similar to: Attached Voicemail does not play mac/linux

Does anyone out there have a recommendation for tools that will monitor the quality of VoIP systems? I am looking for jitter and MOS monitoring. I have a custom Nagios plugin that is alerting me if the jitter jumps out of a 20ms but I am looking for a little more detail. I would not be against writing something in Perl for Nagios to do but I don't really know where to start on measuring jitter

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

I've just got a batch of 301s and 501s in and am trying to get them to work. I'd like to manually configure everything via FTP rather than the web or phone interfaces, but I can't seem to find a good source of definitions for all the options in the sip.cfg or phoneX.cfg files. Anyone know of any? Also, I'm having quite the problem getting the Polycom SP 501 to send *any*

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

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

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

I was wondering if there was a middle ground between POTS lines and a T1. I have a new office with a T1 line and while it's working well, it's a lot of money and we will never use anywhere near 23 lines at one time. Is it possible to get a few ISDN lines or something and bundle them together? Basically I would like to get the digital features of the T1 PRI (DID number, etc...) but

Hi! I've got MWI working just fine for my 501, but it's on if I have -any- VM messages. I only want it on if there are *new* messages. Any ideas as to what I should be changing? Thanks! -Ken

Hi guys, I am busy planning to implement SE Linux on my asterisk box. Either that or I will use AppArmor from Suse. I just want to know what are others experiences/incidents with SE Linux or AppArmor thanks, yusuf

can the default voicemail folders (old, work, friends, etc.) be changed? for example, i'd like to configure asterisk so that there are only folders called friends and old. thanks -ben

Hi, I'm planning to install a new Asterisk server with a Digium TE410P card. Can I use Red Hat Advanced Server 4 (latest update)? Is this a good choice? Is recompiling Asterisk simple with kernel 2.6? Thanks -- Domenico Viggiani

Hi, I know that I can have an AutoAttendent menu play when someone is in a queue to say something like "Press 1 now to leave a message, or to continue holding stay on the line..." However, is there anyway to prevent that from happening until the caller has been on hold for say 5 minutes? In other words, I don't want the caller to leave a voicemail UNTIL they have been on hold for