search for: powderday

Hi I thought this might be of general interest. Recently I purchased an X100P from a Digium reseller in the UK. Very pleased with the card; works perfectly. My friend (known for his deep pockets and short arms) purchased an X100P card from Ebay. He's had no end of problems with line noise, dropped called etc so I thought I would compare the two cards. Plus mine was delivered within two

Hi all. I'm looking to set up a fax via email service so that users can email a specific mailbox and receive fax's to a specific mailbox. Can this be done? I've had a look an SpanDSP and I think that's what I want but I'm not sure. Cheers Matt

Hi all, I've been through the wiki and the archives and I've been unable to find what I'm looking for. Basically I have a phone that I don't want to be able to dial out. I've only got one context at the moment; but from my investigations I think I might need to create another for this specific phone. Can anyone point me in the right direction? Thanks Matt

...sionsonly" and then within those contexts include the relevant contexts. If you look at the sample configs you can see how this is done for the international and local and you can extend that concept for your extension only dialling. Steve -----Original Message----- From: Matt [mailto:matt@powderdays.com] Sent: 21 June 2004 10:54 To: 'asterisk-users@lists.digium.com' Subject: [Asterisk-Users] Restricting outbound dialing on a specific phone Hi all, I've been through the wiki and the archives and I've been unable to find what I'm looking for. Basically I have a phone tha...

List, The D'Link phones are not reliable at this time. I am trying to get them fix their Firmware to my specifications. It is half done so far. However there are still hurdles. below email is self explanatory. At present if you want to use these phones, you need to buy D'Link's SIP Server and run this as one of your SIP servers in the blend to call to Asterisk. Seshu Kanuri "G

Hi all, I want to use my * box to control entry to a building. I was wondering who else has done this and what phones they might recommend. The phone itself needs to be externally mounted so will have to be durable. Functionally I would like it to just dial and extension when picked up. Any comments on your experiences would be very much appriciated. Best regards Matt

...: Thu, 22 Jul 2004 11:37:18 -0700 Reply-To: asterisk-users@lists.digium.com What about the DPH-1008 or is that just the US version?? -Chris ----- Original Message ----- From: "Kanuri, Seshu" <seshu.kanuri@citigroup.com> To: <asterisk-users@lists.digium.com> Cc: <matt@powderdays.com> Sent: Thursday, July 22, 2004 11:11 AM Subject: RE: [Asterisk-Users] D-Link DPH-80S vs * > Matt, > > Google does not give all the information you need, some times. > > Read my post on DPH80 > > Seshu Kanuri > > -----Original Message----- > From: asterisk-use...

This might have been talked about before, but I'm posting anyhow. I've got down to testing Asterisk yesterday, and I couldn't help but compare it with Skype (a Windoze only product, yet, but extremely efficient for some reason). Skype has almost unperceptible delay (LAN), while there is almost half a second of delay regardless of the codec on Asterisk. An even if we were to

Sorry to bug the entire list with this as this is really a question for those who have been sucessful in configuring [*] to place and receive a SIPCALL call. Everying looks right in my config, I can see it registered etc but when I try to place the call I get: -- Executing Dial("SIP/100-2371", "SIP/8703409095@sipcall/04") in new stack Apr 29 22:50:34 WARNING[27089840]:

Hi everyone, I've just got my X100P card installed and working but there seems to be an issue with hang-up supervision. If I stuff a call out over the X100P card onto the PSTN that's fine. When I hang up the SIP phone the PSTN call ends. If I receive a call from the PSTN, it's answered and everything is ok until the remote party hangs up. Asterisk thinks the call is still active and

Hi everyone. I've just tried installing the SIP image and i'm getting a very odd error. It says: Configuring VLAN then Configuring IP then Protocal Application Invalid Help!!!! Matt

Hi all My * implementation is working brilliantly with only one small fault left to kill. I'm using IAXTalk from Telappliant for my incoming/outgoing calls to the pstn network; if I set my codec to GSM everything works great - no pauses but quality is a bit poor. If it set the codec to alaw (I think I'm using the correct one - I'm in the UK) I get intermittent pauses on the call.

Hi everyone, I've got my * system up and running and I'm really pleased. I've gone with G.711 (alaw) and I've stumbled across a problem; when people place calls internally some people think they have been cut off if the line is quiet for a few seconds. Is there a way of getting comfort noise on the call? I'm using the STABLE release and cisco 7960 phones under FC-1 Cheers

Hi everyone I'm interested in using the Telappliant/voip-talk offering as an alternative to my DDI analog problem. (see [Asterisk-Users] Multiple DDI & Hunting on Analog Lines (UK) for details) Does anyone on the list have any recent comments on reliability etc? I would really appricated some positive and negative comments. Cheers Matt

Hi all, I'm stuggling with how to present calleds to a specific DDI (DID) with Music on hold whilst the call is hunted around 3 phones, then if not answered within a certain period forwarded to voicemail. So far I've got the queue working and the voicemail but not both together. Ive had a look on the wiki and the archives but can't spot anything that might point me in the right

Hello all. Has anyone managed to get SIPCALL.co.uk's service working with the [*] box? I've managed to register with other SIP providers but not SIPcall. The debug just show's [*] attempting to register. But receiving a 401 error everytime. Cheers Matt

Hi all, This is probably a really stupid question so I apologise in advance; I've been looking at this all day and after 12hours I've got nowhere fast. The situation is: I've got a couple of 7960's on wireless adapters, the wireless network can be, shall we say, a little flakey. The phones that are wired into the network directly via ethernet are always registered and work

I'm thinking about using asterisk as an IVR system with an existing avaya index system. I've got 2x PRI 30 lines coming in to the Index, and I have 4 spare PRI cards in the Index. I was thinking about using a QUAD PRI card from Digium and having the calls come into the Index then transfer to Asterisk for IVR then back to the Index. That way if we get 60 inbound calls we'd in

Hi everyone, I want to get multiple DDI's and hunting across those DDI's in case one of the lines is busy using analog phone lines. The system is for a large house so I want 3 x PSTN lines. 3 x DDI's and the ability for those DDI's to be presented across all three PSTN lines. BT say you can't have more than one DDI number associated with a PSTN line and that you can't

I've got a number (10) Cisco 7960's connected to my network. All the phones work perfectly except one. The asterisk console keeps throwing up: Jun 22 15:39:15 NOTICE[-1147470928]: chan_sip.c:5887 sip_poke_noanswer: Peer '4001' is now UNREACHABLE! Jun 22 15:39:27 NOTICE[-1147470928]: chan_sip.c:4925 handle_response: Peer '4001' is now REACHABLE! Jun 22 15:42:08