Re: [Asterisk-Users] IPComms Setup
it is trying to match the did in your context which it can't do ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] soft phones for Zaurus PDA
Ziaxphone might fit your needs. http://www.kauss.org/Stephan/ziaxphone/ Haven't used it recently since someone broke the screen on my Zaurus =( -- William ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pause during dialing to enter another number
I'd suggest Dial(trunk/1800555,30,D(1wwww2) That will cause it to dial that DMTF string on connect and w causes a pause. I haven't tested it just referenced http://www.voip-info.org/wiki-Asterisk+cmd+Dial and http://lists.digium.com/pipermail/asterisk-users/2004-November/071853.htmlOn 8/22/05, Joseph [EMAIL PROTECTED] wrote: I need to execute account number, device number after dialing mainnumber; what is the best solution?Is it possible to pause during dialing.Dial 1-800-numberpress 1 for Englishwait 5sec enter device number wait 5sec enter device IDWhat are my best options?--#Joseph___Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] Bigger problems than ogg
rm -rf /usr/lib/asterisk/modules/ ___ 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] Bigger problems than ogg
You can also remove /etc/asterisk to erase the configs that were installed but the major issue between STABLE/HEAD is the modules. The version mismatch in the modules is what caused all the errors you got such as Aug 14 15:04:33 WARNING[4860]: /usr/lib/asterisk/modules/app_realtime.so: undefined symbol: ast_register_file_version ___ 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] 7960 TFTP
http://www.ohse.de/uwe/software/utftpd.html worked fine for me. ___ 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] Hop-On WIFI Phone MSRP $40
Unfortunately no. Someone say the press release and bugged me about it as well but I haven't seen anything that would indicate they plan on doing anything more than parting with carriers with large rollouts of these phones. That MSRP seems too good to be reality 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] Cepstral partnership with Digium
You will be able to purchase Cepstral voices from Digium just like you dor for G729 already. I would guess it's 1 way to show the power of asterisk by putting all the TTS orders thru a company such as Digium. ___ 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] Junction Networks
Which of their services are you refering to? Conference Bridge worked fine in my tests but I haven't used them for anything else to date. ___ 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] Random Blip
What codec are you using to asterisk and what codec to VPC? Also does this occur if you test the service with another ITSP (nufone/voipjet/teliax) ___ 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] Public vs. Private Network
Point to Point connectivity if they are close enough. Only use DSL/Cable if you have to since results may vary depnding on location/route/utilization/ISP. On 5/19/05, Andrew Latham [EMAIL PROTECTED] wrote: yes On 5/19/05, David Sampson [EMAIL PROTECTED] wrote: Hello I am looking at connecting 7 10 locations together using Asterisk and possibly some VoIP gateway appliances. I need to insure best voice quality as these trunks will be used primarily for customer calls. I am considering implementing a full T1 frame relay circuit to each location which can be done for a reasonable cost. DSL and Cable are currently at each location and setup for automatic failover. Should I remove one of my public connections and replace it with a private circuit for best quality? Thank you, Dave ___ 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 -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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] VoipSupply.com
I concur with Ed. The web orders get put in a massive queue and are harder to follow up on. When you use a rep then they are there to help you with your sales concerns so you use them for your other needs they can fill. -- 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] voipjet anyone?
I'm #11 but I have notice of late a few problems but nothing major given the price differences assuming you don't have the volume to commit to another carrier directly for the destinations you are after. -- 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] VoiceXML
FWD sends all the 411 calls to TellMe.com which also provides professional VoiceXML services and development resources (studio.tellme.com) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Put a wait in a .call file.
or create a file in another dir. Change the time on the file then put it in the call spool. It should be covered on the WIKI as well. Or you could write your own app to use the manager api to originate the calls depending on the needs you have. ___ 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] g729 license
Yes same provess you did to register the license in the first place. You can rereg the license I think 3 times or so before you have to call Digum and have them manually change what your license is tied to. ___ 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] YAC and IPs
Why not DBGet from the SIP.Registry in ASTDB? Wouldn't that be cleaner since it would only return the 1 you want instead of parsing what could be a load of sip peers? -- 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] Broadvoice Down?
Seems what we all want but since it's new there are always problems especially since we as a whole complain when they charge too much. There will be a happy medium eventually but for now it's probably best not to having too important dependent on voip origination since unlike termination you can't just use whoever it working properly at the current time. Atleast for me I've found giving out my cell # and doing all call backs over voip as been the safest bet for now until voip origination improves. ___ 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] Has anyone used Libretel DIDs with Asterisk?
FWD is availabe via iax as well as sip. Easiest solution would be to sign up for a FWD acc and enable iax. You could even use sip plenty of examples between the list and voip-info.org that should get you going. ___ 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 touch voicemail on Cisco 7940/60
I prefer to use a numerical exten. but same result. ___ 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] 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] * - 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] 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] 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] 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] 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 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] 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] 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] 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] 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] 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] 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] 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] 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] 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] 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] 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] 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] 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] 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] 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] 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] Deploying multiple Sipura 3000s with Asterisk
Bonus: Sipura SPA-3000s purchased from Voxilla include the following: * One free month, with all activation fees waived, of any Broadvoice's unlimited plan, including Unlimited World Plus; * Up to 100 free calling minutes through iConnectHere; * One free month, with activation fee waived, of VoicePulse unlimited US-48 calling (US residents only, see Terms and Conditions ); * Access to Voxilla's Sipura user group forums for technical support. ___ 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
Re: [Asterisk-Users] Looking for a SQL or ODBC Application
there should be 1 addons for mysql and anthm wrote res_sqlite which would add the same functionality but use sqlite to backend it ___ 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] [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] 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] 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] 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] 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, 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] 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] 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] 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] 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] 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] 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] 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] OT: For Sale Cisco 7960 7905 IP Phones
Could you read the post and reply off the list like it was requested? I agree that the -biz list is a better place for it as well though. ___ 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
someone pack a wrt54gs and create your own wifi =) On Fri, 17 Sep 2004 16:12:00 -0500, Kristian Kielhofner [EMAIL PROTECTED] wrote: Michael Welter wrote: Does anyone know if the Marriott has Wi-Fi? LAN connection in the room? Mike ___ 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 Mike, It sure does. On the astricon.net site you will find High-speed internet access provided by STSN under hotel features. I know from experience in other Marriots that there is an adaptor in the room (an xDSL I'm sure) with a 10/100 ethernet on it. You register when you connect for $9.99 per 24hr span to connect. I am not sure about Wifi. It would be bad if the hotel for Astricon didn't let us use our SIP phones! -- Kristian Kielhofner ___ 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] Auto Dial With An Extension number?
Dial has the D flag for doing just that Not sure how you would do it for the call spool though ___ 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 on Fedora Core 2
ztdummy should be able to work natively on the 2.6 kernel w/o need of usb. I use it on a fc2 box single processor w/ a 2.6 kerenl w/o issue and a rh 9 w/ 2.4 w/ usb - Original Message - From: Chad Brown [EMAIL PROTECTED] Date: Wed, 15 Sep 2004 18:59:38 -0700 Subject: RE: [Asterisk-Users] ztdummy on Fedora Core 2 To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] No takers? I'd love to get some advice from folks running ztdummy and fedora core 2. RememberI'm also running a dual proc machine. I understand that this may limit other timing alternatives. Thanks for any advice! Chad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chad Brown Sent: Wednesday, September 15, 2004 12:33 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ztdummy on Fedora Core 2 I followed the Wiki instructions to get zaptel to work on Fedora core 2. It looked like everything went perfect including the loading of ztdummy. However, I am having meetme and MOH problems synonymous with ztdummy not loading. Take a look at my lsmodAny ideas? (I am running stable Asterisk on a DL360 - Dual processor) Module Size Used by snd_pcm_oss46201 0 snd_pcm81733 1 snd_pcm_oss snd_page_alloc 12233 1 snd_pcm snd_timer 25157 1 snd_pcm snd_mixer_oss 17985 1 snd_pcm_oss snd47397 4 snd_pcm_oss,snd_pcm,snd_timer,snd_mixer_oss soundcore 10785 1 snd ztdummy 7364 0 zaptel231684 1 ztdummy crc_ccitt 5825 1 zaptel md5 7745 1 ipv6 233701 16 autofs420165 0 sunrpc128805 1 tg379045 0 floppy 54481 0 sg 33377 0 microcode 10209 0 dm_mod 49477 0 ohci_hcd 22097 0 button 8793 0 battery11085 0 asus_acpi 13017 0 ac 7373 0 ext3 99497 2 jbd58457 1 ext3 cciss 41765 4 sd_mod 20801 0 scsi_mod 102025 3 sg,cciss,sd_mod ___ 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] Virtual queue member
I don't really see how that's possible with the current Queue setup. I don't see why you couldn't use AGI or an app to query a callback table and orginate the call back and connect it to an available agent. I'm curious on this as well so feel free to contact me offlist. I'm going to add it to 1 of my asterisk concepts to work thru. -- William On Thu, 9 Sep 2004 17:33:41 +0100, Ben Merrills [EMAIL PROTECTED] wrote: I was wondering if anyone knew how to do the following Call comes in, gets put into a Queue, say `Sales`. Then the queue member is presented with the option to exit the queue, yet have the phone system sit in their place for them. When the virtual member reaches the front, call back the caller and connect them to the agent. Any ideas? Did i explain that ok? :) Cheers, Ben Merrills skrusty. ___ 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] Store data from call to database
Sounds like it be best as a custom app or AGI depending how many calls you will be taking and how bad the performance hit of using an AGI vs Compiled app is for your needs ___ 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] VM access
It is but you need to modify your dial plan to make it work.
I do it like such
[inbound] ; context that takes inbound calls and matches em and routes according
exten = 91808,1,Macro(stdexten,101,SIP/101) ; fwd
exten = 55,1,Goto(all-exten,101,1)
; fwd goes start to my stdexten to 101 which doesn't have the options
to press * at vm
; the did goes to all-exten which has some added dialplan features to
make * work for vm
[all-exten]
exten = 0,1,Macro(stdexten,0,SIP/0)
exten = _[1-6]XX,1,SetVar(VMBX=${EXTEN})
exten = _[1-6]XX,2,NoOp(${VMBX})
exten = _[1-6]XX,3,Macro(stdexten,${EXTEN},${EXTEN})
exten = a,1,VoicemailMain(${VMBX})
exten = a,2,Hangup
the a extension is called when * is pressed while in vm . The macro
returns to the context and goes to the a extension
- Original Message -
From: Larry Shields [EMAIL PROTECTED]
Date: Mon, 6 Sep 2004 12:14:26 -0500
Subject: RE: [Asterisk-Users] VM access
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
On most VM systems you can press the * key or # key to get a login
prompt during your greeting. Is that not possible with this system?
Thanks,
Larry
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lyle
Giese
Sent: Monday, September 06, 2004 11:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] VM access
You could dedicate a PSTN line( phone number) for that purpose. You
could put a menu system(auto-attendant style) and just dial 8500(demo
is set for this exten to be the gateway to VM). Or if your operator
answers, have her transfer your call to 8500.
Lyle
- Original Message -
From: Larry Shields
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Sent: Monday, September 06, 2004 11:48 AM
Subject: [Asterisk-Users] VM access
Can someone tell me how to get to a mailbox login prompt when
accessing the Asterisk VM remotely via a PSTN line? I am running
version CSV 8/25/04.
Thanks,
Larry
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Re: [Asterisk-Users] Asterisk Conferencing using g729
Good call Daniel I didn't even notice that. As far as number of license it really depends on how many concurrent calls you will be doing and if asterisk needs to transcode at all. If you call from g729 device to g729 you are fine but g729 to vm would be 1 license etc. On Mon, 06 Sep 2004 04:51:26 -0500, Daniel Jimenez [EMAIL PROTECTED] wrote: box100 wrote: My iax.conf file includes the following under the general section A SIPURA is a SIP device, configure the codecs under sip.conf not IAX.conf. disallow=all bandwidth=low allow=g729 allow=ulaw Thanks, Roger Easlick ___ 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 -- Daniel Jimenez djimenez[at]pobox[dot]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] cvs server problem
On Mon, 06 Sep 2004 13:22:51 +0300, Vladyslav [EMAIL PROTECTED] wrote: Today morning cvs server checkout problem: cvs server: Updating asterisk-addons/format_mp3 cvs server: failed to create lock directory for `/usr/cvsroot/asterisk-addons/format_mp3' (/usr/cvsroot/asterisk-addons/format_mp3/#cvs.lock): Permission denied cvs server: failed to obtain dir lock in repository `/usr/cvsroot/asterisk-addons/format_mp3' cvs [server aborted]: read lock failed - giving up -- Best regards Vlad ___ 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 try again they should be round robin still and probably be fixed by now ___ 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 sudo from httpd
why not use a tcp socket and use the manager api and avoid the
permission issues all together
enable it in manger.conf and you connect over tcp log in and execute
the command nice and cleanly in your application. There should be
decent examples on voip-info.org
On Sun, 5 Sep 2004 23:52:13 +0200, Roland Zagler [EMAIL PROTECTED] wrote:
Hello!
I want to use asterisk -rx show version from a php script called in
the browser using the local apache, which runs as user apache.
Asterisk is running as root.
I added the following line to /etc/sudoers using visudo:
apacheALL = NOPASSWD: /usr/sbin/asterisk
When i am on the command line of my linux box it looks like this:
# sudo /usr/sbin/asterisk -rx show version
Asterisk 1.0-RC2 built by [EMAIL PROTECTED] on a i686 running
Linux
# sudo -u apache /usr/sbin/asterisk -rx show version
Unable to connect to remote asterisk
strace showed me that there is an access problem with
/var/run/asterisk.ctl:
munmap(0xbf334000, 4096)= 0
socket(PF_FILE, SOCK_STREAM, 0) = 3
connect(3, {sa_family=AF_FILE, path=/var/run/asterisk.ctl}, 110) = -1
EACCES (Permission denied)
close(3)= 0
time([1094419366]) = 1094419366
fstat64(1, {st_mode=S_IFCHR|0620, st_rdev=makedev(136, 0), ...}) = 0
mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1,
0) = 0xbf334000
write(1, Unable to connect to remote aste..., 37) = 37
munmap(0xbf334000, 4096)= 0
exit_group(1) = ?
System description:
Fedora Core 1
Kernel 2.4.22
Sudo 1.6.7p5
Apache httpd 2.0.50
Asterisk 1.0-RC2
Can anyone please help?
Thank you in advance!
Roland Zagler
mailto:[EMAIL PROTECTED]
@fog smart partners
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Re: [Asterisk-Users] Asterisk Conferencing using g729
Roger, I haven't had any problems doing confs w/ g729. My guess is the Sipura is asking for ulaw first. Try adjusting the codec priority on the sipura side. IF you still have problems I an get my spa-3000 out and trying and solve it for you. -- William - Original Message - From: box100 [EMAIL PROTECTED] Date: Mon, 6 Sep 2004 00:28:50 -0400 Subject: [Asterisk-Users] Asterisk Conferencing using g729 To: [EMAIL PROTECTED] Could anyone who has successfully configured Asterisk to use g729 to conference 10-20 people please post their configs. I purchased and successfully installed 2 g729 licenses and but when I dial into my conference number on the Asterisk box from a SPA-2000 set to allow all codecs, it always appears to connect using ULAW. My iax.conf file includes the following under the general section disallow=all bandwidth=low allow=g729 allow=ulaw Thanks, Roger Easlick ___ 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] Free WWT (WorldWideTelco): Utopia, or just a matter of organization?
Best bet for such a CoOp would be a give and take relationship. If they also give you access to something of theirs they are more likely to treat your stuff with care as well. But it is risky. On Sat, 4 Sep 2004 22:11:37 +0100, Kevin Walsh [EMAIL PROTECTED] wrote: Marconi Rivello [EMAIL PROTECTED] wrote: I was thinking: we could build an Asterisk network, maybe go even further and make it P2P like skype (but I believe it's not necessary at the beginning), and every user would share it's phone line, and be able to place calls to PSTN through the other users' phone lines. So, I let Japanese people call my neighbours for free, and an Italian guy may let me use his phone, so he can use the Indian guy's phone... In US, local calls are free. So it wouldn't be a problem to make such a network to get rid of long distance calls. But in other countries (like here in Brazil) local calls are charged. So there could be some king of billing (without commercial purposes, just to pay for the costs), or something... What you're suggesting is possible, but has its drawbacks: 1. A home user who has one phone line, and opens it up to the world for local calls, may find that the line is in use by a bunch of Brazilian people when he goes to use it, or tries to make an emergency call. 2. Your phone line may be used to make crank calls, or to place fake pizza orders etc. 3. A wide range of other issues that seem to have slipped my mind. This sort of thing is best organised centrally, rather than by a bunch of people opening up their phone system to the world. A central body would have control over who gets to use the service, and can cancel a subscription, and take more effective action, if abuse is proven. A central body would also be better prepared to trace who made that crank call. Although it's a nice idea, it's not really practical, in my opinion. That sort of setup is best left to companies who want to allow local dialout from any office, rather than as a publicly-accessible effort. Of course, if you want to open up your phone line then don't let me stop you. :-) You could persuade a bunch of people you know to do the same, but I'd advise against opening up such a network to the unwashed general public. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] Call recording
check voip-info.org for call recording. There is a dial plan example using Monitor for that - Original Message - From: William C. Lohr Jr. [EMAIL PROTECTED] Date: Sun, 5 Sep 2004 00:35:57 -0400 Subject: [Asterisk-Users] Call recording To: [EMAIL PROTECTED] Newbie here. Learning a lot by reading the lists. Does anyone know If Asterisk will record call if you want it to. ie. for a small call center. Or would some programming need to be done on the workstation side if you were creating a softphone of sorts? Bill Lohr ___ 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] digitnetworks card issues?
Digitnetworks is profiting off the cards so they should support them.
If it wasn't for Digium there wouldn't be Asterisk anyway. So doesn't
that make it better to support the primary company for software that
many of you use every day at home and work?
On Fri, 3 Sep 2004 08:40:59 +0100, Kannaiyan Natesan
[EMAIL PROTECTED] wrote:
Does it mean that we cannot talk about Cisco or other FXS products since
IAXy is released??
I hope this list for every member who uses asterisk not Digium's products
users alone.
- Original Message -
From: Jay Milk [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
[EMAIL PROTECTED]
Sent: Friday, September 03, 2004 8:09 AM
Subject: RE: [Asterisk-Users] digitnetworks card issues?
Have you contacted digitnetworks for support? This list is owned and
maintained by Digium, who already gave you Asterisk for free. Probably
not the best forum to ask for support for a competitive product here.
-Original Message-
From: Imran Akbar [mailto:[EMAIL PROTECTED]
Sent: Friday, September 03, 2004 1:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] digitnetworks card issues?
Hi,
I've purchased two x100p clones, and when I try accessing a line
from asterisk with something like this:
exten = _1NXXNXX,1,Dial(Zap/2/{$EXTEN})
(is that only supposed to put you on channel 2 or actually dial the #
for you?)
but I first hear noise, then a dial tone, but as soon as I
start dialing
numbers I get feedback and noise, and the call doesn't go through.
Any suggestions?
Thanks,
Imran
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Re: [Asterisk-Users] Commercial CID spoofing system
star38.com .25 connection .07-.13 per min What a bargin On Thu, 02 Sep 2004 16:16:26 +0200, Stefan de Konink [EMAIL PROTECTED] wrote: Brian Capouch wrote: FYI. Reading is free; if you don't have an account it is trivial to sign up, and they're very politically correct, as might be imagined, about using email for selling purposes. http://www.nytimes.com/2004/09/02/technology/02caller.html?hp Bugmenot.com: Login details for www.nytimes.com Account #1 xxgeo 12345 ___ 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] Any UK PipeCall/PipeMedia users?
I know someone who was looking into it but they decided not to make the investment at this time versus other options they had available. Prices did look decent though. On Thu, 2 Sep 2004 11:10:37 +0100, David Gurr [EMAIL PROTECTED] wrote: Has anyone out there used the PipeMedia/PipeCall PSTN gateway? Anything good/bad to say about it? I'm considering using them for a new customer. They seem to have good rates, good provisioning tools and (better still) give commission on usage to dealers. -- David Gurr Congruity Ltd. Fax: 0871 661 1756 Hemel Hempstead UK ___ 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] Sorry, Newbie here
In theory yes. Depending on if you wish to use your own PRIs or remote termination for asterisk as well as the phones you choose it could be done quite economicly versus the other options. Although due to the nature of asterisk you have alot of decisions as to how to go about it. Since you haven't worked with it before I'd suggest you have a consultant quote the install to meet your specs versus you taking the time to learn the ins and outs yourself. -- William - Original Message - From: William C. Lohr Jr. [EMAIL PROTECTED] Date: Thu, 2 Sep 2004 22:33:11 -0400 Subject: [Asterisk-Users] Sorry, Newbie here To: [EMAIL PROTECTED] I never heard of Asterisk before today, but from what i'm looking at on the website and hearing, it sounds pretty incredibly. If I understand correctly with a 1,500.00 Wildcard TE410p T1 card, a good BSD or Linux Server, and a couple IP phones or Netmeeting on a few workstations, and of course, Asterisk which is free; I call have a small call center. This can't be? I was looking at tens of thousands for a Cisco solution. Any comments or insight is welcome. Thanks, Bill Lohr William C. Lohr Jr. Lohr Technologies, LLC www.lohrtechnologies.com (301) 334-8758 [EMAIL PROTECTED] ___ 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] Migrating Asterisk
1) should be more than enuf for 1 channel. I use a P2 400 here for testing and it worked ok for transcoding besides the schedule notices. 2) Depending how much timing you need to do X100P or ztdummy could even work just fine. 3. -head 4. i'd rebuild it from src and just copy your configs and any of your stored VM On Wed, 1 Sep 2004 14:39:12 -0500, Jay Milk [EMAIL PROTECTED] wrote: Hello All, My asterisk installation has now been running for over two months without a hitch, and I've decided it's time to move things around a bit. It's currently installed on a 2.7GHz Celeron under RH9 installed on a 10GB leftover drive. Thanks to the strange marketing method called Mail-In-Rebate, I have a fresh 160GB drive ($50), and I'm itching to install a GenToo Linux distro. I also have a 1.2ish GHz Duron with Mobo sitting around here, which may just be enough to power my (barely ever transcoding) asterisk install. Should be enough, even if one channel were transcoded occasionally, no? Let's say I start with a fresh machine, GenToo (2.4 Kernel), and a recent Asterisk (which one? I'm running HEAD from 05/02/2004 right now, heavy on SIP, no problems), and move one of my two X100P for the timing source... Would it be enough to copy over the Asterisk config and VM files? (yes, yes, they'll share IP addresses, so I don't have to reconfigure my devices) So... 1) 1.2Ghz Duron, enough for transacoding a single channel? 2) X100P sufficient timing device for *? 3) Which * source does the list recommend? 4) \var\lib\asterisk, \etc\asterisk and zaptel.conf are all that's needed to migrate the current state of *? TIA! ___ 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] Why are you guys promoting a Rippoff
The wiki allows everyone to post pages on whatever they wish. This means a company can post settings in reference to their company or anyone else could for that matter. On Wed, 1 Sep 2004 21:56:30 -0400, Michael Workman [EMAIL PROTECTED] wrote: On your web you have a link http://www.voip-info.org/wiki-Asterisk+settings+Diamondcard To Setup Calling with Diamondcard.us and I signed up and paid the money according to Stephen Karrington it was all automated... And it was automated to take money but when you look for service hookups or information you don't get it. I have tried now for last little while to contact them for support and got nothing. ___ 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] Distinctive Ring Cadences
just store the cids of your high paying accs and give them vip treatment or a different did to call in =) On Thu, 26 Aug 2004 12:39:49 +1200, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On 25 Aug 2004 at 21:34, Nicolas Gudino wrote: On Wed, 2004-08-25 at 20:38, Chris Shaw wrote: Cool! I could see this being very useful, for example you could have an IVR that says something like Please set the priority of your call, 1 for urgent, 2 for normal or 3 for low then if 1, bellcore-r4, if 2 bellcore-r3, if 1 bellcore-r1! What for? People will always hit 1 g That's why you kinda need to make it an after call thing. LOL you could even use it in a queue... I.E. caller id starts with rating of 50 (max 100, min 0) After call press 1 for annoying, 2 for useful Then every time you press 1 their rating goes down...which could cause the queue priority to be higher...so if someone calls in with a rating of 25 and someone else with 75 you answer the 75 first! :-) Matt ___ 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] Telemarketer screening
astdb dbget for the cid would probably be cleaner and if doesn't return a result then unknown cid but good idea On Thu, 26 Aug 2004 22:22:33 -0400, Mark Woods [EMAIL PROTECTED] wrote: David, Yes I have, and also with call through direct for friends. What I've done is implemented a caller ID check that works this way: 1) Incoming call is checked for friends/family caller ID and sent directly to the inside extension (9491). You will see that as the first part of the config below, with the numbers/names edited for the list, of course. 2) Incoming call is checked for annoying caller ID and sent to a goodbye message and then disconnected. 3) All other calls get a standard greeting. If you know the inside extension to dial, you can dial it while the greeting plays (for friends/family that are not calling from their normal number). If you don't you'll be told in the message to dial another extension that takes you directly to voicemail. If you don't dial anything, you'll time out and be disconnected. This works well, it seems. Friends/family get through fine, annoying people won't get anywhere, and everyone else gets a standard greeting. This standard greeting takes care of almost 100% of the telemarketer calls because they're not human and can't interpret my message. Their system just thinks it's an answering machine and hangs up. A human telemarketer can choose to leave a message if that person so desires, but, to be honest, between this and the do not call list, I haven't had an issue. Hope this helps...config below. -Mark .. Here's what's worked for me: ; We begin by matching caller id for dialthrough ; Answer the line exten = s,1,Wait,1 ; Wait a second, just for fun exten = s,2,Answer ; Answer the line ; Match caller ID, if you can exten = s/##,3,Goto(9491,1); Work exten = s/##,3,Goto(9491,1); alt work exten = s/##,3,Goto(9491,1); Mom at home exten = s/##,3,Goto(9491,1); Moms cell etc, etc, etc, all of my friends and family ; Throw unwanted callers away exten = s/##,3,Goto(t,1) ; Annoying person... ; Continue to prompt if no callerid match exten = s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten = s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten = s,5,BackGround(markintro) ; Play the intro david kwok wrote: I have been bugging by a telemarketer who does not take any cue at all. So I look up the Asterisk Handbook and send his call with the respect caller id to my voicemail. Has any one implemented any of this feature with database for more caller ids to be included?? David Kwok D ___ 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] Grandstream Budgetone BT-101 and VoipJet
I had this issue with a grandstream as well a week or so ago and have yet to solve the issue. Until I get my Budgetone here physically again I won't be able to mess with it hands on. What did you use for codec/signaling and did your asterisk box see any warnings or errors? On Tue, 24 Aug 2004 15:56:38 -0700, John Week [EMAIL PROTECTED] wrote: Is anyone using this combination successfully? I have a dell 500sc running rh9 and asterisk 1.0rc1. It is configured with an x100p. I have a Sipura SPA-2000, laptop with Xlite and a Grandstream Budgetone BT-101. I have signed up with Voipjet (they use iax2). I also have an FWD number via iax2. I can sucessfully call back and forth to all devices via zap, sip, and fwd. I can successfully place calls using voipjet with everything except the grandstream. When I place a voipjet call with the grandstream, I can hear the party I'm calling, but they can't hear me. I have tried all the different codecs the grandstream supports without luck. I am running the 1.0.5.10 firmware. I've emailed voipjet support about it, but they don't have one. 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
Re: [Asterisk-Users] SIP unphones
Post some pictures when you are all done. Looks like an interesting task just no need to dive into it myself at this time. On Tue, 24 Aug 2004 18:08:28 -0500, Jay Milk [EMAIL PROTECTED] wrote: Yep, great idea, that's what's next -- and I have two extra extensions (Sipura) -Original Message- From: Chris Shaw [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 24, 2004 12:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP unphones Check out my ATA idea though, with a regular cheap analog doorphone and a HTX86 or even Sipura, you can program the ATA to dial an extension as soon as the button on the intercom is pressed and then with some extension logic you can do neat things... You can get a doorphone anywhere even radio shack I think and the HTX86 is like $60-70... - Original Message - From: Jay Milk [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Monday, August 23, 2004 7:16 PM Subject: RE: [Asterisk-Users] SIP unphones Thank you -- funny thing is, I had the same bookmarked, but it just seemed too expensive for the application -- for $300, I can stick a cheap IP phone in a hole in the wall :) I think it's time to get a Budgetone. -Original Message- From: Chris Shaw [mailto:[EMAIL PROTECTED] Sent: Monday, August 23, 2004 3:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP unphones I recently saw something just like this and I had it bookmarked... It looks like what you're talking about, but I don't think it uses SIP. Rather some proprietary protocol that transmit RTP... I could be wrong... Check it out... http://www.digitalacoustics.com/lanplay.htm I would agree that it really should be SIP, you wouldn't want to have to rip it out of the wall when the protocol becomes obsolete or when a SIP-Compliant alternative comes out... -Chris - Original Message - From: Jay Milk [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, August 23, 2004 7:55 AM Subject: [Asterisk-Users] SIP unphones Does anyone know if there are additional SIP devices out there which aren't phones? I'm basically looking for a fully-automatic SIP speakerphone. I'd like to be able to dial a sip-extension and make an announcement (PA) and/or simply listen in to a room (baby-monitor). Yes, I know, some of the more advanced phones can be configured to behave like that, but it seems to a waste of money to have all those fancy displays and keys tucked away behind a speakergrille and drywall. BTW, I'm not dead-set on SIP, but it seems to be the most logical protocol for this app (NOTIFY msg can carry directions on mike/speaker/two-way, etc) ___ 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/aster isk-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/aster isk-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
