search for: skypeout

...an not see how to solve the question. Thank you for your help. With Best, Cathy ========================================================== ?????q?????V???B50?W http://edm-prg.epaper.com.tw/click.php?ad_code=1196890 ========================================================== SkypeOut ?????[0.7??/?? ???j??0.9??/?? ???Y???R>> http://skype.pchome.com.tw/skypeout.htm

I'm running Asterisk 1.6.2.5 with chan_skype on a x64 linux platform. When a user calls from skype (not skype-in) to asterisk, dtmf (basically menus for a conference system) works just fine. But when a user from the inside (soft or hardware sip phone) calls out via skype-out dtmf doesn't work. I have tried setting the codec to alaw, and dtmfmode to all possible options (auto, inband and

Anyone have any ideas why DTMF would not work on only one number? Looking through the logs, anytime a button is pressed, this is what shows up: 2006-04-13 11:39:18 DEBUG[22441] chan_zap.c: Exception on 9, channel 1 2006-04-13 11:39:18 DEBUG[22441] chan_zap.c: Got event Dial Complete(9) on channel 1 (index 0) 2006-04-13 11:39:18 DEBUG[22441] chan_zap.c: Echo cancellation already on We

I am aware of how to match a particular caller-id or a caller-id pattern and do something with the call like this: exten => 15554441212/_888NXXXXXX,n,Playback(GoAway) What I am curious about, is the best way to block unknown, private and 000-000-0000 calls. I know I can do this for 000-000-0000 calls: exten => 15554441212/0000000000,n,Playback(GoAway) Is there a better way to catch

...t; >> translation path from 0x100 (g729) to 0x8 (alaw) > >> > >> on the console. > >> > >> Fired up a sip client, made the same call, and all was ok. > >> > >> Any clues ? > > > > The clues are in the documentation; SkypeIn and SkypeOut use G.729 for > > nearly all calls, so handling calls via those paths requires a G.729 > > transcoder on the system if the target of the call will not also be > > using G.729. This is why the Skype For Asterisk license includes > > licenses for Digium's G.729 software tr...

...O! I just see a skype channel as something good for asterisk. Skype has broad coverage. I can't imagine that skype wouldn't be interested in selling corporate accounts "skype trunk lines". Imagine having unlimited or X amount of continious calls coming in on SkypeIN and out on SkypeOUT from Asterisk. Internal Phones would all talk IAX or SIP to asterisk and use all PBX features of Asterisk. Asterisk could do lowest cost routing for internal to external calls through dundi or skype lookups (or enum later on) or through your PSTN hardware. Skype would really benefit from Asterisk...

...-mark $P2PMARK $IPT -t mangle -A PREROUTING -p udp -m ipp2p --ipp2p -j MARK --set-mark $P2PMARK # referente ao skype SKYPEMARK="21" $IPT -t mangle -A PREROUTING -p tcp -m layer7 --l7proto skypetoskype -j MARK --set-mark $SKYPEMARK $IPT -t mangle -A PREROUTING -p tcp -m layer7 --l7proto skypeout -j MARK --set-mark $SKYPEMARK $IPT -t mangle -A PREROUTING -p udp -m layer7 --l7proto skypetoskype -j MARK --set-mark $SKYPEMARK $IPT -t mangle -A PREROUTING -p udp -m layer7 --l7proto skypeout -j MARK --set-mark $SKYPEMARK # referente ao msn MSN="22" $IPT -t mangle -A PREROUTING -p al...

...tp class. Someone can help me ? Here is my script : #!/bin/bash IPT_BIN=/sbin/iptables TC_BIN=/sbin/tc INTER_OUT=ppp0 LINK_RATE_UP=1000Kbit RATE_ACK=200Kbit RATE_DEFAULT=100Kbit RATE_12=12Kbit RATE_13=13Kbit RATE_14=14Kbit NB_filtre_12=1 NB_filtre_13=2 NB_filtre_14=4 PROTO_12_1=http PROTO_13_1=skypeout PROTO_13_2=skypetoskype PROTO_14_1=edonkey PROTO_14_2=gnutella PROTO_14_3=applejuice PROTO_14_4=bittorrent # Delete all qdisc on $INTER_IN and $INTER_OUT $TC_BIN qdisc del dev $INTER_IN root 2> /dev/null > /dev/null $TC_BIN qdisc del dev $INTER_IN ingress 2> /dev/null > /dev/null $TC_...

We are in the process of preparing to move our Asterisk server to a Digital T1 interface card instead of a analog card (via an Adtran which is now connected to the T1). I did a preliminary test the other day and hooked the T1 line up to the T1 card, bypassing the Adtran. This worked rather well I must say. The two issues I ran into are: 1) Caller ID is not working even though I enabled

...N while [ ${j} -le ${i} ]; do iptables -t mangle -A ${dev[2]}_SKYPE -m layer7 --l7proto `sed -n ${j}p /tmp/2` -j RETURN j=$(($j+1)) done iptables -t mangle -A ${dev[2]}_SKYPE -m layer7 --l7proto skypetoskype -j ${dev[2]}_CON_VOIP iptables -t mangle -A ${dev[2]}_SKYPE -m layer7 --l7proto skypeout -j ${dev[2]}_CON_VOIP>/dev/null 1>/dev/null 2>/dev/null 3>/dev/null 4>/dev/null iptables -t mangle -A ${dev[2]}_SKYPE -j RETURN } ipt_int() { iptables -t mangle -N ${dev[2]}_CHECK iptables -t mangle -N ${dev[2]}_IMQ iptables -t mangle -N ${dev[2]}_PRIO if [[ ${dev[0]} =~ '...