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Hi Got a new iphone, want to know about peoples experience with any apps that work well with asterisk and run on a iphone Alex -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20091213/1f81f008/attachment.pgp

Hi I am having some problems installing a paravirt RHEL guest on a deb amd64 xen machine uname -a = Linux debxen 2.6.20-1-xen-amd64 #1 SMP Tue Apr 24 22:23:22 UTC 2007 x86_64 GNU/Linux ============== xm info ============== host : debxen release : 2.6.20-1-xen-amd64 version : #1 SMP Tue Apr 24 22:23:22 UTC 2007 machine : x86_64

Hi I had a working system, until recently - its asterisk 1.6.1 from debian - not the lastest as the last doesn't seem to work. but somebody who rang me said my voice mail announcement was all stuttery. so i dialed my voicemail box and its really stuttery... so I have done a reboot and its just as bad, now I am not sure what to check to try and get this working again ..... Alex

Hi I am in the process of installing a new box and using dahdi. I have a tdm410 + hardware echo canceller. I have just read in the read me for dadhi that VPMADT032 support has been removed and unlike with the zaptel stuff i could just download and install the firmware I can't with dahdi what is the best way forward to recompile with hardware echo canceller support. Thanks Alex

Hi recently bought a soekris net5501 and a tdm410 to place in there. I am having some issues attaching 12V power to the card via the molex card - basically the box for the motherboard is too small. I have read up about a PWR2400b and it seems to use 2wire pin, I am guessing to connect to P8 just behind the molex connector on the tdm410. can any one here confirm this, or have any info on the

Hi, In a normal PBX environment a user usually calls in and IVR's are played according to a predefined dialplan. Iam trying to develop an application where asterisk dials out to a user and initiates an IVR instead (please note that the IVR is not static and may vary according to different condtions). Can someone guide me how this is possible using Asterisk. Do I need to write some sort of

Hi I was reading this article on installing asterisk 1.6 + debian http://www.howtoforge.com/installing-and-configuring-asterisk-1.6-and-postgresql-to-manage-cdr-and-realtime-config-on-debian and I noticed they suggested to recompile to 1000Hz enable kernel, I currently have a 250Hz stock standard kernel. I am running on a soekris board - amd geode cpu. Is recompiling the kernel to the

I''ve applied julian''s paches to a 2.6.14 gentoo kernel with the appropiate options enabled, and i''m using a modified version of the mpath2.sh script also available on julian''s site http://www.ssi.bg/~ja/ Overall everything works nearly perfectly. Incomming connections to either the fios (PPPoE) connection, or cable modem get routed back out correctly. The

Hi I am trying to setup asterisk to do a mass deploy of some snom phones. I can't find where i configure asteriks to listen to the multicast address, nor where to set the notify reply. I was hoping to not have to use dhcp options alex -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc:

Hi I have a site that has asterisk install with a tdm410 one port is connected to a pstn that is used as a backup outbound line when/if the internet/voip is unavailable. Currently my dial plan for this line is to ignore it, I just basically do a s,1,noop s,n,wait (60) s,n,hangup what I would like to do is if I call from a certain number be given an option to run some commands or for the

Hi my setup debian amd64 with ii samba 2:3.4.2-1 SMB/CIFS file, print, and login server for Unix ii samba-common 2:3.4.2-1 common files used by both the Samba server and ii samba-common-bin 2:3.4.2-1 common files used by both the Samba server and ii samba-doc

Hi I have just moved my firewall from a 2.6 debian machine to a 2.4.30 openwrt (linksys wrt54gs) box. I orginially had this working with 2 isp, 1 cable 1 adsl and dyndns. Now when i have moved to 2.4.30 I am having problems. Everything else is working fine except when I DNAT packets from the firewall to an internal address, ie my web browser is inside so I DNAT from the external IP to the

Hi, I have a custom agi which at times does not exit gracefull and crashes in between. The logging options are set to the maximum but I dont see something conclusive in the asterisk log. I have noticed it crash after issuing the "SAY NUMBER" and "GET DATA" agi commands and the agi is spawned with no apparent reason after that. I tried running the application locally and

Hi! I''m trying to create a routed VPN using OpenVPN - and having trouble with the routing concepts involved. Let me see if I can properly describe my current topology: Server - LAN, with both local workstations and remote bridged workstations on the 192.168.0.0/24 network (this works without reservation). Server located at 192.168.0.71, 192.168.0.72, 192.168.0.222, and few

Hi I have come across a problem, with my tdp410 and soekris board (basically pc on a chip amd geode cpu). I am using the box as a firewall/asterisk box. The problem occurs when I drop ppp and I get dead loop dectiotn going, I seem to lose interrupts and get lots of messages in syslog from wctdm24xx saying missed interrupt increasing latency its out lined here

Hi I have setup forwarding - xfering - where you press # and then the extension. I add t to the dial cmd. My problem is that when you call something like internet banking they want #, but when # is pressed asterisk gets it instead. is there a way around this ? I haven't been able to get asterisk to listen to flash either Alex -- Why is it there are so many more horses' asses than

Hi, I am facing this issue, where I get a delay of aroud five seconds when switching between contexts (through extension.conf) . This is how my extensions looks like. [salesivr] exten => _X.,1,NoOp(Incoming call from user ${EXTEN} and caller id ${CALLERID}) exten => _X.,2,Playback(emptyy) exten => _X.,3,Background(Main_Sales) exten => _X.,4,WaitExten(2) exten => _X.,5,Goto(_X.,3)

* sorry for the other schema, it came out a mess. I hope this one is understandable. hi: I have this 2 boxes set up like this: 2020::2/128 2020::254/128 3030::254/128 192.168.0.2/24 192.168.0.254/24 192.168.30.254/24 +-------+ eth0 eth3 +-------+

Hi This is the output from show dialplan dial-sipmnf-sippt-pstn [ Context 'dial-sipmnf-sippt-pstn' created by 'pbx_config' ] 's' => 1. Verbose(1,Dialing ${ARG1} on mnf pt pstn) [pbx_config] 2. Dial(SIP/${ARG1}@${SIPMNF},${ARG2},${OUTBDIAL}) [pbx_config] 3. Set(GLOBAL(FOUNDME)=${DIALSTATUS}) [pbx_config]

...expect it to just sort of work with the digium card. the os is debian amd64 2.6.26 #dpkg -l asteri* | grep ^ii ii asterisk 1:1.4.21.2~dfsg-3 Open Source Private Branch Exchange (PBX) ii asterisk-barbarast.com 0.0.0-1 asterisk setup for hme1.samad.com.au ii asterisk-doc 1:1.4.21.2~dfsg-3 Source code documentation for Asterisk ii asterisk-sounds-extra 1.4.7-1 Additional sound files for the Asterisk PBX ii asterisk-sounds-main 1:1.4.21.2~dfsg-3 Core Sound files for...