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...},1,SetCIDName(SIP - ${CALLERIDNAME}) exten => ${SIPPHONEUSERID},2,Dial(SIP/20,15) exten => ${SIPPHONEUSERID},3,Goto(mainmenu,s,1) exten => ${SIPPHONEUSERID},4,Hangup ... Any suggestions/help would be greatly appreciated. Thanks, Alvin PS: Please cc me directly on replies: a a n (at) crlogic (dot) com

Hi, I am getting the following error when I attempt to listen to voice messages by dialing 9999 (I can hear nothing): --Executing VoiceMailMain ("SIP/2092-6918", "2092") in new stack --Playing 'vm-password' (language 'en') WARNING: app_voicemail.c:4922 vm_authentication: Unable to read password. I read in previous posts that this may be to do with the dtmf

Hello Asteriskers, :-) We're trying to set up an outbound notification calling for system alerts with Asterisk 1.4.0. We generate a call file in /var/spool/asterisk/outgoing and the outbound call is originated through Zap/1 (Sangoma A200D to a Canadian POTS line). The problem is that Asterisk does not wait for the other side to answer before it starts playing the message. So the

Hi everyone, I'm trying to get a Sangoma A101D-X card talking to a Sasktel PRI (Megalink) circuit and having some trouble getting it to handshake. Thanks for any help or suggestions to diagnose this problem. The problem is that Asterisk has trouble bringing up the link. I see the following lines every couple of minutes: == Primary D-Channel on span 1 up == Primary D-Channel on

0.8 appears to have been released. Start with that. It is very quick to update your current 0.6 or 0.7 iso. Just do this with rsync to do a differential copy: $ mv asteriskathome-0.7.iso asteriskathome-0.8.iso $ rsync -av --progress --partial \ prdownloads.sourceforge.net::sourceforge/a/as/asteriskathome/asteriskathome-0.8.iso . $ rsync -av \

After searching around, I've been unable to to find any relevant info on this. Perhaps the group can help? I am seeing something strange with a new Sipura SPA-3000 (and I've noticed this also with an IAX softphone): When I dial 777, this dialplan (in extensions.conf) is run: exten => 777,1,Dial(Zap/1/2345678) exten => 777,n,Hangup The number is answered by the called

Hello, I'm wondering how one might set up a feature to add (in real-time) the current CallerID information to a junk-callers database. After answering a call from an outside line and determining that the call was from a telemarketer or the like, the user could dial an easy specific code (like ** or 77), which would cause the call to be transferred to a specific extension within the

Hi everyone, I have a user that needs a little extra volume on his Polycom IP 601 phone set for all calls (beyond what the volume control currently offers). Is there a provisioning setting for this anywhere? (I'd like to avoid a separate amplifier between the phone and handset if possible.) Alternatively, is there a way to have Asterisk 1.4.x boost the volume to a particular SIP device

Anyone successfully using the Polycom SoundStation VTX 1000 with Asterisk? I can't see any mention of it on the wiki page: http://www.voip-info.org/wiki-Polycom+Phones Thanks, Alvin

Hi Polycom experts, I'm having a problem getting changes to the Polycom IP 601's (mac)-directory.xml file to update the button list on the phone. If the phone is newly provisioned (i.e. if I "Format File System" on the phone) then the new list will show up on the buttons, but of course this is pretty drastic way to do it. - Environment: Asterisk test setup with 7 phones,

One more Polycom IP601 question please (sorry for the long intro here to document) ... In order to closely approximate the behavior of the previous telephone system that many of the users are familiar with, I have set up call parking like this: - features.conf [general] section contains: parkext => ** ; What extension to dial to park parkpos => 10-11 ; What

Hello, I'm looking for a recommendation on an Asterisk consultant to take over support for a working Asterisk 1.4.x system in the London, Ontario area. The company is looking for a local support partner. The system uses a PRI line from Bell, and mainly Polycom IP600 phones. Please contact me OFF-LIST if you have had a good experience with an Asterisk expert in that area. Thanks, Alvin

When mirroring a directory containing versioned files with rsync, it seems that extra data is being transferred unnecessarily when a new version of the file is put in the directory. For instance, a directory contains files: someprog-3.7-beta1.ext otherprog-9.2.ext ... and this is mirrored to another system with rsync. Then the site updates someprog-3.7-beta1.ext to

Hello, In Asterisk 1.4 beta 3, the UPGRADE.txt file says: Variables: * The builtin variables ${CALLERID}, ${CALLERIDNAME}, ${CALLERIDNUM}, ${CALLERANI}, ${DNID}, ${RDNIS}, ${DATETIME}, ${TIMESTAMP}, ${ACCOUNTCODE}, and ${LANGUAGE} have all been deprecated in favor of their related dialplan functions. You are encouraged to move towards the associated dialplan function, as these

Hello all, The wiki has a fairly detailed description of the the issues involved with encryption of Asterisk calls: http://www.voip-info.org/wiki/view/Asterisk+encryption I'm interested in hearing what is working for people today. I think the ideal solution would be a hard phone that could be plugged in almost anywhere (dsl/cable modem, hotel, etc) and connect securely to a remote