similar to: Quality monitoring

I remember seeing something like this on the list a while ago, but I'm darned if I can find it. We have a number of Polycom IP501 phones, some of which have more than one registration on them. When a voicemail is left for a phone with only one registration, the MWI lights up and stays lit until the voicemail is listened to. However, on our phones with more than one registration, the MWI

Hi there, I'm getting a bunch of these errors from Polycom phones in 1.2.1: ERROR[24301]: chan_sip.c:10790 handle_request_subscribe: Got SUBSCRIBE for extensions without hint. Please add hint to 4003 in context internal I've searched the Wiki and archives to no avail - what do these errors mean? Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web:

Trying to figure out if a bug report should be submitted. Can anyone on 1.2.x verify of this has been corrected? I am on CVS 8/2005 If a call comes in to an extension that dials more than one channel (rings at more than one phone) both calls in the CDR show a status of answered when only one is answered, the source channel is bridged to only one of the two destination channels, but both CDRs

You can even set it to zero. Mine works well when in zero. The line pick up immediately :> -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris Stenton Sent: Thursday, February 09, 2006 6:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SPA-3000 VOIP-PSTN

Greetings, We are trying to make our corporate directory (around 400 entries) available via TFTP to some Polycom IP501 phones. A small (~40 entries or so) file works, but the full file fails to load. Does anyone know what the upper limit on directory entries is? The size of the XML file itself is only 60K - you'd think that would all fit into the phone with no problems..... I would

Greetings, The Polycom SIP 1.5 Admin Guide says this: "3.1.8 Connected Party Identification Where possible, the identity of the remote party to which the user has connected is displayed and logged. The connected party identity is derived from the network signaling. In some cases the remote party will be different from the called party identity due to network call diversion."

Hello all, I have Asterisk 1.2.1 and a TE110P connected to a Nortel Meridian Option 81C system. The PRI line is currently setup as DMS100. Here are the relevant lines from zaptel.conf and zapata.conf: zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone = us zapata.conf: [channels] language=en context=from-internal musiconhold=default switchtype=dms100

Hi, I successfully connected 2 servers via IAX but I'm pulling my hair to connect 2 extra servers , Anyone connected 3 or 4 servers together ? is it possible ? I d like to share the dialplan so _2XXXX goes to server A _3xxxx goes to serverB _4xxxxx goes to server C etc from the 4 servers any example of which one is peer, which one is user or friend would help me :-) thanks jl

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Just emailing the list to see if anyone out there has used Vitelity? If so what has been your experience with service, support etc? Thanks Curt

I put iptables on my asterisk box and an odd thing occurs. I allow 5060 and 10000-20000. As soon as I start iptables and make a call it literally takes 60-90 seconds before the call even starts to ring. As soon as I shut iptables off, the call goes through immediately again. Its quite odd. The call does eventually go through and talks fine but it takes sooo long to connect. Anyone have some

The TDM01B is 4 port capable but has only 1 FXO module. I'm running asterisk 1.2.7 and zaptel 1.2.5. I cannot seem to get the TDM01B working. When I do the zttool it shows 4/1/0. I can dial out from a POTS phone up to the point that the cable plugs into the card. Here is my /etc/zaptel.conf loadzone=us fxsks=1 and here is my /etc/Zapata.conf [channels] language=en #include

Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them? I called Polycom tech support, who where utterly useless. Of course Polycom won't officially support it anyway, as they only support Asterisk Business Edition. We're using 1.2.9, but it's been ocurring for quite some time. We have about 35 phones and

I just implemented a system using a TE411P hardware echo cancellation card. Per Digium, I setup zaptel.conf, and zapata.conf the same way as I always have. To my surprise calls out to the PSTN had a terrible echo. 1 - 2 second delay, and quite clear. The echo was so bad that I had to remove the hardware echo cancellation module from the card. We are only using the 1st span of this card right

What is everyone out there doing in an all IP termination environment to change trunks when quality drops to a certain provider automatically? Thanks Curt

I wanted to see if there was anything reasonable in price out there yet that performed an RF to IP bridge via asterisk. What I mean by this is callers from PSTN can be patched to a UHF/VHF radio and vis-?-vis. I know there is an option available for the Avaya systems but it?s a little out of the price range I?m looking for (~$200/channel). Has anyone out there found a stable way to do this?

Sorry if this shows twice but it appears my first message was quarantined because of my digital signature. All, I have been tasked with setting up video conferencing utilizing asterisk. One of the requirements is a softset that has video capabilities. Eyebeam looks promising but I was just wondering if anyone out there knows of any freeware with comparable features of Eyebeam that they

I am having an issue with doing FTP auto provisioning of Polycom 501's when they are behind a NAT. If I put the phone on the same subnet as the provision server it loads the configs and changes fine but as soon as I put in behind a NAT it comes up with cannot contact boot server. I have tried behind and replicated this behind a PIX 501, a Linksys SOHO router and a Motorolla SOHO router.

I am using AgentCallbacklogin to logon agents. I am trying to avoid agents being logged in more than once in different extensions (is this a bug?) by passing the callerid to the AgentCallbacklogin funtcion as an option. The problem is that by doing this, agents are not asked for an extension and they cannot logoff (by pressing the #). Any ideas how can agents logoff? -------------- next part

Hello, I am using asterisk as voicemail for my sip proxy. When a user (1234)dials 1111, the call is forwarded to asterisk. However I receive the following error: --Executing VoiceMailMain("SIP/1234-9afc", "1234") in new stack --Playing 'vm-password' (language 'en') [WARNING]: app_voicemail.c:3359 vm-execmain: Unable to read password ==Spawn extension