the call > to a sip provider: > -- Executing [88017736288...@direct:1] Dial("SIP/490115-092bacc8", > "SIP/us/88017736288155") in new stack > -- Called us/88017736288155 > -- Call on SIP/us-092acb78 left from hold > -- SIP/us-092acb78 is making progress passing it to SIP/490115-092bacc8 > -- SIP/us-092acb78 is ringing (here it gives me a fake ring) > > how can i disable this ringing . > > > > -- > > From: wassim...@hotmail.com > To: asterisk-users@lists.digium.com > Date: Fri, 13 Feb 2009 20:08:20 + > Subject: [asterisk-users] linksys PAP2t and asterisk > > > Hi all: > when i make a call from linksys pap2t to an asterisk server a fake ring is > heard some times ,but when sending calls between 2 asterisk servers through > sip no fake ring is heard but real one. > any suggestions please. > > > > > -- > > Windows Live™: E-mail. Chat. Share. Get more ways to connect. Check it > out.<http://windowslive.com/explore?ocid=TXT_TAGLM_WL_t2_allup_explore_022009> > > -- > Windows Live™: E-mail. Chat. Share. Get more ways to connect. Check it > out.<http://windowslive.com/explore?ocid=TXT_TAGLM_WL_t2_allup_explore_022009> > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Jose Flores Galicia <> BriefCode && Code Based Training ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] linksys PAP2t and asterisk
this post is attached to the prevoius post, this is what i have on CLI when i call from Linksys pap2t to asterisk and then asterisk bridge the call to a sip provider: -- Executing [88017736288...@direct:1] Dial("SIP/490115-092bacc8", "SIP/us/88017736288155") in new stack-- Called us/88017736288155-- Call on SIP/us-092acb78 left from hold-- SIP/us-092acb78 is making progress passing it to SIP/490115-092bacc8-- SIP/us-092acb78 is ringing (here it gives me a fake ring) how can i disable this ringing . From: wassim...@hotmail.comto: asterisk-us...@lists.digium.comdate: Fri, 13 Feb 2009 20:08:20 +0000Subject: [asterisk-users] linksys PAP2t and asterisk Hi all:when i make a call from linksys pap2t to an asterisk server a fake ring is heard some times ,but when sending calls between 2 asterisk servers through sip no fake ring is heard but real one. any suggestions please. Windows Live™: E-mail. Chat. Share. Get more ways to connect. Check it out. _ Windows Live™: E-mail. Chat. Share. Get more ways to connect. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_t2_allup_explore_022009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] linksys PAP2t and asterisk
Hi all: when i make a call from linksys pap2t to an asterisk server a fake ring is heard some times ,but when sending calls between 2 asterisk servers through sip no fake ring is heard but real one. any suggestions please. _ Windows Live™: E-mail. Chat. Share. Get more ways to connect. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_t2_allup_explore_022009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users