Re: [Asterisk-Users] Looking for quality inbound DID - IAX providers, UK, USA, Australia
Chris, How many you need in the US and UK? I know someone who is working to commit to 2 carriers to get coverage for both US and UK DIDs. I been working on getting DIDs since Aug and it's a rough market with alot of people selling the same suppliers at a wide range of pricing. Feel free to contact me off list =) Otherwise we'll have a commerical spam from every reseller of dids =) Also there is a -biz list for topics such as this at lists.digium.com but that also seems to end up being mostly people plugging their solutations vs feedback based on needs. -- William ___ 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] CallerID Name with IAX Providers
More of a case that in many cases the voip carrier would have to do lookups for CNAM from either their telco or an external CNAM service. These tend to carry an extra cost so that's why it's not wide spread on dids via VOIP. -- William ___ 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] * - SMS w/out PSTN
http://www.bayhamsystems.com/ has a app for sending SMS with asterisk. Don't know how their prices stack up for the UK though. ___ 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] ata vs digium card
They do have the IAXY which could be considered a single port IAX ata ___ 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] Sipura 2000 x dual g729 channels x other choices?
According to the small print in the bottom graphic: http://www.sipura.com/products/spa2100.htm The SPA 2100 would give u 2 ports + 2 RJ45 as well as 2 G729 ___ 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] distribute outbound calls
Groups for each trunk and check the dial plan groupcount and cycle thru the trunks or keep a list of trunks in a DB and just loop thru that first call route 1 second route 2 etc. I'll give it some more thought when I wake up but I think you would have to track concurrent channels per trunk to balance it properly. -- William ___ 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] Stop this I'm trying to help you.(Fwd: Please confirm your message)
Many of these scripts are based on the from which for the most part on this list is whoever posts a reply. When you reply it goes to the list address but the from is infact that of the author of the current message which causes vacation/spam/.. filters to go crazy. For example I just got a mail box full from a member of this list. Odds are this and each new post until they fix it will also cause the same issue. -- William ___ 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] finding current codec?
I guess if you know the channel ID you can get info on the channel and convert the format number to the proper codec. I'd be interested how others have addressed this too. -- William ___ 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] iaxtel
1800,1866,1877,1888 are all toll free numbers in the us ___ 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] Happy Wednesday Morning SMS question, slightly OT
Some commerical SMS gateways can provision a # for routing inbound messages. An example or 2 would be clickatell and ippipi ___ 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] Asterisk Setup Documentation
http://www.asteriskdocs.org is a work in progress document project for Asterisk between that and the wiki you should be ok. If that isn't enough there is plenty of posts in the archives of this list and odds are someone else has already had the issue you are faced with. ___ 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] SMS Gateway
I've used Ipippi.com and clickatell for sms. Clickatell seems to be quite established in the space. Both have APIs that could be used to be intergrated into an app for asterisk ___ 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] Softphone for Linux recommendation
Yes iaxcomm is an IAX softphone. I know Xten is working on improving their linux support for their SIP based shoftphones. ___ 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] No compatible codecs
I've heard problems with the Grandstream G729 and the new digium G729 by MAC ID. Could be a compatibility issue with the implementations. Did you ever use the Grandstream against asterisk with the old Voiceage G729? I've heard that works just fine. -- William ___ 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] Is it possible to ID payphone calls?
The carrier of your toll free should send you indication that it is from a pay phone or not since some do enforce a surcharge to calls originating from a payphone. Probably be best to contact who providers the toll free DID to get proper clarification based on how their system works. -- William ___ 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] Advanced Agents - Need a nice web interface
http://bugs.digium.com/bug_view_page.php?bug_id=0003252 ___ 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] CVS or release?
The stable tree from cvs includes any patches since release that was also commited for the v1-0 tag since some issues were found after the release but not major enough for a new tar ball release. ___ 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] ASTCC vs AreskiCC
Astcc is mysql driven w/ perl based web ui Areski is same concept based on postgres w/ a php frontend also tied in w/ Areski other scripts for reports and such ___ 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] Asterisk Interop w/ Level 3
Seems to be a popular move on this list I'm sure some of those that have taken the plunge already could be of assistance. LiveVoip/Teliax/Netlogic are 3 that I've heard use L3 currently that are on this list. Probably more of a -biz question though then the general user population. -- William ___ 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] ASTCC and NuFone billing is different!!
NuFone service bills in industry standard billing increments, which are: six (6) seconds for the US48, sixty (60) seconds to Mexico and fifteen (15) seconds to the remainder of the world. From: http://www.nufone.net/tac.html ___ 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] Is there a way to # of agents logged into a queue ?
I'd be interested in the patch as well On Thu, 2004-01-22 at 13:51, Bill Hamel wrote: Hi Chris, This sounds what I am looking for, many thanks ! Also, I do not see an attachment, the patch that is :) I dont know if the list strips attachments, perhaps send it to my email address [EMAIL PROTECTED] Thanks again, -bh Quoting C. Maj [EMAIL PROTECTED]: I attached a patch I've been using to show the # of agents (members) and callers on a per queue basis. It adds a new manager command, AgentQueues. It returns on the manager interface the following for each queue: Queue: queuename Agents: # Callers: # There's another manager command, QueueStatus, that might be what your are looking for. There's also Queues but that is a PITA to parse. Fine if you just want to display it in a text widget or something. --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info Pronunciation Guide: Maj == May Fingerprint: 43D6 799C F6CF F920 6623 DC85 C8A3 CFFE F0DE C146 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by The CCIS.net MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. -- This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] asterisk php status viewer
Looks interesting I will check it out and see what I can do with it =) On Sat, 2004-01-31 at 08:17, Brancaleoni Matteo wrote: since I was annoyed this morning, I wrote this simple php script to output channel status from asterisk manager. disclaimer that's very bad written, nor commented... I wrote that just for fun /disclaimer and if someone will use that / improve it , just lemme know. http://asterisk.espia-net.net (wrote with php 4.3.3 and depends on Event: StatusComplete, so a recent * cvs version is needed) ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] IPKall-FWD-Asterisk
Joshua, I've been looking into doing the same for my biz as well. I haven't heard of IPKall and perhaps they aren't setup for what you want to do. If this a vital part of your business I'd consider using a commercial IAX provider to give # a toll free or local # for users to call in. If you want more information on how I do it you can reach me at [EMAIL PROTECTED] -- William Suffill On Tue, 2004-02-03 at 20:47, Joshua Colp wrote: Hi Folks, I recently setup an asterisk system in order to provide a telephone phone system for my web hosting business at a very low expense. My problem is that DTMF tones are not being recognized when calling the IPKall phone number. Calling my server via FWD and IAXTel works out fine however. Has anybody experienced this with the IPKall service? are they not passing the DTMF tones or am I doing something wrong? I've tried switching dtmfmode to all the options, but still nothing. Thanks for your help! - Joshua Colp. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VOIP Deployment Concerns
Before I got into my question for the day I'd like to applaud all the helpful folks and time spent behind the asterisk project to get it where it is today. Great work and between this list, the doc list and the irc channel it's been a pleasure to deal with people willing to help others when and if they can. I will be moving in a few months and I'm concerned as to what kind of bandwidth I would need to work effectively. The reason I posed the question here is simple most of my work is remote SSH to various BSD/Linux machines but a majority of my business calls from the office and clients will be routed through Asterisk. Currently I use a SIP phone to my local * then IAX2 w/ 2 providers mentioned on this list. Unfortunately quality begins to suffer based on the amount of upstream used by the other users here. Comcast only has a 256 upstream on it's basic package. I've read the IAX2 Trunking Comparisions from John Todd. Where I'll be moving will more than likely be installing broadband from Adelphia Cable 256up 3mbit down unless I have a valid reason to require the 512up/4mbit down prem. package at approximately 80 a month. I don't have any personal experience with them since I do live in NJ and will be relocating to FL. Any advise would greatly appreciated. 1 last thing without starting a flame war who on this list sells the Grandstream BudgetTone's. Yes I know there are probably better options but I need to keep costs down for internal and personal deployments where some other options would be overkill. Sincerely, William Suffill ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] iax, trunking, etc.
I use Saww.net and Nufone for IAX2 to PSTN at a per min basis. So if i pushed 5 calls i'd be charge per min for each call. Granted both the companies above cater to * quite heavily. On Wed, 2004-02-04 at 01:40, Chris Clifton wrote: The majority of sip to pstn gateway providers (vonage, voicepulse, and others) appear to be setup for a one line only type of set up. Their web sites seem to be heavily geared for these one line setups. Anyone willing to comment on what type of pricing plans these providers offer when using iax2 trunking or other methods with asterisk to send multiple (and possibly simultaneous) calls through their gateways ? Thanks, Chris Netlabz, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Caller-ID is being sent wrong. How to fix it?
u probably should upgrade to 0.7.2 but as far as the caller id that would be from your sip.conf being set improperly add to your sip.conf callerid=Caller Name # for each sip entry and that should clear it up. On Sat, 2004-02-07 at 00:23, John Fraizer wrote: I'm running Asterisk 0.5.0 and using Cisco 7960 phones in a sip only configuration currently. Everything is working except that caller ID is hosed. Say for example extension 100 calls extension 200. 200 sees 100 as the name but 200 as the number. IE, it gets its own number as the supposed CLID of the calling party. This is flat out wrong. Am I doing something wrong or is Asterisk just terribly broken with respect to sending caller ID information properly? Is this something that only effects Cisco phones? Thanks, John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX Softphone Errors
I've been considering deploying an IAX softphone for some remote users that want to interface with my PBX. It seems as though IAXcomm just prints that it was rejected if they dial an extension unassigned on the PBX. Firefly on the other hand crashes if you dial an extension that doesn't atleast exist in the dialplan. I can't have that of course so I added a catch all group of extentions so if they dial any extension not defined prior it will just play invalid. I was wonder if anyone had any cleaner method to do this other than exten = _X,1,Macro(invalid). I wrote 12 variations to cover all the possible conditions I could think of but a program should crash over such an issue. I get a memory reading error with Firefly if I dial a # not defined in the context that my iax acc is part of. I noticed the Firefly network kept this from becoming an issue by making their dialplan give some feedback on all #'s dialed ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] s/asterisk mailinglists/asterisk forum/g ?
i search them just fine in Evolution. Filters to a different folder than my other mailing lists and works quite well. Different pop3 acc from my isp too =) Why use bandwidth on my colo'd boxes when I can use something I already paid for =) On Sat, 2004-02-07 at 10:30, Eric Wieling wrote: On Sat, 2004-02-07 at 09:06, Roy Sigurd Karlsbakk wrote: now that the lists are at 2-300 email messages a day, perhaps it's time to move it to a web forum instead? This can give us lots of categories (all the apps and channels etc etc), an easily searchable thing such as phpbb and it'll be a lot easier to find the actual info. I don't really think that making the mailing lists totally usable my moving them to a web based forum is the answer. Perhaps instead everyone that thinks the mailing list has too many messages/day to be useful should invest in a few moments to set up some basic filtering in their mail client. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] New Firmware for Grandstream Phones - Supports CFG by MAC address
i saw something about that on the voip-info wiki On Mon, 2004-02-09 at 11:23, Matthew B Marlowe wrote: The newest firmware from grandstream supports configuration by mac address. Simply upload a file cfgmac address.txt Does anyone know the format of a cfg.txt? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Can asterisk make a call to a phone?
use call files there is should a sample in the asterisk src On Mon, 2004-02-09 at 12:21, John Chambers wrote: Newbie question coming up ... Is it possible to use the asterisk to initiate a call to a phone? What I'm trying to determine is ways for software to connect to a phone and send it a sound file with a message like: Hello Mr. Jones. How are you doing today? Press 1 if you're OK. Press 2 if you need help. Or press 3 and start talking, and your message will be passed to a person. The application is probably pretty obvious. I've been digging around in asterisk to see if it can handle this, so far without finding the right docs. I've read a lot on this list about handling incoming calls, which we may want to do eventually, too. But the immediate question I'm trying to answer is how our software can react to events by making calls like the above and doing something useful with them. The immediate goal is play a sound file and then record record reply and/or button presses from the phone. (Speech recognition will come later. ;-) So am I looking in the wrong place? Or is there an example somewhere in asterisk of initiating a call from the computer side to a phone? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] central voicemail with remote offices
Darren,
To achieve voicemail on a central location I did the follow.
In a context I call
exten = _[1-5]XX,1,Macro(stdexten,${EXTEN})
So extensions 100-500 are all routed thru a macro unless previously
defined
macro-stdexten contains:
[macro-stdexten]
exten = s,1,DBget(caller=EXTEN/${ARG1})
exten = s,2,DBget(dnd=DND/${ARG1})
exten = s,3,Voicemail(u${ARG1})
exten = s,4,Hangup
exten = s,102,Macro(invalid)
exten = s,103,Dial(${caller},30,Tti)
exten = s,104,Voicemail(b${ARG1})
exten = s,105,Hangup
In words this is how it works:
User Dials 222
Check Database Family EXTEN Key 222 For an entry
If found store result in caller
Else invalid extension
Assuming the extension is still Valid
It checks the DB again for Family DND Key 222
If that is set it goes straight to voicemail unavailable
otherwise tries to dial Caller
If caller doesn't answer goes to voicemail on central server
Hangs Up
I just used the asterisk DB to pull this off
Some example entries are below
/DND/101 : YES
/EXTEN/10 : SIP/10
/EXTEN/101 : IAX2/wsuff
/EXTEN/500 : IAX2/wsuff
If my iax2 client dials the voicemail extensions on the central server
they can retrieve their messages as well. If you are using Asterisk
servers in the remote offices you could set extensions on the central
server to IAX2/remoteofficea/directuserextension
You could even set an extension on each office pbx to connect to the
office and interface w/ the voicemail
If you need any specific help feel free to contact me off the list.
On Tue, 2004-02-10 at 14:36, Darren Martz wrote:
Thanks for the email William.
I guess the main challenge is to setup the system in a way that's
manageable. I didn't really understand your voicemail notification idea. So
when vmail is left at the central server, you have the server call the
remote office extension and leave a vmail there that they have a message on
the central server?
You know what would be grand, extending the IAX protocol to allow remote
voicemail checks. For example (maybe we can already do this), assigning a
zap channel a mailbox like:
Mailbox=centraloffice/[EMAIL PROTECTED]
So it builds on the virtual domains that are already part of the system but
allows the check to be performed on another asterisk box. For that matter it
would be even better to do the following in an extensions file:
Exten = 700, 1, VoicemailMain(,centraloffice)
Exten = 1234, 1, Dial(Zap/1,15,t)
Exten = 1234, 2, Voicemail([EMAIL PROTECTED],centraloffice)
Exten = ...
Again where centraloffice is identified in the IAX config file. To me this
would allow a separate box to handle all the voicemail calls.
Internally I suppose that would require a transfer to the centralserver and
possibly back again. Maybe someone that has worked closely with the vmail
code can comment?
-Original Message-
From: William Suffill [mailto:[EMAIL PROTECTED]
Sent: Monday, February 09, 2004 9:02 AM
To: Darren Martz
Subject: RE: [Asterisk-Users] central voicemail with remote offices
Darren,
I don't think it's crazy just very involved. I need to write the same idea
into my dial plan since it's moving to a central office tomorrow.
I would be curious to how your dialplan comes out but here's my thoughts on
the issue
In your dial extensions set the dial timeout to 20 then fwd it to an
extension on the central server using IAX2 that will put it into voice mail.
As far as the voice mail situation i'd keep it so all client calls are fwded
by iax2 to the central voicemail system. Once a msg it left you can use a
call file to ring user in office X to notify them. If they don't answer the
central office callback that is set to VM running in that office. That way
when they return they can check their msgs and see they have a notification
that someone left them voicemail in the master office.
Same concept I have to approach as well since I'm a admin for a Webhost as
well as a freelancer. I'm not going to be living on site so VOIP is my only
link to the office incoming phone lines at the office etc.
Anything further feel free to contact me off the least at [EMAIL PROTECTED]
I tend to neglect my other account due to the sheer mailing list volume Hope
that helps, William
On Mon, 2004-02-09 at 11:22, Darren Martz wrote:
Is this a crazy idea? I thought this would be ideal for a failover plan.
Anyone with experience on this?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren
Martz
Sent: Saturday, February 07, 2004 1:32 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] central voicemail with remote offices
I'm having trouble figuring this dialplan out. I have a central
asterisk box that has voicemail and the auto attendant for incoming
calls. Things get complex when I add three remote offices connected only
through the Internet.
The locals seem easy enough to handle. Redirecting
[Asterisk-Users] Looking for Incoming # for Area Code 713 (Houston, TX)
A customer is looking to change to VOIP but he wants a local incoming # where he lives. Anyone know a provider that offers them via SIP/IAX. I'll be running Asterisk to run all the features. Sincerely, William Suffill ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Minimum voice mail message limit?
From Posts on this list on Sat. w/ the subject Voicemail brought to light that there is a patch for some more advanced VM features after a message is left. http://bugs.digium.com/bug_view_page.php?bug_id=156 On Mon, 2004-02-23 at 12:56, Walt Reed wrote: Looking through the Wiki and mailing list, I didn't see an answer to this. Is there a way to set the minimum voice mail message size? Hangups seem to generate 4 to 5 second messages. If I set a min to 6 or 7 that should eliminate most of these. The main voicemail app also seems kind of thin. There are no caller options such as playing back a message you left, deleting it and starting over if you mess up, etc. Voicemailmain also is rather thin - you can't listed to your currently available greetings for example. Is there an alternative voicemail at this time? Patches? FYI, I'm running * from CVS as of Feb 19. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Web based UA
why not load a client on their system they are using? There are quite a few iax soft phones for both linux/win32 On Wed, 2004-02-25 at 13:58, [EMAIL PROTECTED] wrote: You may be right here. I was thinking of an ActiveX plug-in. I don't expect them to use public internet kiosks so they should be able to use the ActiveX approach. I was hoping that something IAX based could be found as it would make the connectivity easier and open port risk reduced. Michael Original Message Subject: Re: [Asterisk-Users] Web based UA From: Jonathan Moore [EMAIL PROTECTED] Date: Wed, February 25, 2004 11:16 am To: [EMAIL PROTECTED] I think xten is supposed to have an active X control version of their softphone that would probably do what you are talking about. On Wed, 25 Feb 2004, Michael Graves wrote: Hello All, Does anyone here have any experience with web based soft clients for *? I'm thinking about putting a page up on our corp web server that would let staff in the field connect to our in-house phone system via the internet. This could help staff making overseas calls while on trips, without demanding that they use a particular laptop/soft phone. They could use an PC on a broadband connection. Thanks, Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] It is dangerous to be correct about matters when the established authories are wrong. - Voltaire ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jonathan Moore Technology Coordinator Winfield Public Schools Office 316-221-5100 Fax 316-221-0508 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Newbie Qu.
are you on a machine that is slow or running alot of stuff? The ongoing answer is the thread that is run by asterisk can't complete it's task fast enough due to lack of system resources so it creates the notice below. On Wed, 2004-02-25 at 20:55, Carl Lougher wrote: When I call Voicemail I get a very slow underwater sounding voice for the first few seconds then it corrects itself. Any idea? Output from Console: -- Executing VoiceMailMain(SIP/2101-20db, ) in new stack -- Playing 'vm-login' (language 'en') Feb 26 14:45:58 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:45:58 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:45:59 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:45:59 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:45:59 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:45:59 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:45:59 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:00 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:00 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:00 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:00 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:00 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:01 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:01 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:01 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:01 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:02 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:02 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:02 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:02 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:02 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:02 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:02 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:03 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:03 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:03 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:03 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:03 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:03 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:03 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:03 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:03 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:04 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:04 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:04 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:04 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:04 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:04 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:04 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:04 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! _ Watch high-quality video with fast playback at MSN Video. Free! http://click.atdmt.com/AVE/go/onm00200365ave/direct/01/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED]
Re: [Asterisk-Users] Asterisk Venture
There are many options for remote support including Digium directly or 3rd party consultants that are on this list On Thu, 2004-02-26 at 10:09, John Benson (Solutios Ltd) wrote: Dear Mark We have a customer who would like an Asterisk server setting up. Do you provide this service, please? I read in a news posting that you could provide remote support? Regards JB __ John Benson Managing Director Solutios Ltd 10 Wilkes Street London E1 6QF United Kingdom Email: [EMAIL PROTECTED] Telephone: +44 (0) 7976 159911 3Video: +44 (0) 7782 309550 Fax: +44 (0) 20 7250 4718 __ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Hotel wake-up
All the digits should already be recorded so you could easily skip that part and play back any digit from the AGI 1-9 that it was assigned. On Sun, 2004-02-29 at 00:03, Robert Lawrence wrote: I would be interested in the AGI Script. As for the voice prompts, I am having Allison record some stuff for me on Monday, including prompts for such a wake up system, that I plan to donate back to the Asterisk community. This is what I have for Allison: Wake up call! This is your requested wake up call! To request a wake-up call, press 1. To confirm a wake-up call, press 2. To cancel a pending wake-up call, press 3. Enter the two digit hour of the wake up call. Enter the two digit minute of the wake up call. Press 1 for A.M. or press 2 for P.M. You have requested a wake-up call for You do not have a scheduled wake-up call. Your wake up call has been canceled. Hours must be between zero one and one two. Minutes must be between zero zero and five nine. Will these prompts be compatible with your script? Robert Rob Fugina wrote: On Sat, Feb 28, 2004 at 08:39:26PM -0500, Bill Michaelson wrote: Anybody know how to implement a hotel wake-up call feature with *? I just wrote an AGI for it. I literally just got it working the day before yesterday, so it's not really 'pretty' yet. I also don't have all of the voice prompts I need, so it's a little rough there, too. I don't have time to go into more detail at the moment, but send me a message directly if you're interested... Rob ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] OTish: Firefly Crashing with *
if u add #'s to your contact list w/ @networknameinyourclient they are connected thru that network such as firefly or others On Sun, 2004-02-29 at 15:05, asdasd wrote: You know what would be nice? If Firefly could have a Network to use assigned to a contact. I.E. I use 800 to check my voicemail at work and call work extensions etc so I have to have IAX as my internal calls...but this means I can't contact people on the firefly network... Kind regards, Matt Riddell ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] OTish: Firefly Crashing with *
don't thank me it's documented in the app just remembered stumbling on it in the network tab. On Sun, 2004-02-29 at 15:46, asdasd wrote: sweet, cheers - Original Message - From: William Suffill [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, February 29, 2004 8:44 PM Subject: Re: [Asterisk-Users] OTish: Firefly Crashing with * if u add #'s to your contact list w/ @networknameinyourclient they are connected thru that network such as firefly or others On Sun, 2004-02-29 at 15:05, asdasd wrote: You know what would be nice? If Firefly could have a Network to use assigned to a contact. I.E. I use 800 to check my voicemail at work and call work extensions etc so I have to have IAX as my internal calls...but this means I can't contact people on the firefly network... Kind regards, Matt Riddell ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] consultative call transfert with mgcp
force all the users to a meetme extension ? On Tue, 2004-03-02 at 11:46, Daniel ANDRE wrote: Hello, I am faced to a problem with call transfert with a MGCP Phone. I use this to make a consultative call transfert: 1. send flash event 2. dial the number and speak with the other person 3. send flash event At this point asterisk tries to make a conference call with the three channels. My phone device doesn't support this. How should I do to make call transfert without trying to build a conference? Best regards, Daniel ANDRE ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Hanging GS101 in a upright position
Take some pics =) On Tue, 2004-03-02 at 21:29, Matthew Marlowe wrote: I've converted it... :) I cut, sanded and crazy glued a plastic notch and made a whole on the handset.. Looks like it came like it. Works perfect. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dkwok Sent: Wednesday, March 03, 2004 7:13 AM To: Asterisk Users Subject: [Asterisk-Users] Hanging GS101 in a upright position Has anyone tried to hang GS101 phones on a wall? It has recess holes at the back of the base where you can hang it on a wall. What it lacks is that the handset is not supported for this upright position. Has anyone done any modification on it? I was thinking about velco the handset. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] control which * pbx to use
line 1 is always default for calls when a line isn't selected prior to dialing. Best bet would just be reverse the order you have them on the Cisco line 1 as primary line 2 as secondary. On Mon, 2004-06-07 at 12:57, Dragan Mickovic wrote: I have a SIP phone (Cisco 7960) registered to 2 * pbx, is there anyway to control which * pbx will be used for making calls? I know by default the cisco will use and I want to change that. thanks micko ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] ztdummy running, but moh meetme don't work
2.4 kernel? I have a RH 9 w/ 2.4 using ztdummy just fine a bit older though. Message seems to show that the phones have trouble reaching each other. Did Sip to Sip between the phones work fine? On Tue, 6 Jul 2004 09:43:18 -0700 (PDT), Jack Turer [EMAIL PROTECTED] wrote: Any thoughts on the following? I am running asterisk from CVS (downloaded yesterday's version, just to be sure) on a test system with no digium cards in it, so I have installed ztdummy (see logs and screenshots below) as a timing source. When I call the music on hold extension from a Sipura Sip connected analog phone, I hear nothing and start getting Warning[98310]: chan_sip.c:674 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Response) As well, I set up a meetme conference, and dial it, the first user (also a Sipura sip phone) gets 'there are no other users on the conference.., which is OK, then a second user comes in, but they are not conferenced anymore. I can hang up both phones, and dial back to the conference, but I won't even hear the 'there are no other users message anymore'. usb-uhci and ztdummy are loaded fine (see lsmod), and this system is running Redhat9 standard install with linux sources. Any thoughts what might be wrong? I have already spent the whole night googling and looking around, so I think I covered all the basics already. I tried to use zaptelrtc as an alternative to ztdummy, but it doesn't compile on redhat9 (log below as well), so that is not an alternative either. Is ztdummy fairly reliable, or does it not work on some motherboard usb chipsets? (this is a compaq deskpro pentium 400mhz) Is there something I need to do with my kernel (recompile?) so that ztdummy works, or anything else. (I suspect the cause is ztdummy, since both MOH and Meetme are broken..) Thank you --- Logs/Listings #service zaptel start Loading zaptel framework: [ OK ] Loading zaptel hardware modules: wcusb Running ztcfg: [ OK ] #modprobe ztdummy --lsmod listing #lsmod Module Size Used byNot tainted soundcore 6116 0 (autoclean) ztdummy 2532 0 (unused) parport_pc 17508 1 (autoclean) lp 8580 0 (autoclean) parport33952 1 (autoclean) [parport_pc lp] iptable_filter 2316 0 (autoclean) (unused) ip_tables 14488 1 [iptable_filter] autofs 12148 0 (autoclean) (unused) e100 56644 1 wcusb 20064 0 (unused) zaptel179840 4 [ztdummy wcusb] keybdev 2720 0 (unused) mousedev5204 0 hid20772 0 (unused) input 5632 0 [keybdev mousedev hid] usb-uhci 24652 0 [ztdummy] usbcore73088 1 [wcusb hid usb-uhci] ext3 64704 2 jbd47828 2 [ext3] --extensions.conf (relavent part) ;dial 500 to join the conference (doesn't work though) exten=500,1,Answer exten=500,2,MeetMe(1234) ... ;dial 6000 to hear music on hold (doesn't work though) exten = 6000,1,Answer exten = 6000,2,MusicOnHold,default --Meetme.conf [rooms] ; ; Usage is conf = confno[,pin] ; conf = 1234 --musiconhold.conf [classes] default = quietmp3:/var/lib/asterisk/mohmp3 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] VoIP hackers gut Caller ID
Just asking for abuse though unless it is restricted or grounds for termination without a refund, People prefer to set their CID to a proper call back number such as myself but it has can be used for less positive uses. On Wed, 07 Jul 2004 11:45:48 -0400, Jeremy McNamara [EMAIL PROTECTED] wrote: Chris Foster wrote: The Register is carrying a article written by Kevin Poulsen of Securtiy Focus, calling asterisk ..the most powerful tool for manipulating and accessing CPN data.. http://www.theregister.co.uk/2004/07/07/hackers_gut_voip/ I hope NuFone doesn't drop asterisk-set-able callerid's after this article; i've been wanting that feature from voicepluse for a long time. Then NuFone customers better not abuse this power. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] New PBX Help
Even to interface analog lines with asterisk you'd need hardware too which perhaps will put it out of the reach of your small organization. $100 for a x100p (a analog port for asterisk) On Wed, 07 Jul 2004 12:27:38 -0400, Mike Wagner [EMAIL PROTECTED] wrote: That's all extremely way over my head. I have no pbx knowledge at all... and we're a small organization, so we can't afford to buy the modem cards just to test it out. Guess I'm going to have to do some reading. I don't want a VOIP based solution. We'd like to get numbers through the phone company, and use Asterisk as a standard pbx. -MW Andrew Thompson wrote: Mike Wagner wrote: Is there any reccomendations as to how I might set this up??? Keep in mind that I know next to nothing about pbx's and phone systems. What is your asterisk knowledge level? Have you set it up in your home/office? Have you fiddled with meetme, call parking, call transfer, DISA? (your users WILL want some if not all of these) Have you connected your box to FWD, IAXTEL? Have you made outbound voip calls through voicepulse, nufone, iconnecthere or some other provider? I think once you've done that, you'll be ready to ask better questions. The first few that come to mind are soft versus hard phones, T1 or ISDN or not, channel bank or not, etc. - Andrew Thompson http://aktzero.com/ http://www.retirequickly.com/43653 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] multiple days on a GotoIfTime command?
well then lever it db driven and set the #'s in the db and update that to the proper call order as needed On Wed, 07 Jul 2004 13:51:10 -0300, Gelson Dias Santos [EMAIL PROTECTED] wrote: The problem is, there is no pattern. It´s not an open/close scenario. This month I need to call NUMBER1, NUMBER2 and NUMBER3 on those days. Next month, who knows? I´ll receive another schedule to implement on asterisk. I see no way to avoid changing those lines each month. What I´m trying to do is reduce the number os files involved. Gelson brian wrote: I see the pattern.. let me think for a second.. and I'm sure I can get you something that's simpler than 31 gotoif's bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of brian Sent: Tuesday, July 06, 2004 5:24 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] multiple days on a GotoIfTime command? You're making this WAY too complicated its simpler than you can even imagine. Mind answering my original question first? WHAT THE HECK is the pattern your logic? What times are you open.. what times are you closed? What? bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Roger Gulbranson Sent: Tuesday, July 06, 2004 4:20 PM To: [EMAIL PROTECTED] Cc: Roger Gulbranson Subject: Re: [Asterisk-Users] multiple days on a GotoIfTime command? On Tue, 2004-07-06 at 17:03, Gelson Dias Santos wrote: brian wrote: What are you trying to do? What is the end result and what hours are you open? Exactly what I said. Need to call a number if time and day matches what is on the rule. This month I have to: call NUMBER1 if day = 1,2,3,4,5,8,14,17,18,20,23,26,29 call NUMBER2 if day = 6,9,10,11,12,15,21,27,30,31 call NUMBER3 if day = 7,13,16,19,22,24,25,28 I have it working now using 31 GotoIfTime lines, one for each day of month but I would like to optimize it. If I could group all days related to a number somehow, I would end up with just three GotoIfTime lines. You are making this way too complicated. Use DBget to retrieve a number which is the extension you want and then dial that extension. Have a cron job (or something similar) set the extension you want via DBset. You can put all of your time logic into the cron job. There may be even simpler solutions. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] feature - VM gain adjust?
Normalize for Linux can tell you the levels of a wav and can be used to adjust it according. Been toying with using it for some of my streaming media clients since it sucks to go from too low and having to up the volume to very loud. On Mon, 12 Jul 2004 10:31:08 -0400, Seth Remington [EMAIL PROTECTED] wrote: What about a post processor that performs Compression/Normalization on the recorded voice mail file? On the down side I can see this being a big CPU hog if you are handling a huge amount of calls and trying to normalize a 5 minute long voicemail at the same time. On the upside you don't have to concern yourself determining line loss or similar things. You also wouldn't have to worry about what I call the Seinfeld Syndrome: quit talker / loud talker issues. You would just have two new variables in voicemail.conf - normalization=yes or no and another to set the db value. -Seth On Mon, 2004-07-12 at 08:46, Rich Adamson wrote: Are you suggesting such a thing exists, or that that would be a proposed future application? I propose to think if an AGC / dynamic compressor could be used instead of a config variable. Most sound editors have modules for this. So how would you detect the remote caller is 14.7 db away from * and adjust the 'outbound' voice message to be at some higher audio level? I like the AGC approach, but I'm not sure its realistic in terms of consistently being able to identify the transmission loss from each and every vm call. Since we know what the loss is for each pstn line (to the central office), it would appear that static value would be a good starting point and the user could adjust from there. Much easier (and more likely) to implement. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] RE: [Asterisk-User] asterisk compile problem
Using bison 1.35 here - Original Message - From: Fletcher Bonds [EMAIL PROTECTED] Date: Wed, 14 Jul 2004 09:09:48 -0700 Subject: [Asterisk-Users] RE: [Asterisk-User] asterisk compile problem To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] From: Nik Martin [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] asterisk compile problem Date: Wed, 14 Jul 2004 09:22:38 -0500 Organization: Radiance Technologies, Inc. Reply-To: [EMAIL PROTECTED] Fletcher Bonds wrote: Hello all As of 5pm PST today (7/13), I pulled Asterisk down off of cvs.digium.com:/usr/cvsroot and tried to compile it on Linux ES 2.1 Actually, I pulled down zaptel, libri asterisk and compiled them in that order as per my install guide. When I try to compile asterisk with make clean; make install, it runs okay for a bit and then I get the following error: (ignore Outlooks insistence at capitalizing the first letters of these lines/sentences - it's all lowercase) Bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c Ast_expr.y:110: unrecognized: %locations Ast_expr.y:110: Skipping to next % Ast_expr.y:141: invalid @-construct Ast_expr.y:141: $. Is invalid [these last two lines repeat iterating the line number (141) up to 155 then:] Make: *** [ast_expr.c] Error 1 And it stops. I've looked at this source file starting at the 110 line location, (I'm not a C programmer though) and I don't see anything obviously wrong to fix. Additional info on my system: This is a fresh install of Linux ES 2.1 on a HP ProLiant DL380 - It's custom install with Development Kernel Development packages installed as well as OpenSSL-Devel, Readline41, Ncurses4, Ncurses C++ Devel, SOX mpg123 packages. Other than that it's completely clean. It's being installed on a partition with loads of space available to it and the install is being run as root. Can anyone tell from that error if I'm missing something or what the problem may be? Thanks a bunch Yep, you need bison I have bison. # bison V GNU Bison version 1.28 Is it expecting a different version than that? Thanks! Fletcher Bonds Operations Software Tester TeleCommunication Systems, Inc. (TCS) Enabling Convergent Technologies www.telecomsys.com [EMAIL PROTECTED] office: 206-792-2366 cell: 425-736-7993 fax: 206-792-2001 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] CISCO 7960G FIRMWARE
You need a cisco smartnet license to legally download the firmwares for the phone. This would include the sip firemware On Wed, 14 Jul 2004 20:26:27 +0200, xfastjackx [EMAIL PROTECTED] wrote: Hi everybody, I will receive my CISCO 7960G tomorrow. I've ordered it as a global spare without any callmanager licence. Now I don't know if I can get firmware-updates so could please someone send me the SIP-firmware? Is the default firmware the skinny one? Wich would be better to use with asterisk? Thank you very much ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] Where can i get an UK SIP account with UK number?
voiptalk.co.uk On Wed, 14 Jul 2004 16:36:51 -0700, Dameon D. Welch-Abernathy [EMAIL PROTECTED] wrote: On Wed, 2004-07-14 at 11:41, Johannes van Hulst wrote: Can somebody help me with some names of good UK SIP providers? I am looking for a UK number to connect to my asterisk server. www.gossiptel.com provides UK numbers www.iconnecthere.com provides UK numbers There are probably others. -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Small setup
i use a p2 400 here and it has problems with the scheduling but for 1 or 2 calls that would be ok. Depending on the volume you expect at 1 time adress the hardware according. I'd suggest atleast a 1ghz or so On Thu, 15 Jul 2004 08:11:43 +0100, Simon Chappell [EMAIL PROTECTED] wrote: Hello All, I have a very small setup of 4 users and a X100P. Asterisk is currently running on an Athlon 1800 but the server it is running on is also our imap/web/mail/development/samba server and we are having a few issues with asterisk which I believe is down to to many tasks. What I intend to do is build a box just for asterisk(Ill call the old one obelix). I have a few machines in the loft but wandered if anyone could give me an idea as to mimimum spec for such a small installation. Thanks in advance Simon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] spa-3000 review?
Seems quite interesting. Any suggestions of where to order one and about how much? On 15 Jul 2004 16:54:03 -0700, Wolfgang S. Rupprecht [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] (Tom Neville) writes: ; FXO port - Line from our office PBX. [40] ... secret=NOPE Have you gotten asterisk to work for dial-out to the PSTN when using a md5 authentication? I can only dial out when I tell the SPA-3000 to use no authentication. Eg: admin-PSTN Line-VoIP Caller Auth Method-None Changing it to the following doesn't work (adapting the example to use your values from above): VoIP Caller Auth Method: HTTP Digest(their name for MD5 digest) ... VoIP User 1 Auth ID: 40 VoIP User 1 Password: NOPE Turning on sysloging on the sipura wasn't informative at all. (All I got was a bunch of lines like this: Jul 14 16:42:11 hsephone [1:5061]64.142.50.224:5060 Jul 14 16:42:11 hsephone [1:5061]64.142.50.224:5060 Jul 14 16:42:11 hsephone Jul 14 16:42:11 hsephone Jul 14 16:42:11 hsephone [1:5061]-64.142.50.224:5060 Jul 14 16:42:11 hsephone [1:5061]-64.142.50.224:5060 Etherdump also showed quite a few invalid syslog lines coming from the sipura. Mostly they were missing the local0.debug. Some went to local2. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Analog ports via USB
The ipo11's were 25 each when I ordered them + import costs since it comes from TW. Yet to use them w/ asterisk but it worked fine w/ their supplied software in windows since they are Tigerjet based adapters. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] VoIP Termination
1 port so easier w/ nat + it can trunk(lowering overhead) for multiple calls to 1 provider. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] New PRI with DID in US?
quickest would be pattern matching and just make the reoccuring patern of #'s so you don't have to list em one at a time. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] My Boss wants background music!!!!
We are looking at the Polycom IP300 or the Sipura SPA-841 for low end type client needs at this point. We didn't feel comfortable with the GS to our type of customers but if it fits your needs that's an option as well. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] SMS - how to send one
between asterisk boxes and fixed line SMS I believe but never was 100% sure on this either. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Per extension/user CDR?
If each account has an account code it should spawn off a CSV CDR or you can just do a mass select from SQL by account code. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Per extension/user CDR?
Should be an account code field in the DB table that can be used in queries to just pull 1 accounts records ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] OT- Callwave neat app
7. How Much Does It Cost? Sign up today for a RISK-FREE 30-day trial of CallWave! Keep it, and you'll pay a special, introductory rate of only $3.95 per month. Cancel any time before your trial ends and you pay nothing. Hmm seems they aren't exactly sure what to expect. TOS didn't seem to have any usage clauses but it's only an introductory rate so when it catches on they will hike the price. =/ I agree it could probably be implimented with Asterisk too =) -- William ___ 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] One SIP peer use 2 diff codecs?
Give the FAX SIP device a different account and force it to Ulaw. For example if the user was account you could create F for fax and V for voice and have sperate allow/deny codecs ___ 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] cron job to reboot GS101
curl could also be used. Since people asked I'm going to write it up tonight since I use a GS as well until my Cisco shows up. On Sat, 2004-04-03 at 09:52, Duane wrote: Walker Haddock wrote: I know that you can reboot the GS phones by hitting the rs.htm URL on the phone. But, you have to log in to the web interface before doing this. lynx has username/password options, well unless they used sessions... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: [Asterisk-Users] softphone (SIP) with multiple profiles
Would it be possible to use an IAX softphone in your situation? I know iaxcomm is available for both Windows and Linux and can handle multiple accounts. On Tue, 2004-04-06 at 10:26, WipeOut wrote: Martin Mielke wrote: Hi Markus, Markus Miertschink wrote: The one I know of is X-Pro/X-Lite from http://www.xten.com/ I doubt that there is a Linux version available... Markus I contacted X-Ten and they told me they are working on a Linux version of X-Lite... let's see... Martin They have been working on it for the last year.. I havent even seen a beta.. :( ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: [Asterisk-Users] softphone (SIP) with multiple profiles
I see my solution was to put a * server in at home and have that link to my office over IAX and 3 remote terminations over IAX. Same could be done in your case using IAX/SIP/H323 client to a local server and register with your other providers from * On Tue, 2004-04-06 at 11:26, Martin Mielke wrote: William Suffill wrote: Would it be possible to use an IAX softphone in your situation? I know iaxcomm is available for both Windows and Linux and can handle multiple accounts. yes, iaxComm works for both Linux and Windows, but the sound quality is poor compared to SIP softphones such as SJphone or Kphone (always on Linux)... I do need a SIP-capable softphone at home because some other VoIP providers don't support IAX... :-/ Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Presence
They modified iax to include the presence packet but only works on their customized firefly network. I was thinking along the lines of a software app for those of us who use hardware phones but still want to keep TXT chat and presence and perhaps integrated into 1 of the iax soft phones as well to provide a full solution. On Wed, 2004-04-07 at 20:40, Duane wrote: Shad Mortazavi wrote: I think integration/gateway between Asterisk and Jabber would be a amazingly wonderful product. firefly, while not 100% bug free I think it has this feature, although I haven't played with it enough to work out how to show someone as being online... ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Presence
I'm not familiar with the protocol used in Firefly. If that was known then it would be possible to add the functionality to * so anyone can have the simple presences by dialing extensions in their dial plan or crafted packets at a software level. Jabber is already deployed in my organization so I would lean toward integration to that standard as well. On Wed, 2004-04-07 at 21:05, Duane wrote: William Suffill wrote: They modified iax to include the presence packet but only works on their customized firefly network. I was thinking along the lines of a software app for those of us who use hardware phones but still want to keep TXT chat and presence and perhaps integrated into 1 of the iax soft phones as well to provide a full solution. Question is then, how well does their system work? Already have an IAX2 compatible soft phone with that stuff in it, why not make use of the fact and just work out what needs to be sent to their client... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cell Phone, *, Portability
Currently the plan is to forward all PSTN calls on our 2 incoming PSTN lines 2 remote toll free's via IAX2 to staff. 3 different delivery methods 1) Users local to the office where the lines come in with GS/PSTN phones 2) IAX2 to remote location * server then Cisco 7960 on that lan 3) IAX2 to remote softphone That assumes they are in the office though. Recently I've been out more than I've been in my office so a VOIP wasn't an option. In this case calls went to me cell and ran up quite alot of minutes. I was wondering how others handle this. Also what carrier you suggest for 2 business cell phones? -- William Suffill ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] External access to voicemail
in the context of the incoming DID assuming their Caller ID is equal to
the mailbox for their voicemail aka DID #
exten * = 1,VoicemailMain(${CALLERIDNUM})
You might want to improve this though like so:
Add all assigned DIDs to an Asterisk DB
On * check if callerid is a valid did u assigned
if yes then VoicemailMain(${CALLERIDNUM})
else VoicemailMain()
This would cause them just get get a password prompt for their VM
if they call from a caller id matching the mailbox and press *
If the callerid doesn't match then call Voicemailmain without an arg
then they will get a prompt for the voicemail exten then password so
they can enter their did manually before the password prompt.
On Thu, 2004-04-08 at 14:41, Steven Kokinos wrote:
in my setup i have several users with DID lines coming in from various
sip/iax providers. within our old phone system, a user could call
their own DID line, then hit the * key when they hear their voicemail
greeting and be prompted for their password.
is there any way this could be replicated within asterisk? i'm having
trouble figuring it out since it steps through things sequentially,
whereas i want to scan for input during the playback.
any help would be greatly appreciated.
regards,
-steve
___
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Re: [Asterisk-Users] BGM Music
Thinking about it further you could set the 6th line to autoanswer and have the pbx call you and play MOH when none of your lines on the asterisk box are in use. On Thu, 2004-05-13 at 10:57, Joseph wrote: Is there any way to play background music on a sip phone while the phone is not in use like many legacy pbx's offer? Could you take 7960 and use the 6th line in a similar fashion to the all setup maybe? Thoughts ideas? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Can asterisk be programmed to make alarm calls?
Sure you could even use the examples posted here and the wiki to use the outgoing spool to make calls. Just use a crontab to place a call file in the outgoing spool every x # of days and problem should be solved. On Thu, 2004-05-13 at 14:41, Mark Phillips wrote: Those of you whom have a free Washington State phone number from ipkall.om will know that one has to use the number at least every 30 days or else the number becomes disconnected. I have 3 numbers pointed at my asterisk my which work very well but I still had the 30 day problem. Is there a way that I can program asterisk to make a call to my WA numbers so that they wont get disco'd? I'm thinking of something like a alrm call that one has in a hotel room. YOu pick up the phone and program a ring back time. Hope this make sense. Thanks G7LTT/KC2ENI Mark Phillips ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] *** Asterisk sunday news: Read the sampleconfigs, Luke!
Billy, Attachment seems to be due to a GNUPG sig file -- William On Sun, 2004-05-09 at 12:00, Billy Huddleston wrote: Mark, Would you please re-config or use a different mail client as to not send your replies back as attachments?? It's VERY kludgy, and, I'm just going to stop reading them.. along with all the other folks.. Thanks, Billy - Original Message - From: Mark Elkins [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, May 09, 2004 8:41 AM Subject: Re: [Asterisk-Users] *** Asterisk sunday news: Read the sampleconfigs, Luke! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] blocked caller id
check the caller id in your incoming extension before you pass to to a end user. Reset $calleridname to unavaliable if no number is given On Tue, 2004-05-18 at 15:18, Roger wrote: I have a question - if a user calls up w/ blocked caller id I get the following on my phone Incoming call from asterisk This is the same on my Cisco 7940s and Polycom phones. For average users this is not intuitive at all.. I'd like to configure this so if I deploy this at a customer site it says caller id unavialable. With the spelling done right Any ideas on how this wold be accomplished? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Conference Server
ztdummy will suffice. A Zaptel interface is used as a timing device for the conference. On Thu, 2004-05-27 at 11:58, pesb wrote: Hi there, I need to implement a SIP Conference Server. I've saw that asterisk has an application called meetme. But, it says that A ZAPTEL INTERFACE MUST BE INSTALLED FOR CONFERENCING FUNCTIONALITY. Is there any other way to implement a conference server without the need of having a ZAPTEL Interface? I need my conference server to work only with my SIP Phones. thanks in advance, Pablo Salinas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Re: FireFly doesn't work with 3rd party anymore
I just downloaded it today and the config menus just have for Firefly no SIP or IAX2 On Thu, 2004-05-27 at 12:14, Tony Mountifield wrote: In article [EMAIL PROTECTED], brian [EMAIL PROTECTED] wrote: Just an FYI FireFly no longer works with anything but the FireFly network. No more SIP, No more IAX. It was a damn good IAX client... too bad its crap now. Are you sure? http://www.virbiage.com/firefly/download/ still says the following: Standalone SIP / IAX mode: If you want to use Firefly on our Firefly phone network (with your own voicemail etc.) then you will need to register a phone number. However, you can also use Firefly as a SIP or IAX client on your own network. Cheers Tony ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Asterisk as an outbound call machine?
I wouldn't trust it to do any real detection. I use the press # mod in 6 sec mod to be able to fwd to other phone #s without risking hitting the answering machine or wrong person. I don't believe there is any real way to detect what you are after as far as if the call is picked up. You would get status for busy and such though. -- William ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] zaptelrtc for 2.6.x
why not use ztdummy which doesn't require USB on 2.6.x? Uncomment it in the zaptel make file and away you go =) ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Astricon pictures
their permission might be a good idea too =) Don't want anyone to get hostile when you show the pics to the community. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Astricon pictures
Good idea Matt. Tad far for you unfortunately and too costly for me at this time but hearing all the latest and greatest news would be supper. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Status of conference calls at Astricon ?
the dev conf is friday from 9am - 4pm EST as far as i know Any more info would be cool. I think an outline of the topics are on astericon's site ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Asterisk 1.0 released
If anyone who got the 1.0 tar's would be able to get them to me I'd be more than willing to donate traffic toward the effort by mirroring it on some bandwidth. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] 1.0 Mirrors
Glad it was mirrored. I will contribute a mirror as well when I return to the office. No reason Nacs should be the only one taking the burdon. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] 1.0 Mirrors
Probably should just create a page like SF that would round robin the HTTP links and as 1's are removed and added the users wouldn't need to find a different url. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Billing Fun - anybody know where to get a NPA/NXX db?
There used to be an NPA NXX sql on 1 of the asterisk site's. http://www.fnords.org/~eric/asterisk/ I doubt you will find a nice complete 1 for free unless you parse the npana data yourself which you could do. I did it recently not exactly fun. Still might not be 100% though. -- William ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Thank you Mr. Mark Spencer and Asterisk Community Members
Agreed. It's a big accomplishment and wouldn't be possible with Mark/Digium starting it as well as those of the community that give whatever time they can besides their normal jobs to help other users. We all started at the beginning one time or another why not give back where we can to help those just starting out and to move the project as a whole forward. --- William ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] GSM phones, bluetooth and general happiness
Interesting. I think either the phonelabs adapter or cellsocket might be an interesting idea. We are moving to a biz mobile package I use iax2 term to fwd to a nextel since it's free inbound but having a cell on the asterisk box is probably a better fit. Besides on a biz plan w/ tmobile and others you can add a line for $10 on the pooled mins plans. Very interesting idea ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Asterisk 1.0 released
Cirelle did you delete the .version file in the src tree on your box? I doubt cvs is 2 wks behind since I got cvs commit emails this morning. I believe make update will remove the .verision for you too which will fix that issue. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Linksys PAP2-NA
Anyone here have any pointers of where to get 1 of the PAP2-NA. Given all the talk about it I'd be curious as to testing one myself . -- William ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Looking for a commercial version of an IAX2 Softphone
Depending on your needs I don't know if you will find 1 that used IAX2 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Looking for a commercial version of an IAX2 Softphone
Sorry about that cut off . Like I was saying I'm not sure if you will find once advanced enough using IAX2 currently. Firefly was the most evolved when I too was looking but their oem terms weren't exactly what I wanted to spend given the fact that I probably would be going hardphones eventually. Depending on your need IAXPhone isn't bad for windows. Iaxcomm is my preference for cross platform. Perhaps it will take modifying an open source client and adding new features for this area to progress. -- William ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Limiting use of an account
In short yes. You put users in a context and only allow certain features in that context. As far as the limit you probably wish to write an agi or app to handle the tracking of the mins used per day and disconnect the user in need be. It could be all done in extensions with dbput and dbget or sqlite too if you wish. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] how can I test canreinvite effectivness?
Ntop.org probably could fit you needs from the console. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Cheap, Highquality IP Phones
Ya good question. Looks like a nice phone with 2 lines for $100. Maybe one of the places that carries sipura stuff will get them in and start pushing them. It says they should be available to the public in Nov. I guess we just wait and see. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Wonderful Success with PAP2-NA
Hooked a 4-line vtech phone up to 2 PAP2-NAs and basically had created a 4 line ATA for $100. 2 ATA's w/ 2 Ports each I think. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Cheap hosted servers and Asterisk
Scott, I use an AMD 2400 hosted in The Planet (www.theplanet.com) to host my asterisk box currently. They don't directly offer AMDs but a provider that colocates there does. $60/mnth. SeverMatrix.com is the low end dedicated biz of The Planet directly. It is only 60ms from my home in NJ even in TX and I have all my voip routes into that. I use notransfer and G729 for most routes and been fine for the most part. Cisco 7960 here to TX via sip and in/out for origination/term by SIP or IAX2. It is a nice change since my system is reachable even when my cable decided to take a hiatus which is not unheard of with Comcast. I also configured it to forward calls to my cell phone if my VOIP extension isn't available which is nice when I'm out or Inet is down. Sure it costs me the mins addition for that leg but I preferred that over not getting the call at all. I would suggest looking around and finding one with good routing to your DSL But there isn't a shortage of providers that offer low end dedicated. Any specific questions feel free to contact me off list. -- William ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] MusicOnHold() - how to restart player from the beginning on each call? (fwd)
Why not just create a context that plays static msgs whenever someone is transfered thereThank you for calling Monthly special etc ... then transfer them back when the person at the biz picks up On Sun, 24 Oct 2004 14:23:04 -0400, Emilio Panighetti [EMAIL PROTECTED] wrote: Looks like what you want is not music on-hold, but rather a streaming server On Oct 22, 2004, at 4:23 PM, Ryan Courtnage wrote: On Fri, 2004-22-10 at 16:05 -0400, Kanwar Ranbir Sandhu wrote: On Fri, 2004-10-22 at 05:56, Manfred Petz wrote: [snip] Is there a way to force MusicOnHold() to be restarted from the beginning for each call which has been answered? [snip] Why? What would be the point? off the top of my head ... promotional messages. Manfred - I don't think there is a graceful way to do this. I know that if you do a killall mpg123 at your command line, the next call MOH answers will start playing the mp3s at the beginning. Of course this would affect others that are listening, but if you build out some logic you might be able to make some use of it. Ryan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] SIP Conferencing Server
Wouldn't http://www.areski.net/asterisk-meetme/about.php?s=0 already provider the webbased/db frontend to manage something like the above request? I haven't used it myself but I came across it when looking for other asterisk related scripts. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] [PATCH] DUNDi for 1.0.2
Great job Jeff. Lets hope the dbscret can be patched up soon too but this is a great leap forward. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Is NuFone messing up for anybody else?
Could be a case of routing from you to them and the various links inbetween. Hard to really pinpoint given the numerous factors that could cause such issues ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] ATCC - Astcc-Admin.cgi File
Sounds more like a requirement for custom development since I'm sure your needs will vary from some others that are also using astcc as a starting point for their prepaid cards -- William ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Grandstream BT100 - Does not recognize DTMF
What codec and signalling is being used? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
