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 <[email protected]> > 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: [email protected] > To: [email protected] > Date: Fri, 13 Feb 2009 20:08:20 +0000 > 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 <<[email protected]>> BriefCode && Code Based Training
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