similar to: Got SUBSCRIBE for extensions without hint

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

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

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 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

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."

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

We recently deployed 10 SNOMs as part of a PBX hosted solution. We have one phone setup as the receptionist phone, using hints to show busy office lines. This all works as expected. This is a new installation, and people are just starting to setup their phones. For those of you not familiar with SNOM phones, there is a row of keys on the right side of the phone which SNOM calls function keys. In

hi, To post, and to reply to a post, i have to goto my email. Now, if there is a web interface to these mailing list, things would be easier.

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 will be on vacation from <22/06/06> to <30/06/06>. I will not be reachable on my mobile. I will have limited access to mails, and please expect a delayed response. In my absence, please contact the following: Ray Richard or Safeer Mohammed Thanks H.Gireesh

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

Hi R-helpers, I would like to subset my dataframe, keeping only those rows which satisfy the following conditions: 1) the string "dnv" is found in at least one column; 2) the value in the column previous to the one "dnv" is found in is not "0" Here's what my data look like: ??? POND_ID 2009-05-07 2009-05-15 2009-05-21 2009-05-28 2009-06-04 4 ? ? ? 101 ? ? ?

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

Greetings, I have just updated our test server to 2.6.9-34.EL and get the following error messages when compiling zaptel: make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-i686' CC [M] /usr/src/zaptel/zaptel-1.2.1/zaptel.o /usr/src/zaptel/zaptel-1.2.1/zaptel.c:383: error: syntax error before "zone_lock" /usr/src/zaptel/zaptel-1.2.1/zaptel.c:383: warning: type defaults

Greetings, We are currently testing a Sipura SPA-3000 as a gateway from our Asterisk system to a PSTN line for 911 access. We have a number of locations and want to place an SPA-3000 in each, connected to a PSTN line that will provide the correct ANI/ALI information to 911 for each location. It all works great, except for a reasonably significant (4 seconds) delay between when the SPA-3000

Hi there, We're trying to get IMAP voicemail storage working on an MS Exchange server - I would be grateful if anyone who has successfully done this could post the magic soup here, as extensive Google searching has yielded nothing other than tantalizing references to it being done without any specifics. -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web:

Hi R-helpers, I'm trying to use this code > pvh_dnv<-pvh[sapply(pvh=="dnv"),] to make a new dataframe containing the rows from pvh that contain the value of "dnv" in ANY column. But, it's not working. I get this error Error in match.fun(FUN) : element 1 is empty; the part of the args list of 'is.function' being evaluated was: (FUN) which, to

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