search for: albanach

...et=yyyy host=sip.gradwell.net context=flat fromdomain=sip.gradwell.net nat=yes allow=all canreinvite=no dtmfmode=inband qualify=yes [talklite] type=peer username=9479xxxx qualify=yes secret=zzzz host=sip.talklite.net canreinvite=yes disallow=all allow=ulaw [2201] type=friend context=flat username=albanach secret=aaaa defaultip=192.168.1.100 qualify=yes type=friend callerid="Russell Horn" <0000> host=dynamic nat=no ; X-Lite is behind a NAT router canreinvite=yes ; Typically set to NO if behind NAT allow=all =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-...

Hi, I''m trying to set up networking on a new machine. I''m not getting any networking from the domU''s I notice that I have two bridges being created: xenbr0 Link encap:Ethernet HWaddr FE:FF:FF:FF:FF:FF inet6 addr: fe80::200:ff:fe00:0/64 Scope:Link UP BROADCAST RUNNING NOARP MTU:1500 Metric:1 RX packets:3024 errors:0 dropped:0

Hi, I've set up a Gizmo Project account for access on my Nokia E61 because they work through NAT. Trouble is If I include my gizmo account in an asterisk hunt group and I'm not connected (phone is off / outside wireless coverage) the gizmo project always answers. Either the call goes to voice mail or if I turn voicemail off the call gets answered by a recording saying I'm not

Is it possible using asterisk to allow someone to dial in and remotely change where their call is forwarded to? For example, I'm working from home so I want my calls to go to 555 1234, now I need to go out for a bit so I'd like to phone the office and using DTMF tell the asterisk PBX to now forward my calls to my cell phone 555 3456 Has anyone implimented anything like this? R.

I'm able to receive but not make calls with zaptel using an X101P connecting to Asterisk with an Xlite client. My client has context = flat in sip.conf and extensions number 8919 In extensions.conf I've got: [home] ; Line 1 ; exten => 8919,1,Dial(${PHONES1},20,Ttm) exten => 8919,2,Macro(vmessage,${PHONES1VM}) exten => 8919,3,Hangup [outgoing] exten =>

Is anyone else having problems with them? Until today everything was working fine. But now dtmf is not working on incoming calls. Any ideas? I tried calling them and their voicemail is not accepting answers. Is there another source for DIDs in the 314 or 636 area codes? Especially a company that supports something besides ulaw. I am going to hate switching numbers again, my wife is

I've hit a problem with Voicemail. My call gets answered but the 'beep' before I should start recording a message doesn't end - it gets a little quieter. I can leave a message over the top of it, but the recorded message is very quiet. Any idea what might be the cause of this problem? My config is pretty basic at the moment: [general] format=wav attach=yes [default] 101 =>

Hi, We are using MozIAX as a softphone with a USB headset and are making outbound calls using IAX with ulaw encoding to our voip provider. We're running asterisk 1.4 Users are complaining that the ring tone generated by asterisk is much louder than the voice call once connected. They are having to turn the volume down to avoid being deafened by the ring tone, but then have an unacceptably

Hi folks, Moving on to a new install, I'm jumping straight to v1.4 Without using Priority jumping I'm wondering what the 'standard' way to indicate to the calling party that the number the dialed is busy or unavailable. So,if I have an entry in extensions.conf like this: [outbound] exten => _01.,1,SetCallerID(01235554321) exten =>

Hi, Incoming calls to one of my lines are set to ring two internal lines and simultaneously start ringing my cell phone. Something like this: exten => s,1,Dial(SIP/2201&SIP/2202&IAX2/my_cell at carrier),90) The internal lines 2201 and 2202 will both see the callerID for the incoming call, but my cell phone will show the callerID for asterisk, not the calling party. What's the

Hi, I've a DID number that gets passed to three internal phones and a cell phone via my outbound IAX trunk. If the cell phone is off or out of coverage, its voice mail captures the call. What's the best way to avoid this? Is there a recommended way to force the cell phone user to press 1 before the call is passed there ala google voice? Or is there another way to detect the presence of

Hi folks. At the moment I'm using tinc to connect to linux boxes, both of which are behind NAT. This works just fine. I'd now like to connect some roaming windows users to the network so they can browse a samba share like I can already do with linux. These folk will likely have dynamic IP addresses. The server has an IP of 10.0.0.2 My client has an IP of 192.168.1.100 Here's what

Hi folks, Sorry to trouble you all - I'm having some trouble getting tincd up and running - I suspect I'm having problems with subnets. Any help would relaly be appreciated! At the moment I'm trying to get two linux boxes, both of which are running as NAT routers for their respective networks to talk. My goal is to allow staff to connect to the corporate network from their laptops

This is all on SuSE enterprise 10 Our server was moved to a new location last night. After powerup there was no sign of networking so our ISP reconfigured the box. Outbound connectivity had previously been through eth1 but is now through eth0 Since restarting Xen I now have no network connectivity through the hosts. I have tried to provide all the relevant details below. If anyone has a moment

Hi folks, I've been reading the definitive guide to samba3 and Samba3 by example as well as scouring on line for advice. Sometimes however I find different suggested solutions to the same problem, so perhaps the list can give me some help with current strategies for these issues. We are a small office and are currently going through an IT refurb. Previously we've had a mix of laptops and

Since Thursday evening my asterisk box has been failing to register with broadvoice. I haven't changed any of my config files in the last week. Can anyone suggest anything? Asterisk is reporting: *CLI> Aug 28 16:15:17 NOTICE[6150]: chan_sip.c:3914 sip_reg_timeout: Registration for '703XXXXXXX@147.135.8.129' timed out, trying again -- Got SIP response 404 "Not found"