[asterisk-users] Voicemail retention
Asterisk version 1.2.27 We are running into issues where people are not deleting their voicemails and it is filling up the storage for voicemail. We would like to run a script that dumps all voicemail that are older than X days. Can we simply check the date time stamp on the message directory and delete those files older than X days or will that mess up the sequence of the voicemails? Anyone have a smooth way of doing this in 1.2? Thanks Phil ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Show call coming back from Call Parking
Our operator has asked if it is possible that when a call times out in the call parking and comes back to her, if there is someway to show that call has come back from parking. I have looked all over the documentation and have come up with nothing so far. All I see when a call times out is: -- Stopped music on hold on Zap/25-1 == Timeout for Zap/25-1 parked on 702. Returning to park-dial,SIP/214,1 -- Executing Dial(Zap/25-1, SIP/214||t) in new stack -- Called 214 -- SIP/214-09086ff8 is ringing It appears that the park-dial is a context that Asterisk autogenerates so there is nothing I can do in that context. Has anyone else found a way to show that this a call returning and not a new call coming in? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZAP - Can't pickup calls on Analog Trunk
Are the trunks just pots lines (plain old telephone service lines)? If you don't know you could put an analog phone on the incoming lines and verify you can dial out. Also, if you call the line the phone should ring. If this is true then you will need fxoks in your pbx instead of the fxsks. They are lines from the phone company but unlike normal phone lines in the fact that they have multiple numbers per trunk line. Such as trunk 1 has the following numbers assigned to it: 555- 555-1112 555-1113 555-1114 555-1115 On the old PBX these lines are picked up and the then the called number is checked, from there the call is routed to the appropriate desk phone. Because of that I am not sure I can plug a standard phone in and get that phone to ring, I will give it a shot and see what happens. I can however make calls out of the trunk through Asterisk this leads me to believe that it is correct to have FXO ports using fxsks instead of FXS with fxoks. Phil Smith ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Parking - Set ID on return
We have just analog lines coming in to our Asterisk box and so no CallerID information can be gathered, all calls look the same on the phone display. Once a user parks a call and the time runs out it returns the call but keeps the original CallerID information that makes it look like it is just another call from the outside. The operator has to go through the whole company greeting thing again before realizing it was a person who was just parked. Is there a way to set a new CallerID on that returned call so that the operator can skip the intro and go right to asking if they caller would like to go to voicemail instead? Thanks Phil Smith ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZAP - Can't pickup calls on Analog Trunk
We have 4 analog line and 2 analog trunks. On the trunks we have all the DIDs coming into the current phone system. Trying to get everything moved over to Asterisk but having issues picking up the calls on the analog trunk. We can receive calls on the plain analog lines and we can call out on all analog lines and analog trunks. When a call comes in on the trunk line the ZAP channels don't even see anything happening on that channel. Analog lines are channel 1-4 the trunks are on channel 5 and 6. Both cards are Wildcard TDM400. Zapta.conf [channels] usecallerid=no busydetect=yes busycount=6 callerid=Outside Caller 555 echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=-1.5 txgain=0.0 musiconhold=default context=incoming group=1 signalling=fxs_ks channel = 1-6 Zaptel.conf loadzone=us defaultzone=us fxsks=1-6 What are we doing wrong? Phil Smith ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] WiFi Phones
We bought a couple of the UTStarCom phones. They work fine in the office environment where noise is low, but on our production floor it is impossible for me to hear what is being said and the person on the other end of the call also says that they cannot hear a thing from the F1000 when the wireless phone is in a noisy environment. Still looking for a good WiFi phone for production/factory use. --Phil -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vahan Yerkanian Sent: Friday, October 07, 2005 5:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WiFi Phones Andy Hamilton wrote: Anyone have good words to say about any of the WiFi handsets currently available? The UTStarCom F1000 (an 802.11b device) works pretty well. It's about half the $$$ of a Cisco 7920 (which are also pretty nice), but it seems like most of the config is done from the keypad. There is a TFTP option, but it seems that isn't quite perfect. You could check the manual (I programmed the unit without that, except to find that the default password is 88). [snip] The keypad is a touch small, and sometimes I hit the wrong key (and my fingers aren't terribly fat). I also seemed to have a problem transferring calls (using the built in transfer function -- # should still work). Despite many vendors' pages saying that it does 802.1x authentication, it sure looks like WEP is the only available security option. Bought one from VoipSupply too, And yes, it doesn't support 802.1x radius auth (no place to select method, client certificate, etc). I've contacted voipsupply support about this and asked them to remove the 802.1x support listed on the product pages but got a cryptic reply that the phone does support 802.1x MD5.. (md5 is just a method of one of not supported 802.1x auths). Also, the max volume for the headpiece was actually quite low - in noisy environments as on streets you'll have hard time listening to the conversation. Overall, this phone is OK for home and small office use, nothing more. Just my $0.02 in, Vahan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Wengo config and G729(a)
Hello Here you go : [wengo-outgoing] type=peer fromuser= username username= username secret=password host=voip.wengo.fr fromdomain=voip.wengo.fr disallow=all allow=alaw allow=ulaw dtmfmode=inband canreinvite=yes nat=yes insecure=very dtmf=inband context=wengo-outgoing authname= username This is my current working config BUT to have it working, you have to add this entry to your /etc/host file (and reboot your Asterisk config...): 213.91.9.219voip.wengo.fr Hope this help Regards Michel LOPEZ [MVP Exchange] France __ -Message d'origine- De : [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] De la part de Remco Barende Envoyé : lundi 25 juillet 2005 19:45 À : [EMAIL PROTECTED] Objet : Re: [Asterisk-Users] Wengo config and G729(a) One way would do for me, I only use wengo for my outbound calls since they are a lot cheaper than our Royal Dutch KPN :) Which codec did you use and could you post your config lines? Thanks!! Remco On Mon, 25 Jul 2005, Wilson Pickett wrote: Also they switched codecs, now G720a is required to connect. I can only find an (open) G729 codec, is this the same as G729a? I only have it working one-way, no incoming calls. Ironically, when Mark was here we caould have gone to meet them and straighten it out once and for all :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FXO-FXS parameters
Hello, I'm trying to get feed back from other Asterisk users of Welltech WellGate 3701A / 3702A Or Micronet SP5012s / SP5014s Or Immix Tel C3-FXS/FXO Or Euro Teletech VIP-400 (All those are in fact the same product...) Trying to find/share ideas/comments about registrations issue, caller ID issue, Sip or H323, Peer to Peer or Gateway, Voice prompt on/off, Thanks Michel (FWD 627189) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wellgate 3701
Hi everyone I'm trying to setup this Welltech Wellgate 3701 box. If I got to the proxy setup it seems to work but the Pstn incoming call always got a voice prompt from the Wellgate. Going to peer to peer mode seems to be better but I couldn't find any working configuration inside Asterisk. I do not really suffer from the registration problem because I doing all those trials with no password for the 3701 line configuration since I'm in a closed environment. Thanks for any help. Ml ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
