similar to: Recall: Newbie Asterisk: Install Asterisk as non-root

When I downloaded the sip and bootrom from Polycom website, I noticed a file called SoundPointIPWelcome.wav. However, I have no idea where and when it was used. I played the wav file but I have never heard the phone using this wav file before. Does anyone know what it is used for?

I am using Polycom IP600 phone. If I call a phone which has DND (do not disturb) enabled, the message to the caller will be "The person on extension ... is on the phone, please leave a message ...". Is there a way to pick the "person ... not available" message instead?

I thought it was quite easy to implement but I cannot get one-touch recording to work. Here are the changes what I did: I restarted Asterisk after the change (because reload does not work for changes in features.conf). I press *1 on the Polycom IP600 phone to record a conversation but no new wav file appear in /var/spool/asterisk/monitor or elsewhere. Any suggestions? Here is the console log:

I have a question about DHCP and boot server supporting more than 1 model of Polycom phones. According to Polycom "standards", Polycom phone boots up to get a DHCP address and at the same time getting a boot server string (with username and password) to logon to boot server to download SIP, bootROM and etc. That is okay if there is only one type of phone (that requires a specific SIP

Hello, I have been running Asterisk for the past 5+ years on RedHat and I never upgraded it before. All my Asterisk software is of the following release: 1) Asterisk 1.4.21.2 2) Libpri-1.4.4 3) Zaptel-1.4.11 I would like to move the OS to CentOS and then I thought I can at the same time ponder about upgrading Asterisk releases. However, I am bewildered by the myriad of different releases like 1.6,

Dear All Can you please do me favor and let me know what is the problem with my Asterisk call parking as it is not functioning correctly on my Asterisk ? Please find attached my "features.conf" . According to my configuration , the subscriber needs to press hash (pound) key and dial 700 to initiate the transfer . We tried but it didn't get through on our Asterisk . Can you please let

I am struggling to find out how to code hint in AEL2. I did hint(Custom:light1) and it keeps complaining about the : (colon). It works fine for SIP device like hint(SIP/439). Anyone who has tried it before?

Hi Duncan The difference is that in your call to loadNamespace, the versionCheck list has 3 components (name, op and version), whereas the documentation only mentions 2 (op and version). loadNamespace 'works' for me provided I add a third component to the list (even a nonsense one). What I haven't yet had the fortitude to do is track down through the code to see what the arguments

Many thanks Duncan for the quick response. A bug is a relief in a way. I've been digging my way deeper into this (and learning more as I go) for several days now - but it is a diversion from (a diversion from) my main goal :-( Is there somewhere specific I should report or log the bug or will that happen from this mailing-list automatically? (I have seen the Bug Tracking link on the

Hi Duncan, Many thanks (yet again). With the hint given by your earlier email (viz that currently loadNamespace expects a 3rd component called name in the list that is used for the versionCheck argument) I had another look at what was going on with my toy examples yesterday evening. I'm still working on my issue, but thus far I have: 1) Confirmed that internal calls to loadNamespace

In "The future of Telephony", it says "... We should also note for security's sake you should always make sure that your [incoming] context never allows outbound dialing. (If by chance it did, people could dial into your system and make outbound toll calls that would be charged to you!) The book was demonstrating using a PSTN environment and the zapata.conf was something like:

I am trying to build a simple queue for the receptionist phone. In other words, there is only 1 agent and that is the receptionist phone. I just defined a few lines in queues.conf [console] strategy = ringall member => SIP/4000 ;4000 is the console extension In extensions.conf, it is: exten => 4000,1,Answer() exten => 4000,n,Queue(console) exten => 4000,n,HangUp() I pressed

I just managed to put in a TE410 card in an Asterisk box to work with OnRamp 20(E1 downunder). I am able to dial in but was not able to dial out. Can anyone offer me some advice please? In my extensions.conf, I just put in: [default] ... exten => 0,1,Dial(Zap/g1) and I get this on the console when I dialled 0. -- Executing [0 at default:1] Dial("SIP/5166-b76004f8",

I am working on Polycom IP601 console with expansion module. I want to put on the BLF (busy lamp field) feature on all the contact/speed dial names I put on the console but I could not get it to work. *CLI> core show version Asterisk 1.4.13 built by root @ hostname on a i686 running Linux on 2007-11-20 05:26:15 UTC *CLI> sip show subscriptions Peer User Call ID

Hi, I noticed that asterisk manager interface will only accept the originate commands in sequential order. For example, if I want to ring two extensions through the AMI, and while first extension is ringing, AMI won't execute and ring second extension until first extension has answered the call. Anybody has any ideas as I had the same results even tested with telnet commands to AMI interface.

I have been testing with Polycom IP600 phones for a month or so. I found out that there are frequent problems with the handset. The problem is I can hear the other end but the other end cannot hear me. I have already downloaded the latest bootROM 3.1.3 and sip 2.1.2 However, there are no problems with the headset or speaker phone. Has anyone encountered such problems before? Thanks.

I am writing an extension to accept speed dial nos. However, I forgot that these speed dials are the same for all offices and thus would ideally be shared by offices which will host their own Asterisk box. I read from a few postings that this database cannot be replicated to other Asterisk box. I was thinking that if I could just do a simple copy/paste of the speed dial records from the main

I was copying tracks from CD into mp3 files so that I could use it in Asterisk 1.4.21.2 MOH. (BTW, I have already secured proper license to play MOH to callers.) I used MS Media Player version 11 and rip it at 128kbps (smallest) but whenever I listen to MOH, I saw the following message on the Asterisk console. WARNING[20829]: mp3/interface.c:215 decodeMP3: Junk at the beginning of frame 49443303

> > As a result, I just go back to put all users in [default] in > voicemail.conf. > > Am I missing anything? >> What do those contexts mean in your setup (beside being arbitrary >> groups)? I just want to group the mailboxes by say department rather than putting them all under [default]. So, I could use CLI to "voicemail show users for sales" or

Hi I'm still exploring the R programming universe, so if this is being asked in the wrong place, or in the wrong way (e.g. too verbose or lacking in crucial detail or in the wrong format) please let me know I am trying to understand when the version constraints for packages which appear in the Imports field of a DESCRIPTION file are checked. Along the way I've hit a snag