similar to: disable transfer on outgoing calls?

Hello! I''m using ArchLinux with kernel Linux horus 3.10.5-1-ARCH #1 SMP PREEMPT. Mounting and unmounting takes a long time: # time mount -v /mnt/Archiv mount: /dev/sde1 mounted on /mnt/Archiv. mount -v /mnt/Archiv 0,00s user 0,16s system 1% cpu 9,493 total # sync && time umount -v /mnt/Archiv umount: /mnt/Archiv (/dev/sdd1) unmounted umount -v /mnt/Archiv 0,00s user

Hi All, The Asterisk book (the pdf version) is excellent!! I want to thank all the guys that put it together. I am most grateful for it. There is something about writing a dialplan that I'm not clear about. What I'm trying to figure out how to do is this: when I transfer a call to the destination number, and the person called picks up the call, I want to play a greeting message to the

I upgraded from Asterisk 10 to 11 and switched from gtalk.conf and jabber.conf to use motif.conf and xmpp.conf. I disabled gtalk and jabber from loading in modules.conf noload => res_jabber.so noload => chan_gtalk.so After copying my settings to the new conf files and restarting Asterisk I had no errors, but making outgoing calls from clients just kept ringing even though the other side

Dear all, I have 3 backbones for my local network. 1st backbone: down 1024kbps, up 1024kbps through eth1 2nd backbone: down 2048kbps, up 2048kbps through eth2 3rd backbone: down 1024kbps, up 128kbps through eth3 Local network: 192.168.0.0/16 through eth0 Router: Linux Slakware 11 with iproute2 Please let me know how to shape both incoming and outgoing traffic for this case. LARTC doc only

Hello, I''m using a script with tc, to limit my outgoing traffic : tc qdisc add dev eth0 root handle 1: htb tc class add dev eth0 parent 1: classid 1:1 htb rate 500kbit burst 6k tc filter add dev eth0 parent 1: protocol ip prio 1 u32 match ip dst 172.28.54.9 flowid 1:1 I want to test bandwidth with iperf to see if limitation is correct. Sometimes, I get correct measure for certain rates,

there are 3 computers. A-------------B--------------C I want the outgoing packages of B go to C. And receive responding package from A. As soon as insert the routing entry: ip route add 10.0.0.0/8 via C src B dev BC B seems stop receiving packages from A. Use tcpdump on B, I can get the package. But, it seems that kernel check the routing table, and found that the source address of the

In regards to my earlier post, I believe that my problem is that the filter is not matching ssh traffic properly. Here are the rules that I posted earlier, with the filter being the last one: tc qdisc add dev eth0 root handle 1: htb default 3 tc class add dev eth0 parent 1: classid 1:1 htb rate 102400kbps ceil 102400kbps tc class add dev eth0 parent 1:1 classid 1:2 htb rate 102400kbps ceil

Hi, I''ve a router configuration with a dsl connection and two ethernet NICs. How can I control the traffic of the dsl connection when then destination of the traffic can go out of the router using two diferent interfaces? ppp -- router --- lan | DMZ I''ve read that the solution could be IMQ, but i prefer to not to patch kernel and iptables. Any

Hi, I have PRI configured and up but when i am dialing outside i am not getting any ringback tone but my call is connected. following is my example SIP----------------->PRI ------------> mobile I have set progress=yes in chan_dahdi.conf but still not working if i call inbound from my mobile to internal extension ringing working please help me -S -------------- next

Hi. I'm working with i4l with asterisk CVS-02/21/03-13:59:12, plus i4l (chan modem i4l *dsp patched* and kernel 2.4.19 patched to disable dtmf). All seems ok (apart some echo issues that seems gone with mec2 aggressive suppressor), but outgoing dtmf doesn't work . or at least I hear the very first part of the dtmf, but then it seems suppressed. here's my modem.conf [interfaces]

Hi All; I do not know if the SEP_Mac_address.cnf.xml of the cisco file is also has effecting on the DID and Caller ID to appear at the destination, because I found the following: <localCfwdEnable>true</localCfwdEnable> <semiAttendedTransfer>true</semiAttendedTransfer> <anonymousCallBlock>2</anonymousCallBlock>

Hi, My asterisk setup was working when using a Wildcard T100P. I recieved incoming calls and was able to make outgoing calls. I recently decided to change my card to a TDM01B one (one FXO). However when I dial a working extension, I can see on the console that the call goes through sucessfully, but it is never actually executed (i.e if I call my mobile phone, Asterisk shows the call as being

Hi, is there any way to manage outgoing calls from agents? Mi agents are answering in pstn lines. I can send agents outgoing calls to my Asterisk but I don't know wich agent is making the call...because, may be he is unregister... Is there any solution? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am trying to help a school run dansguardian transparent. I added the following to shorewalls rules, and from a tail of messages it seemed to be working, but he called saying no one had Internet. What should the rule be if this does not redirect port 80 to 8080? REDIRECT loc 8080 tcp www - !10.192.0.2 (web on dmz) ACCEPT fw net tcp www

c/c'd to the Asterisk list as this is probably relevant to Asterisk as well. My detailed study of the operation of the 'outgoing' directory reveals that TXFax() does not delete an expired fax batch file (In the 'outgoing' directory) until after the end of the dial plan execution. Is there a way to get it to delete any expired fax batch file before the end of the dial plan

hi all I just tried to test whether the HTB works well or not .. (policy is very simple ^^;;) I tried HTB simulator ''Ethloop'' with lo (loopback ) and I found HTB works well....(very well) So I changed lo to eth0 and tested .. ./tc qdisc add dev eth0 root handle 1: htb default 10 ./tc class add dev eth0 parent 1: classid 1:1 htb rate 500kbps ceil 500kbps ./tc class add

I'm trying now two write a rule that will filter out all outgoing port 80 requests from users. How do I make a list of the variables I'd like to search for? XXX.XXX.XXX.XXX,80 PR tcp len 20 52 -AF IN XXX.XXX.XXX.XXX,80 PR tcp len 20 52 -AR IN XXX.XXX.XXX.XXX,80 PR tcp len 20 52 -S K-S IN the AR, AF and S-K are the three I'd like to search for. Here's a bit of code I've got

Hi List, i've set up an Asterisk 1.6.0.3 Server equipped with an Xorcom Astribank BRI XR0013 (2 Port BRI) and Dahdi 2.1.0. In- and outgoing calls are no problem. But I can't get Asterisk/DAHDI to use the second Number of the BRI. All calls set the primary number as callerid. It wouldn't be a problem, if we would use this one as id for our company, but it's already in use (private).

"sip show channels" shows some info about active sip channels, the current codec included. What does it say? jg" jg, sip show channels reports the Format as being ulaw for 17 active calls. Holds - no Peer User/ANR Call ID Format Hold Last Message Expiry Peer xxxxxxxxxx kbrown xxxxxxxx (ulaw) No Rx:

Hello, according to man (8) ftp-proxy, "-a 1.2.3.4" should instruct ftp-proxy to use 1.2.3.4 as source address for outgoing control connections. But it doesn't. It seems to greatly ignor that directive, since I can pass any address, ieven if the machine doesn't own it. It always uses the EGRES interface's inet address - inet6 not tested. Any ideas why? Thanks, -Harry