[asterisk-users] Setting CallerID UNKNOWN on an outgoing call
Hello, on my ISDN phone I can configure that on the next outgoing call, my telephone number should not be transmitted, instead it should be UNKNOWN. How can I configure Asterisk to do the same? Is this a feature/parameter of the driver (chan_capi) that I'm using? BTW: I'm using ISDN and Deutsche Telekom, if the provider makes any difference. Thanks for your help, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting CallerID UNKNOWN on an outgoing call
On PRI SetCallingPres works fine it should work with ISDN because its the same signaling. -= Info about application 'SetCallerPres' =- [Synopsis] Set CallerID Presentation [Description] SetCallerPres(presentation): Set Caller*ID presentation on a call. Valid presentations are: allowed_not_screened: Presentation Allowed, Not Screened allowed_passed_screen : Presentation Allowed, Passed Screen allowed_failed_screen : Presentation Allowed, Failed Screen allowed : Presentation Allowed, Network Number prohib_not_screened : Presentation Prohibited, Not Screened prohib_passed_screen: Presentation Prohibited, Passed Screen prohib_failed_screen: Presentation Prohibited, Failed Screen prohib : Presentation Prohibited, Network Number unavailable : Number Unavailable On Wed, May 14, 2008 at 2:08 AM, Stefan Guenther [EMAIL PROTECTED] wrote: Hello, on my ISDN phone I can configure that on the next outgoing call, my telephone number should not be transmitted, instead it should be UNKNOWN. How can I configure Asterisk to do the same? Is this a feature/parameter of the driver (chan_capi) that I'm using? BTW: I'm using ISDN and Deutsche Telekom, if the provider makes any difference. Thanks for your help, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No-mobo PC for USB Drives Enclosure?
As people have sugested the ATX power supplies can work without a mobo One thing to watch out for your setup is the actual ampere requirments for your disks i.e Your power supply provides 300W but this is partitioned to different voltages (+5, +12, etc) with different amp charecteristics Disks need 2 voltages. One for the logic (+5V) and one for the motors (+12V) and have different current requirments. Most disk (if not all) mention these ratings on the labels they have What you must do, is to see if by adding the current requirments seperatly for +5V and +12V, does not exceed the power supply's amp rating *for that voltage*, allowing also for a 15% -20% margin, as power consumption will be higher than the nomimal mentioned during disk startup (and you will be starting all your disks at the same time) Also make sure your box has sufficient cooling and there is some short of airflow over the disks, as the number 1 enemy of disks is high temperature and stacking so many disks in a box will create large amounts of heat. I would suggest you to get a good (aka expensive) 500W power supply and use 10-12 disks with it to avoid problems in the long run, Also keep in mind that MTBF specs of SATA disks does not make them an ideal candidate for 24/7/365 operations Stelios S. Koroneos Digital OPSiS - Embedded Intelligence http://www.digital-opsis.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Rubenstein Sent: Wednesday, May 14, 2008 7:31 AM To: Col Ferguson Cc: Asterisk -Users Subject: Re: [asterisk-users] No-mobo PC for USB Drives Enclosure? On Wed, 2008-05-14 at 14:06 +1000, Col Ferguson wrote: If I understand right, your problem is that the power supply won't turn on ? ATX power supplies can be told to turn on by jumpering 2 pins on the motherboard power connector. From memory its the Green wire and one of the black wires, I usually use the next one inwards. Pinouts for the connector can be found via Google. If the power supply also has an external on/off switch you can jumper these pins and use the switch to turn the power on or off. Hope this helps, Thanks, that sounds like exactly what I was looking for. Is there any safety risk from jumpering that sensor? Like some kind of extra sensor, like voltage feedback, temperature or somesuch. If this works, it might point to a good way to reduce redundant Asterisk servers, which suck power, by just plugging the drive from each old server into the USB of a single server with a merged dialplan and a few other tweaks to point at several different mounted drives, rather than one per host/IP#. Col - Original Message - From: Matthew Rubenstein [EMAIL PROTECTED] To: Asterisk -Users asterisk-users@lists.digium.com Sent: Wednesday, May 14, 2008 12:22 PM Subject: [asterisk-users] No-mobo PC for USB Drives Enclosure? I have over a half-dozen different SATA hard drives, each with different data (configs, voiceprompts, voicemail, CDRs, AGIs) for each one's different user groups and applications. Each one's load on the Asterisk server is small enough that one server can host them all, accessed easily over USB. But right now, each one is in its own external USB enclosure on a powered USB hub. I want to combine them all into a single large enclosure. I tried to use a single PC chassis, leaving the USB hub inside with the drives screwed into it, and powered from the PC power supply as internal drives on the proper drive power output plugs. But without a PC motherboard plugged into the power supply, too, the power supply won't start up to power the drives. I don't want to add a motherboard: that costs money, and sucks power, and is totally unnecessary. I just want to make this gutted PC chassis power my drives only, and have them connect to the complete PC sitting next to it via the single USB cable coming out of the drive chassis. How do I do that? Is it possible to use the extra, unused floppy power plugs to power more hard drives, with an adapter? Is it possible to split the existing hard drive power plugs to each power multiple drives? How many drives can I split each power plug into? The power supply is a cheap 300W unit, and the drives draw max under 9W each: http://www.wdc.com/en/products/products.asp?driveid=311 . So can I power 25-30 of these drives, or at least 10? -- (C) Matthew Rubenstein ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG. Version: 7.5.524 / Virus Database:
Re: [asterisk-users] Calls on E1 TDMoE span are dropped at random
On Tuesday 13 May 2008, Steve Totaro wrote: Can you describe exactly how you are utilizing it, including LAN/WAN, switches, ping times, and other network central details. TDMoE adds the E (ethernet) component to troubleshooting and I think do to this, it may be very fragile depending on network conditions. Don't make the mistake of just focusing on Asterisk and Zaptel in your troubleshooting process. Hi! Thank you very much for your suggestions. Yesterday evening, the telco technician disabled a feature they call NT1 on their side, which is supposed to be a protocol for line monitoring. We tested again and were unable to reproduce the call dropping problem. We established 15 calls to our private extensions to fill all 30 channels and had the calls running throughout the night. I just checked, and they were all still up and running. As it currently appears to me, this line monitoring feature caused the problems. Unfortunately I have been unable to find anything related to NT1 and line monitoring on the internet. I've been in touch with the telco (Telekom Austria) and they will try to find some information concerning this feature. I'll report back with more info as soon as possible. Once again, thank you very much for your support! I really hope this issue is solved (I've been searching for the cause for more than two weeks now). Cheers, Florian -- DI Florian Hackenberger [EMAIL PROTECTED] www.hackenberger.at ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues, monitor-join=yes, and volume
Thanks. If I find out some settings for soxmix, do you maybe know where can I change Asterisk settings for soxmix (parameters)? Regards, Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Backeberg Sent: Tuesday, May 13, 2008 5:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queues, monitor-join=yes, and volume On Tue, May 13, 2008 at 10:42 AM, Asterisk [EMAIL PROTECTED] wrote: Is there any way to modify the volume (either lower the volume of the clients, or increase the volume of the agents) while doing the join of the -in and -out files into one recording? Uh-huh. Read the documentation for soxmix. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No-mobo PC for USB Drives Enclosure?
On Wed, May 14, 2008 at 10:41 AM, Stelios Koroneos [EMAIL PROTECTED] wrote: As people have sugested the ATX power supplies can work without a mobo One thing to watch out for your setup is the actual ampere requirments for your disks i.e Your power supply provides 300W but this is partitioned to different voltages (+5, +12, etc) with different amp charecteristics Disks need 2 voltages. One for the logic (+5V) and one for the motors (+12V) and have different current requirments. Most disk (if not all) mention these ratings on the labels they have What you must do, is to see if by adding the current requirments seperatly for +5V and +12V, does not exceed the power supply's amp rating *for that voltage*, allowing also for a 15% -20% margin, as power consumption will be higher than the nomimal mentioned during disk startup (and you will be starting all your disks at the same time) Also make sure your box has sufficient cooling and there is some short of airflow over the disks, as the number 1 enemy of disks is high temperature and stacking so many disks in a box will create large amounts of heat. I would suggest you to get a good (aka expensive) 500W power supply and use 10-12 disks with it to avoid problems in the long run, Also keep in mind that MTBF specs of SATA disks does not make them an ideal candidate for 24/7/365 operations Another thing is voltage feedback. The Gray wire should be grounded when +5 and +3.3 V is ok for m/b. As +5 is shared also for disk connectors, there could be some problems. Also be advised that you should buy good power supply, as the difference is in voltage stability, and hard disks don't like floating voltages much. I would suggest you to go better for some network oriented setup, use NFS ir CURL for configs, etc. Imagine what will happen if that one PSU will fail. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No sound with Playback() and Background()
Hi
I've got a Dell with Intel Xeon and no Zaptel hardware installed.
However, as it needs to do IAX trunking and MeetMe conferences I need
timing enabled using ztDummy.
However, when enabling ztDummy, Playback() and Background() both fail to
play.
If I place NoOp(${PLAYBACKSTATUS}) after the Playback() call it never
reaches it. It hangs on the playback then kills the call after a certain
time.
As soon as I rmmod ztdummy and zaptel it starts working again.
I've found the following mail list thread which seems to be very similar
but no solution came to pass:
http://readlist.com/lists/lists.digium.com/asterisk-users/9/47641.html
I'd be very grateful for anyones input.
All the best.
Doug
--
Essential Systems Ltd.
137 Golden Cross Lane
Catshill
Bromsgrove
B61 0LA
Tel: 0845 867 9002
Fax: 01527 557 282
DiD: 01527 557 288
CoN: 06253751
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Re: [asterisk-users] Setting CallerID UNKNOWN on an outgoing call
Hi, exten = _0[23456789].,1,SetCallerPres(prohib) did it for me. Thank you, Stefan Andreas van dem Helge wrote: On PRI SetCallingPres works fine it should work with ISDN because its the same signaling. -= Info about application 'SetCallerPres' =- [Synopsis] Set CallerID Presentation [Description] SetCallerPres(presentation): Set Caller*ID presentation on a call. Valid presentations are: allowed_not_screened: Presentation Allowed, Not Screened allowed_passed_screen : Presentation Allowed, Passed Screen allowed_failed_screen : Presentation Allowed, Failed Screen allowed : Presentation Allowed, Network Number prohib_not_screened : Presentation Prohibited, Not Screened prohib_passed_screen: Presentation Prohibited, Passed Screen prohib_failed_screen: Presentation Prohibited, Failed Screen prohib : Presentation Prohibited, Network Number unavailable : Number Unavailable -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-dev] UWB Codec / Command-line softphone help
On Wed, May 14, 2008 at 10:15:01AM +0200, Koch Máté wrote: Tim Panton wrote: I think that if you use meetme, you will automatically drop to 8khz sampling because that is what zaptel uses to do the mixing. If you want wideband, you will probably need to make one-to-one calls. That is correct. However, if you install Josh's bridging branch (asterisk/team/file/bridging), and use the ConfBridge application, you can get wideband conferencing. However, you still need a softphone that supports a wideband codec. If you are actually preferring a command line softphone, then my preference is actually to just use Asterisk. It can act as a pretty powerful and highly configurable softphone. :) If you use Asterisk as the softphone, then use chan_console. It is set up to operate in 16 kHz natively. Also, use G.722 as the codec between the servers. Hello, thank you very much for help. Is there any tutorial about how to configure this, what to install and so? How to use Asterisk as a wide-band soft-phone using a local sound card? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting CallerID UNKNOWN on an outgoing call
Hello, Andreas van dem Helge schrieb: On PRI SetCallingPres works fine it should work with ISDN because its the same signaling. -= Info about application 'SetCallerPres' =- [Synopsis] Set CallerID Presentation [Description] SetCallerPres(presentation): Set Caller*ID presentation on a call. Valid presentations are: allowed_not_screened: Presentation Allowed, Not Screened allowed_passed_screen : Presentation Allowed, Passed Screen allowed_failed_screen : Presentation Allowed, Failed Screen allowed : Presentation Allowed, Network Number prohib_not_screened : Presentation Prohibited, Not Screened prohib_passed_screen: Presentation Prohibited, Passed Screen prohib_failed_screen: Presentation Prohibited, Failed Screen prohib : Presentation Prohibited, Network Number unavailable : Number Unavailable I must admit that i am alway a little bit at a loss, about most of the 9 possible settings above. Well, 'allowed' and 'unavailable' are totally clear ;) but the Rest ?? Lets say i have a configured number range from 1000 to 1999 and 1000 is my base number. I make an outgoing call from a phone which sets its CallerID to 1500. Can anyone be so kind to tell me what is shown to the callee in either case? Thank you very much ... -- Tobias Wolf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MeetMeAdmin() working problem
Hello users, This is regarding MeetMeAdmin() administration from DialPlan exten = 12345,1,MeetMe(123|MX) ; Enter conference number 123 ;Exit conference by pressing a single digit exten = 12345,2,Hangup() exten = 1,1,MeetMeAdmin(123|M|1) ;mute the user 1 exten = 2,1,MeetMeAdmin(123|m|1) ;un-mute the user 1 exten = 3,1,MeetMeAdmin(123|k|1) ;kick the user 1 actually i supposed to give the user values from the usernumber field of meetme list confnumber command at CLI i cannot give a channel name (ex: 1000 as in SIP/1000) in the above MeetMeAdmin() command under user and the application storing the first channel in user number 1 and so on... so from the dialplan how can i control the users for management purpose(single user mute,single user unmute ,single user kickout) can it be done??? or cannot?? waiting for valuable suggestions thanks in advance regards srinivas antarvedi Srinivas Antarvedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call only for registered sip users...
On Tue, May 13, 2008 at 7:31 PM, equis software [EMAIL PROTECTED] wrote: What I need to configure in my * to permit make calls only registered sip users?? Nothing. You can't call unregistered SIP users since you don't have any contact information for them so therefore all your calls will only ever be to registered ones. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Asterisk Deployment - Need some tips
On Tue, May 13, 2008 at 12:17 PM, Matthew Ratliff [EMAIL PROTECTED] wrote: I'll be doing a new Asterisk deployment soon, and would like to gather your thoughts. Here are some items that need to be kept in mind: Support 800 phones (400 of which are analog) Concurrent calls ... ? but need to guess high so that the server can handle this. Voicemail will be required along with sending voice mail attachments to email server. Flash panel for switchboard operator. Needs to be a distributed server design for redundancy and fail-over. Will need to be integrated into an existing PBX until each building is switched over to use the Asterisk servers. If calling 911 from a building among multiple buildings, how can EMS find that person based upon the call? What type of data line should be used in this setup? T1? The physical network will support QOS and the like, so that is not an issue. What type of design/setup do you recommend for this? How about server resources...ie...CPU, RAM, Disk space. How about backups? Does imaging work best if a server were to fail? Any thing else you can think of? If this is a project for your work and it's your first Asterisk deployment then definitely don't go the big bang approach in the way you've outlined. If you do you could well be out of that job in 6 months! The first thing I'd recommend you do is find 10 or 20 people who are suitable as early adopters. The set up a single Asterisk server and give the early adopters a SIP phone each thats in addition to their normal desk phone and ask them to see how they go using the SIP phones for calls to each other, external calls and whatever else would make sense. Then 6 months and a lot of learning/experience/frustration later you'll know whether to get answers to your original questions or not. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues, monitor-join=yes, and volume
You can call the sox binary directly from your dialplan, or any other binary that fits your needs. If you post your dialplan where you're doing the recording, we can give input about where to put the calls to sox. On Wed, May 14, 2008 at 4:16 AM, Asterisk [EMAIL PROTECTED] wrote: Thanks. If I find out some settings for soxmix, do you maybe know where can I change Asterisk settings for soxmix (parameters)? Regards, Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Backeberg Sent: Tuesday, May 13, 2008 5:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queues, monitor-join=yes, and volume On Tue, May 13, 2008 at 10:42 AM, Asterisk [EMAIL PROTECTED] wrote: Is there any way to modify the volume (either lower the volume of the clients, or increase the volume of the agents) while doing the join of the -in and -out files into one recording? Uh-huh. Read the documentation for soxmix. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail not sending emails
i think you have to have a mail transport agent like sendmail or postfix
installed and configured on your asterisk box , however if you forward the
mails to say hotmail or yahoo or gamil those servers will reject the mail
transfere
- Original Message -
From: Roberto Milani [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, May 13, 2008 4:48 PM
Subject: [asterisk-users] voicemail not sending emails
Hello list users
I have a very nice installation of asterisk on a mac mini.
Everything seems to work fine, call works, vm works, even message
transfer works but asterisk doesn't send any email.
this is my voicemail.conf:
[general]
mailcmd=/opt/local/bin/msmtp -t; [EMAIL PROTECTED]
;mailcmd=cat \ /tmp/asteriskvm-mail
format=wav
attach=yes
[EMAIL PROTECTED]
emailsubject=New message from ${VM_CALLERID}
emailbody=Hi, ${VM_NAME}!\n\nYou have a new message from $
{VM_CALLERID} in mailbox ${VM_MAILBOX}.
fromstring=My Telephone System
;max and min length of a message
maxmessage = 180
maxlogins = 3
[default]
100 = 4711,Front Desk,[EMAIL PROTECTED]
as you can see I'm using msmtp for mail and I tested it outside
asterisk an it works.
from the commented line you can se that I tried to cat the output to a
file but that never happens.
It really seems that asterisk don't send the emails.
any suggestions?
Thanks
Roberto
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Re: [asterisk-users] Queuing if no one available to answer
It should already work, unles you configured your queue differently? :) l. On Tue, 13 May 2008 14:44:44 +0200, bilal ghayyad [EMAIL PROTECTED] wrote: Hi list; Any one can advise how to put the caller in the queue in case no one available to take his call? All are busy (having calls)? Regards Bilal -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Asterisk Deployment - Need some tips
I would have to agree with Grey Man, a pilot project is one way to start up. I would also seriously recommend buying some consulting time from an experienced Asterisk PBX vendor/dealer/consultant. The cost is negligible in light of the scope of your project. A pilot project will only give you a glimpse of what is required. You have to have a design that incorporates your eventual build out. A pilot by itself is not going to give you that. You will need help from a source that can bring their experience to help you tip toe around the potential land mines you can encounter. regards, John Signorello Managing Partner ispbx.com 866 GO ISPBX Grey Man wrote: On Tue, May 13, 2008 at 12:17 PM, Matthew Ratliff [EMAIL PROTECTED] wrote: I'll be doing a new Asterisk deployment soon, and would like to gather your thoughts. Here are some items that need to be kept in mind: Support 800 phones (400 of which are analog) Concurrent calls ... ? but need to guess high so that the server can handle this. Voicemail will be required along with sending voice mail attachments to email server. Flash panel for switchboard operator. Needs to be a distributed server design for redundancy and fail-over. Will need to be integrated into an existing PBX until each building is switched over to use the Asterisk servers. If calling 911 from a building among multiple buildings, how can EMS find that person based upon the call? What type of data line should be used in this setup? T1? The physical network will support QOS and the like, so that is not an issue. What type of design/setup do you recommend for this? How about server resources...ie...CPU, RAM, Disk space. How about backups? Does imaging work best if a server were to fail? Any thing else you can think of? If this is a project for your work and it's your first Asterisk deployment then definitely don't go the big bang approach in the way you've outlined. If you do you could well be out of that job in 6 months! The first thing I'd recommend you do is find 10 or 20 people who are suitable as early adopters. The set up a single Asterisk server and give the early adopters a SIP phone each thats in addition to their normal desk phone and ask them to see how they go using the SIP phones for calls to each other, external calls and whatever else would make sense. Then 6 months and a lot of learning/experience/frustration later you'll know whether to get answers to your original questions or not. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Understanding Asterisk
I am about to order some DIDs for my first install but I am unclear on how Asterisk will function in either scenario with the two options I can order with. One option is the DID has unlimited connections. Another option for the DID is that it has a maximum of two concurrent calls only. How does Asterisk understand the multiple calls that are coming in and behave for both scenarios? The phone system we are trying to replace and therefore replicate the functionality is that of a very base Meridian system with 3 lines. Thanks for any guidance! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Asterisk Deployment - Need some tips
Ditto. If you need to quantify the consultant to the powers that be just ask for an Infrastructure Audit. I have done several in the past that have saved tons of money that encouraged further phone projects. Finding dead phone lines to discovering unused but rented telcom gear is always fun. Also when setting up you test group make sure they actually use the phone and often... On Wed, May 14, 2008 at 9:32 AM, John Signorello [EMAIL PROTECTED] wrote: I would have to agree with Grey Man, a pilot project is one way to start up. I would also seriously recommend buying some consulting time from an experienced Asterisk PBX vendor/dealer/consultant. The cost is negligible in light of the scope of your project. A pilot project will only give you a glimpse of what is required. You have to have a design that incorporates your eventual build out. A pilot by itself is not going to give you that. You will need help from a source that can bring their experience to help you tip toe around the potential land mines you can encounter. regards, John Signorello Managing Partner ispbx.com 866 GO ISPBX Grey Man wrote: On Tue, May 13, 2008 at 12:17 PM, Matthew Ratliff [EMAIL PROTECTED] wrote: I'll be doing a new Asterisk deployment soon, and would like to gather your thoughts. Here are some items that need to be kept in mind: Support 800 phones (400 of which are analog) Concurrent calls ... ? but need to guess high so that the server can handle this. Voicemail will be required along with sending voice mail attachments to email server. Flash panel for switchboard operator. Needs to be a distributed server design for redundancy and fail-over. Will need to be integrated into an existing PBX until each building is switched over to use the Asterisk servers. If calling 911 from a building among multiple buildings, how can EMS find that person based upon the call? What type of data line should be used in this setup? T1? The physical network will support QOS and the like, so that is not an issue. What type of design/setup do you recommend for this? How about server resources...ie...CPU, RAM, Disk space. How about backups? Does imaging work best if a server were to fail? Any thing else you can think of? If this is a project for your work and it's your first Asterisk deployment then definitely don't go the big bang approach in the way you've outlined. If you do you could well be out of that job in 6 months! The first thing I'd recommend you do is find 10 or 20 people who are suitable as early adopters. The set up a single Asterisk server and give the early adopters a SIP phone each thats in addition to their normal desk phone and ask them to see how they go using the SIP phones for calls to each other, external calls and whatever else would make sense. Then 6 months and a lot of learning/experience/frustration later you'll know whether to get answers to your original questions or not. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Andrew lathama Latham Principal TuxTone Inc. http://TuxTone.com [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Understanding Asterisk
Joseph The DIDs are tied to a circuit. The circuit has a ring order (ascending or descending or other...). So ordering the DIDs is just getting the numbers most of the time, attaching them to a circuit that is setup to handle the calls in a certain way. Andrew On Wed, May 14, 2008 at 9:57 AM, Joseph L. Casale [EMAIL PROTECTED] wrote: I am about to order some DIDs for my first install but I am unclear on how Asterisk will function in either scenario with the two options I can order with. One option is the DID has unlimited connections. Another option for the DID is that it has a maximum of two concurrent calls only. How does Asterisk understand the multiple calls that are coming in and behave for both scenarios? The phone system we are trying to replace and therefore replicate the functionality is that of a very base Meridian system with 3 lines. Thanks for any guidance! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Andrew lathama Latham Principal TuxTone Inc. http://TuxTone.com [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Announcing the first North America Druid Meetups happening Chicago 22 May 2008 and Altanta 27 May 2008
Dear fellow Asterisk users, Voiceroute is proud to announce the first North America Druid Meetups happening in May 2008 in 2 cities (Chicago 22 May 08) and Atlanta (27 May 2008) Druid Meetups are basically fun demo sessions of Druid (Open Source Edition Unified Communications Server). Come and meet other Druid and asterisk users who are using Druid for enterprise communications deployments. Come join us for free pizzas and win goodies like Blackberry Curve 8320, free copies of Druid UCS! For our first meetups, we have the below cool stuff we will be demoing third party integrations with Druid Asterisk using the Druid SOAP API and how people can develop third party apps - Google Android Mobile application that talks to Druid Asterisk to get user call records like missed calls on the mobile - How to do click to call to any independent SugarCRM server - Tutorial on developing your own third party telephony CTI vertical applications related to Asterisk Druid Please join us by signing up free at the below URLs. Thanks again from the Voiceroute team. If you have any questions, please do not hesitate to email me at [EMAIL PROTECTED] Druid Chicago Meetup Sign up at http://druidchicago.eventbrite.com 22 May 2008, 6pm-9pm CST Location: Seiu73.org Offices 300 S Ashland Ave,Suite 400 Chicago, IL 60607 Organized by Druid User (Rajeev Varkey of Seiu73.org) Druid Atlanta Meetup Sign up at http://druidatlanta.eventbrite.com 27 May 2008, 6pm-9pm EST Location: Hilton Garden Inn Atlanta NE/Sugarloaf 2040 Sugarloaf Circle, Duluth GA 30097 Organized by Druid User (Jeffrey Thompson) -- Ming Yong CEO, www.voiceroute.org Druid - Open Source Unified Communications DID: +1-866-915-2407 ext 301 SIP/email: [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Announcing the first North America Druid Meetupshappening Chicago 22 May 2008 and Altanta 27 May 2008
Ming, Are you coming to New York? Would be great to have an Asterisk related meetup here as well. Regards, Dean Collins [EMAIL PROTECTED] Cognation Limited +1-212-203-4357 +61-2-9016-4652 (Sydney indial) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ming Yong Sent: Wednesday, 14 May 2008 10:31 AM To: Commercial and Business-Oriented Asterisk Discussion; asterisk- [EMAIL PROTECTED] Subject: [asterisk-users] Announcing the first North America Druid Meetupshappening Chicago 22 May 2008 and Altanta 27 May 2008 Dear fellow Asterisk users, Voiceroute is proud to announce the first North America Druid Meetups happening in May 2008 in 2 cities (Chicago 22 May 08) and Atlanta (27 May 2008) Druid Meetups are basically fun demo sessions of Druid (Open Source Edition Unified Communications Server). Come and meet other Druid and asterisk users who are using Druid for enterprise communications deployments. Come join us for free pizzas and win goodies like Blackberry Curve 8320, free copies of Druid UCS! For our first meetups, we have the below cool stuff we will be demoing third party integrations with Druid Asterisk using the Druid SOAP API and how people can develop third party apps - Google Android Mobile application that talks to Druid Asterisk to get user call records like missed calls on the mobile - How to do click to call to any independent SugarCRM server - Tutorial on developing your own third party telephony CTI vertical applications related to Asterisk Druid Please join us by signing up free at the below URLs. Thanks again from the Voiceroute team. If you have any questions, please do not hesitate to email me at [EMAIL PROTECTED] Druid Chicago Meetup Sign up at http://druidchicago.eventbrite.com 22 May 2008, 6pm-9pm CST Location: Seiu73.org Offices 300 S Ashland Ave,Suite 400 Chicago, IL 60607 Organized by Druid User (Rajeev Varkey of Seiu73.org) Druid Atlanta Meetup Sign up at http://druidatlanta.eventbrite.com 27 May 2008, 6pm-9pm EST Location: Hilton Garden Inn Atlanta NE/Sugarloaf 2040 Sugarloaf Circle, Duluth GA 30097 Organized by Druid User (Jeffrey Thompson) -- Ming Yong CEO, www.voiceroute.org Druid - Open Source Unified Communications DID: +1-866-915-2407 ext 301 SIP/email: [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] voicemail not sending emails
Date: Tue, 13 May 2008 22:28:33 -0400 From: OCG Technical Support [EMAIL PROTECTED] Subject: Re: [asterisk-users] voicemail not sending emails To: 'Asterisk Users List' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Permissions? Try running msmtp from the asterisk account? (Assuming that is how you have it setup) I don't know msmtp - but is there a maillog equivalent? MD thanks for the replies but the problem persist to recap: msmtp works just fine from the asterisk user, typing: echo hello. | msmtp --debug --account=myaccount [EMAIL PROTECTED] [EMAIL PROTECTED] I have a log and I receive the mail. I renamed sendmail and linked msmtp and tried the above command with sendmail and the link works too. I removed the mailcmd from voicemail.conf but no email on messages not even without attachment It just does not get called from asterisk. is there a way to debug it? thanks Roberto ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Understanding Asterisk
I assume you are going to with a VOIP provider. Essentially, you have one DID and any number of channels/ports. Typically, you pay per port with a minute charge. Some people give you unlimited ports but charge a higher per minute fee. In you case, where you currently have 3 lines, you would need 3 channels. This would provide three concurrent calls to be in place. Asterisk does not know anything about channels in this example. If 3 calls come in it answers three calls. If a 4th caller comes in , the VOIP prvovider will send the busy signal to the caller. Asterisk does not see it. Joseph L. Casale wrote: I am about to order some DIDs for my first install but I am unclear on how Asterisk will function in either scenario with the two options I can order with. One option is the DID has unlimited connections. Another option for the DID is that it has a maximum of two concurrent calls only. How does Asterisk understand the multiple calls that are coming in and behave for both scenarios? The phone system we are trying to replace and therefore replicate the functionality is that of a very base Meridian system with 3 lines. Thanks for any guidance! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Announcing the first North America Druid Meetups ...
Ming Yong schrieb: For our first meetups, we have the below cool stuff we will be demoing third party integrations with Druid Asterisk using the Druid SOAP API and how people can develop third party apps shameless plug Or if you happen to be located in Germany join us at Asterisk-Tag.org (http://www.asterisk-tag.org). :-) /shameless plug Vikram Rangnekar (Lead Developer on the Druid OSE project) is going to talk about Developing unified communications enabled applications using Druid OSE, Using the Druid SOAP-API to make full use of Druid services to add value to your desktop and web applications. Add presence, call-control , access the unified mailbox, etc directly in your own apps. Grüße, Philipp Kempgen -- Asterisk-Tag.org 2008, 26.-27. Mai - http://www.asterisk-tag.org amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail not sending emails
On Wed, 14 May 2008, Roberto Milani wrote: From: OCG Technical Support [EMAIL PROTECTED] Permissions? Try running msmtp from the asterisk account? (Assuming that is how you have it setup) I don't know msmtp - but is there a maillog equivalent? MD thanks for the replies but the problem persist to recap: msmtp works just fine from the asterisk user, typing: echo hello. | msmtp --debug --account=myaccount [EMAIL PROTECTED] [EMAIL PROTECTED] I have a log and I receive the mail. I renamed sendmail and linked msmtp and tried the above command with sendmail and the link works too. I removed the mailcmd from voicemail.conf but no email on messages not even without attachment It just does not get called from asterisk. is there a way to debug it? thanks Roberto Roberto - I noticed in your original email you had the lines something like mailcmd=/opt/local/bin/msmtp -t ; --from blah AND serveremail=from=blah In mailcmd everything after the ; will be ignored as a comment In serveremail - well - it should throw an error... I would probably test by adding the --debug to the mailcmd and watch the logs. I also don't know mstmp but does it have a '-t' option? Brett ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel Install Error
Steve Totaro wrote: This looks like it may be your problem. http://bugs.digium.com/view.php?id=9592 (0070069) qwell - administrator 09-06-07 17:05 Closing. The simple solution here is to just comment out the #define USE_RTC in ztdummy.c. The ztxen module does not appear to be needed. Thanks, Steve Totaro Just to clarify for those that don't want to read through the bug notes.. That bug was a feature enhancement that added support for xen, through a new module named ztxen. The only difference in this new module vs ztdummy, was that it removed the RTC code. In order to mimic this, and get a proper ztdummy on xen, all somebody needs to do is comment out the single #define USE_RTC line in ztdummy.c ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.20-rc3 and 1.6.0-beta9 Now Available
The Asterisk.org development team has released Asterisk versions 1.4.20-rc3 and 1.6.0-beta9. These releases are intended to encourage community testing to improve the quality of the upcoming 1.4.20 and 1.6.0 releases. The testing process has proven extremely useful and we would like to thank everyone who has participated. Please help continue the effort. Any issues with test releases should be reported to http://bugs.digium.com/ or discussed on the asterisk-dev mailing list. Both releases are available for download from the Digium downloads site. http://downloads.digium.com/pub/telephony/asterisk/ Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] voicemail not sending emails
Roberto - I noticed in your original email you had the lines something like mailcmd=/opt/local/bin/msmtp -t ; --from blah AND serveremail=from=blah In mailcmd everything after the ; will be ignored as a comment In serveremail - well - it should throw an error... I would probably test by adding the --debug to the mailcmd and watch the logs. I also don't know mstmp but does it have a '-t' option? Brett Hi Brett msmtp is a stand-in for sendmail (using another SMTP server) so it has a -t option the real problem is that it never get called. even if I use the test mode: mailcmd=cat \ /tmp/astvm-mail to send the output to a file. Ciao Roberto ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Understanding Asterisk
I see. So how does Asterisk assign Lines to the various channels? I intend to have a few Aastra 480i's and these phones I believe have 4 line buttons on them, does the functionality of Asterisk in this scenario allow someone to see Line 1 is in use and either pickup the phone and attach to a free line or simply push Line 2 and attach to that next available line? Thanks so much guys! Jlc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Signorello Sent: Wednesday, May 14, 2008 9:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Understanding Asterisk I assume you are going to with a VOIP provider. Essentially, you have one DID and any number of channels/ports. Typically, you pay per port with a minute charge. Some people give you unlimited ports but charge a higher per minute fee. In you case, where you currently have 3 lines, you would need 3 channels. This would provide three concurrent calls to be in place. Asterisk does not know anything about channels in this example. If 3 calls come in it answers three calls. If a 4th caller comes in , the VOIP prvovider will send the busy signal to the caller. Asterisk does not see it. Joseph L. Casale wrote: I am about to order some DIDs for my first install but I am unclear on how Asterisk will function in either scenario with the two options I can order with. One option is the DID has unlimited connections. Another option for the DID is that it has a maximum of two concurrent calls only. How does Asterisk understand the multiple calls that are coming in and behave for both scenarios? The phone system we are trying to replace and therefore replicate the functionality is that of a very base Meridian system with 3 lines. Thanks for any guidance! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting CallerID UNKNOWN on an outgoing
Tobias Wolf wrote: Lets say i have a configured number range from 1000 to 1999 and 1000 is my base number. I make an outgoing call from a phone which sets its CallerID to 1500. Can anyone be so kind to tell me what is shown to the callee in either case? I can only tell you, that after I set exten = _0[23456789].,1,SetCallerPres(prohib) the callees phone displayed Caller (or in German Anrufer), which is just what I wanted. Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-biz] Announcing the first North America Druid Meetupshappening Chicago 22 May 2008 and Altanta 27 May 2008
Do you have 30 to 50 people in New York? We only tend to get about 10 people to the asterisk meetup events which is disappointing. Who else on the Asterisk list is based in NY that would like to catch up 31 May (6-9 pm) for a Druid event/General Asterisk event Regards, Dean Collins [EMAIL PROTECTED] Cognation Limited +1-212-203-4357 +61-2-9016-4652 (Sydney indial) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ming Yong Sent: Wednesday, 14 May 2008 12:52 PM To: Dean Collins Subject: Re: [asterisk-biz] [asterisk-users] Announcing the first North America Druid Meetupshappening Chicago 22 May 2008 and Altanta 27 May 2008 Dean, If you can help me with getting a cost effective venue in new york for about 30-50 people for 31 May (6-9 pm), let's do a Druid meetup. Ming On Wed, May 14, 2008 at 10:55 PM, Dean Collins [EMAIL PROTECTED] wrote: Ming, Are you coming to New York? Would be great to have an Asterisk related meetup here as well. Regards, Dean Collins [EMAIL PROTECTED] Cognation Limited +1-212-203-4357 +61-2-9016-4652 (Sydney indial) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ming Yong Sent: Wednesday, 14 May 2008 10:31 AM To: Commercial and Business-Oriented Asterisk Discussion; asterisk- [EMAIL PROTECTED] Subject: [asterisk-users] Announcing the first North America Druid Meetupshappening Chicago 22 May 2008 and Altanta 27 May 2008 Dear fellow Asterisk users, Voiceroute is proud to announce the first North America Druid Meetups happening in May 2008 in 2 cities (Chicago 22 May 08) and Atlanta (27 May 2008) Druid Meetups are basically fun demo sessions of Druid (Open Source Edition Unified Communications Server). Come and meet other Druid and asterisk users who are using Druid for enterprise communications deployments. Come join us for free pizzas and win goodies like Blackberry Curve 8320, free copies of Druid UCS! For our first meetups, we have the below cool stuff we will be demoing third party integrations with Druid Asterisk using the Druid SOAP API and how people can develop third party apps - Google Android Mobile application that talks to Druid Asterisk to get user call records like missed calls on the mobile - How to do click to call to any independent SugarCRM server - Tutorial on developing your own third party telephony CTI vertical applications related to Asterisk Druid Please join us by signing up free at the below URLs. Thanks again from the Voiceroute team. If you have any questions, please do not hesitate to email me at [EMAIL PROTECTED] Druid Chicago Meetup Sign up at http://druidchicago.eventbrite.com 22 May 2008, 6pm-9pm CST Location: Seiu73.org Offices 300 S Ashland Ave,Suite 400 Chicago, IL 60607 Organized by Druid User (Rajeev Varkey of Seiu73.org) Druid Atlanta Meetup Sign up at http://druidatlanta.eventbrite.com 27 May 2008, 6pm-9pm EST Location: Hilton Garden Inn Atlanta NE/Sugarloaf 2040 Sugarloaf Circle, Duluth GA 30097 Organized by Druid User (Jeffrey Thompson) -- Ming Yong CEO, www.voiceroute.org Druid - Open Source Unified Communications DID: +1-866-915-2407 ext 301 SIP/email: [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz -- Ming Yong CEO, www.voiceroute.org Druid - Open Source Unified Communications DID: +1-866-915-2407 ext 301 SIP/email: [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Understanding Asterisk
Those buttons are call appearances. They function based on how the phone is configured and how you program asterisk to process calls. For example, you have a phone that is ext #101, you have 4 call appearances on the phone device. You receive 2 phone calls within the span of 2 seconds, 2 of the call appearances will light up. You can answer one , put that call on hold and answer another. There are countless variations on this theme. The scenario you described is correct. John Signorello ispbx.com cogoblue.com Joseph L. Casale wrote: I see. So how does Asterisk assign Lines to the various channels? I intend to have a few Aastra 480i's and these phones I believe have 4 line buttons on them, does the functionality of Asterisk in this scenario allow someone to see Line 1 is in use and either pickup the phone and attach to a free line or simply push Line 2 and attach to that next available line? Thanks so much guys! Jlc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Signorello Sent: Wednesday, May 14, 2008 9:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Understanding Asterisk I assume you are going to with a VOIP provider. Essentially, you have one DID and any number of channels/ports. Typically, you pay per port with a minute charge. Some people give you unlimited ports but charge a higher per minute fee. In you case, where you currently have 3 lines, you would need 3 channels. This would provide three concurrent calls to be in place. Asterisk does not know anything about channels in this example. If 3 calls come in it answers three calls. If a 4th caller comes in , the VOIP prvovider will send the busy signal to the caller. Asterisk does not see it. Joseph L. Casale wrote: I am about to order some DIDs for my first install but I am unclear on how Asterisk will function in either scenario with the two options I can order with. One option is the DID has unlimited connections. Another option for the DID is that it has a maximum of two concurrent calls only. How does Asterisk understand the multiple calls that are coming in and behave for both scenarios? The phone system we are trying to replace and therefore replicate the functionality is that of a very base Meridian system with 3 lines. Thanks for any guidance! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-biz] Announcing the first North America Druid Meetupshappening Chicago 22 May 2008 and Altanta 27 May 2008
Anyone out there use Druid and can comment on it? I found out it was once closed source by the fact that they announced that it is now opensource. Before that, I had never heard of it. Usually I pick up on chatter if something is good. I see there are only 19 threads in the forums and some are not all that great while some are great. I also see great marketing words but lack of documentation or comparisons. No chatter on the users list expect a post or two kind of worries me. I just want to know because I have tested many different platforms. The only things I have seen go from closed to opensource like this are things that didn't sell. Take Pingtel for instance. Anyways, if someone that has used Druid OSE can comment on it, I would be very grateful. I am four hours from NYC so there are two reasons for asking. Thanks, Steve Totaro On Wed, May 14, 2008 at 1:02 PM, Dean Collins [EMAIL PROTECTED] wrote: Do you have 30 to 50 people in New York? We only tend to get about 10 people to the asterisk meetup events which is disappointing. Who else on the Asterisk list is based in NY that would like to catch up 31 May (6-9 pm) for a Druid event/General Asterisk event Regards, Dean Collins [EMAIL PROTECTED] Cognation Limited +1-212-203-4357 +61-2-9016-4652 (Sydney indial) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ming Yong Sent: Wednesday, 14 May 2008 12:52 PM To: Dean Collins Subject: Re: [asterisk-biz] [asterisk-users] Announcing the first North America Druid Meetupshappening Chicago 22 May 2008 and Altanta 27 May 2008 Dean, If you can help me with getting a cost effective venue in new york for about 30-50 people for 31 May (6-9 pm), let's do a Druid meetup. Ming On Wed, May 14, 2008 at 10:55 PM, Dean Collins [EMAIL PROTECTED] wrote: Ming, Are you coming to New York? Would be great to have an Asterisk related meetup here as well. Regards, Dean Collins [EMAIL PROTECTED] Cognation Limited +1-212-203-4357 +61-2-9016-4652 (Sydney indial) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ming Yong Sent: Wednesday, 14 May 2008 10:31 AM To: Commercial and Business-Oriented Asterisk Discussion; asterisk- [EMAIL PROTECTED] Subject: [asterisk-users] Announcing the first North America Druid Meetupshappening Chicago 22 May 2008 and Altanta 27 May 2008 Dear fellow Asterisk users, Voiceroute is proud to announce the first North America Druid Meetups happening in May 2008 in 2 cities (Chicago 22 May 08) and Atlanta (27 May 2008) Druid Meetups are basically fun demo sessions of Druid (Open Source Edition Unified Communications Server). Come and meet other Druid and asterisk users who are using Druid for enterprise communications deployments. Come join us for free pizzas and win goodies like Blackberry Curve 8320, free copies of Druid UCS! For our first meetups, we have the below cool stuff we will be demoing third party integrations with Druid Asterisk using the Druid SOAP API and how people can develop third party apps - Google Android Mobile application that talks to Druid Asterisk to get user call records like missed calls on the mobile - How to do click to call to any independent SugarCRM server - Tutorial on developing your own third party telephony CTI vertical applications related to Asterisk Druid Please join us by signing up free at the below URLs. Thanks again from the Voiceroute team. If you have any questions, please do not hesitate to email me at [EMAIL PROTECTED] Druid Chicago Meetup Sign up at http://druidchicago.eventbrite.com 22 May 2008, 6pm-9pm CST Location: Seiu73.org Offices 300 S Ashland Ave,Suite 400 Chicago, IL 60607 Organized by Druid User (Rajeev Varkey of Seiu73.org) Druid Atlanta Meetup Sign up at http://druidatlanta.eventbrite.com 27 May 2008, 6pm-9pm EST Location: Hilton Garden Inn Atlanta NE/Sugarloaf 2040 Sugarloaf Circle, Duluth GA 30097 Organized by Druid User (Jeffrey Thompson) -- Ming Yong CEO, www.voiceroute.org Druid - Open Source Unified Communications DID: +1-866-915-2407 ext 301 SIP/email: [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz -- Ming Yong CEO, www.voiceroute.org Druid - Open Source Unified Communications DID: +1-866-915-2407 ext 301 SIP/email: [EMAIL PROTECTED] ___
[asterisk-users] Fw: voicemail not sending emails
- Original Message -
From: gres [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 14, 2008 4:23 PM
Subject: Re: [asterisk-users] voicemail not sending emails
i think you have to have a mail transport agent like sendmail or postfix
installed and configured on your asterisk box , however if you forward the
mails to say hotmail or yahoo or gamil those servers will reject the mail
transfere
- Original Message -
From: Roberto Milani [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, May 13, 2008 4:48 PM
Subject: [asterisk-users] voicemail not sending emails
Hello list users
I have a very nice installation of asterisk on a mac mini.
Everything seems to work fine, call works, vm works, even message
transfer works but asterisk doesn't send any email.
this is my voicemail.conf:
[general]
mailcmd=/opt/local/bin/msmtp -t; [EMAIL PROTECTED]
;mailcmd=cat \ /tmp/asteriskvm-mail
format=wav
attach=yes
[EMAIL PROTECTED]
emailsubject=New message from ${VM_CALLERID}
emailbody=Hi, ${VM_NAME}!\n\nYou have a new message from $
{VM_CALLERID} in mailbox ${VM_MAILBOX}.
fromstring=My Telephone System
;max and min length of a message
maxmessage = 180
maxlogins = 3
[default]
100 = 4711,Front Desk,[EMAIL PROTECTED]
as you can see I'm using msmtp for mail and I tested it outside
asterisk an it works.
from the commented line you can se that I tried to cat the output to a
file but that never happens.
It really seems that asterisk don't send the emails.
any suggestions?
Thanks
Roberto
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[asterisk-users] OT: DRUID
Steve Totaro schrieb: Anyone out there use Druid and can comment on it? I found out it was once closed source by the fact that they announced that it is now opensource. Before that, I had never heard of it. Usually I pick up on chatter if something is good. I see there are only 19 threads in the forums and some are not all that great while some are great. I also see great marketing words but lack of documentation or comparisons. Here's what I found on Google: http://www.druid-project.eu/cln_007/Druid/EN/home/homepage__node.html?__nnn=true Welcome to DRUID ... Driving under the Influence of Drugs, Alcohol and Medicines lol Grüße, Philipp Kempgen -- Asterisk-Tag.org 2008, 26.-27. Mai - http://www.asterisk-tag.org amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-biz] Announcing the first North America Druid Meetupshappening Chicago 22 May 2008 and Altanta 27 May 2008
Tried to install it on a dev box that had been running Trixbox. Kernel panic midway through install. Happened twice in a row, so we gave up. I've heard some people really like it, which is why we wanted to have a look, but no joy for us. N. Steve Totaro wrote: Anyone out there use Druid and can comment on it? I found out it was once closed source by the fact that they announced that it is now opensource. Before that, I had never heard of it. Usually I pick up on chatter if something is good. I see there are only 19 threads in the forums and some are not all that great while some are great. I also see great marketing words but lack of documentation or comparisons. No chatter on the users list expect a post or two kind of worries me. I just want to know because I have tested many different platforms. The only things I have seen go from closed to opensource like this are things that didn't sell. Take Pingtel for instance. Anyways, if someone that has used Druid OSE can comment on it, I would be very grateful. I am four hours from NYC so there are two reasons for asking. Thanks, Steve Totaro On Wed, May 14, 2008 at 1:02 PM, Dean Collins [EMAIL PROTECTED] wrote: Do you have 30 to 50 people in New York? We only tend to get about 10 people to the asterisk meetup events which is disappointing. Who else on the Asterisk list is based in NY that would like to catch up 31 May (6-9 pm) for a Druid event/General Asterisk event Regards, Dean Collins [EMAIL PROTECTED] Cognation Limited +1-212-203-4357 +61-2-9016-4652 (Sydney indial) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ming Yong Sent: Wednesday, 14 May 2008 12:52 PM To: Dean Collins Subject: Re: [asterisk-biz] [asterisk-users] Announcing the first North America Druid Meetupshappening Chicago 22 May 2008 and Altanta 27 May 2008 Dean, If you can help me with getting a cost effective venue in new york for about 30-50 people for 31 May (6-9 pm), let's do a Druid meetup. Ming On Wed, May 14, 2008 at 10:55 PM, Dean Collins [EMAIL PROTECTED] wrote: Ming, Are you coming to New York? Would be great to have an Asterisk related meetup here as well. Regards, Dean Collins [EMAIL PROTECTED] Cognation Limited +1-212-203-4357 +61-2-9016-4652 (Sydney indial) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ming Yong Sent: Wednesday, 14 May 2008 10:31 AM To: Commercial and Business-Oriented Asterisk Discussion; asterisk- [EMAIL PROTECTED] Subject: [asterisk-users] Announcing the first North America Druid Meetupshappening Chicago 22 May 2008 and Altanta 27 May 2008 Dear fellow Asterisk users, Voiceroute is proud to announce the first North America Druid Meetups happening in May 2008 in 2 cities (Chicago 22 May 08) and Atlanta (27 May 2008) Druid Meetups are basically fun demo sessions of Druid (Open Source Edition Unified Communications Server). Come and meet other Druid and asterisk users who are using Druid for enterprise communications deployments. Come join us for free pizzas and win goodies like Blackberry Curve 8320, free copies of Druid UCS! For our first meetups, we have the below cool stuff we will be demoing third party integrations with Druid Asterisk using the Druid SOAP API and how people can develop third party apps - Google Android Mobile application that talks to Druid Asterisk to get user call records like missed calls on the mobile - How to do click to call to any independent SugarCRM server - Tutorial on developing your own third party telephony CTI vertical applications related to Asterisk Druid Please join us by signing up free at the below URLs. Thanks again from the Voiceroute team. If you have any questions, please do not hesitate to email me at [EMAIL PROTECTED] Druid Chicago Meetup Sign up at http://druidchicago.eventbrite.com 22 May 2008, 6pm-9pm CST Location: Seiu73.org Offices 300 S Ashland Ave,Suite 400 Chicago, IL 60607 Organized by Druid User (Rajeev Varkey of Seiu73.org) Druid Atlanta Meetup Sign up at http://druidatlanta.eventbrite.com 27 May 2008, 6pm-9pm EST Location: Hilton Garden Inn Atlanta NE/Sugarloaf 2040 Sugarloaf Circle, Duluth GA 30097 Organized by Druid User (Jeffrey Thompson) -- Ming Yong CEO, www.voiceroute.org Druid - Open Source Unified Communications DID: +1-866-915-2407 ext 301 SIP/email: [EMAIL PROTECTED] ___ -- Bandwidth and
Re: [asterisk-users] [asterisk-biz] Announcing the first North America Druid Meetupshappening Chicago 22 May 2008 and Altanta 27 May 2008
On Wed, May 14, 2008 at 7:50 PM, Steve Totaro [EMAIL PROTECTED] wrote: Anyone out there use Druid and can comment on it? I found out it was I don't use it per se, but afyter a conference with Voiceroute, I promised to install it and I did so on a test box. The install was great and the auto-detect of Digium hardware was very cool. I haven't had enough time to actually turn the machine back on since, but I think they've done some stuff very well. Problem is, there are so many flavors available these days, it's hard to try them all for any length of time. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-biz] Announcing the first North America Druid Meetupshappening Chicago 22 May 2008 and Altanta 27 May 2008
On Wed, May 14, 2008 at 3:03 PM, randulo [EMAIL PROTECTED] wrote: On Wed, May 14, 2008 at 7:50 PM, Steve Totaro [EMAIL PROTECTED] wrote: Anyone out there use Druid and can comment on it? I found out it was I don't use it per se, but afyter a conference with Voiceroute, I promised to install it and I did so on a test box. The install was great and the auto-detect of Digium hardware was very cool. I haven't had enough time to actually turn the machine back on since, but I think they've done some stuff very well. Problem is, there are so many flavors available these days, it's hard to try them all for any length of time. /r This is *exactly* where I am. It installed fine on an HP DL380 and Digium TDM400P I had laying around and looked good, but I am interested in some real world testimonials. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Understanding Asterisk
You could do it that way but there is really no need. If you are getting DIDs you can just have them ring a certain phone, a group of phones, an application, a queue.. You can just abandon the whole notion of Lines. Thanks, Steve Totaro On Wed, May 14, 2008 at 12:44 PM, Joseph L. Casale [EMAIL PROTECTED] wrote: I see. So how does Asterisk assign Lines to the various channels? I intend to have a few Aastra 480i's and these phones I believe have 4 line buttons on them, does the functionality of Asterisk in this scenario allow someone to see Line 1 is in use and either pickup the phone and attach to a free line or simply push Line 2 and attach to that next available line? Thanks so much guys! Jlc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Signorello Sent: Wednesday, May 14, 2008 9:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Understanding Asterisk I assume you are going to with a VOIP provider. Essentially, you have one DID and any number of channels/ports. Typically, you pay per port with a minute charge. Some people give you unlimited ports but charge a higher per minute fee. In you case, where you currently have 3 lines, you would need 3 channels. This would provide three concurrent calls to be in place. Asterisk does not know anything about channels in this example. If 3 calls come in it answers three calls. If a 4th caller comes in , the VOIP prvovider will send the busy signal to the caller. Asterisk does not see it. Joseph L. Casale wrote: I am about to order some DIDs for my first install but I am unclear on how Asterisk will function in either scenario with the two options I can order with. One option is the DID has unlimited connections. Another option for the DID is that it has a maximum of two concurrent calls only. How does Asterisk understand the multiple calls that are coming in and behave for both scenarios? The phone system we are trying to replace and therefore replicate the functionality is that of a very base Meridian system with 3 lines. Thanks for any guidance! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question about SS7
Hi, I have read about SS7 recently and learnt that it is a signalling protocol used in PSTN for call management, setup, etc. The thing that I don't understand is how SS7 plays a role in VOIP. When I make calls between landline and Asterisk via PSTN, I don't need to do anything with SS7. Is it because the SS7 signalling is already done by Asterisk already? From the prespective of implementing Asterisk, what kind of SS7 support is needed? Is SS7 something needs to be concerned about when using Asterisk with T1/E1? I hope someone can help me to clearify these doubts that I am having. Thanks, Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] anyone from Joplin, MO
I'm trying to convince my employer to deploy an Asterisk based system, but one member of the leadership team is against it. The rest of the team is for it, but he's convinced them that we should find other organisations in the Joplin, MO area who are using Asterisk first because, we don't want to be the first in our area. I can't fathom how that's relevant, since I have shown them case studies from much larger organisations than us around the country, and personally talked with other organisations using Asterisk, and found one small business with about 10 phones in Joplin, MO and talked to them, but he's insistent we find an organisation with 50-75 or more phones in the Joplin, MO area using Asterisk before we go ahead. So, is there anyone out there from the Joplin, MO area using Asterisk, or do you know of someone who is? If so, please contact me so I can go forward with this project. Thanks, Bryson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] anyone from Joplin, MO
Bryson Medlock wrote: I'm trying to convince my employer to deploy an Asterisk based system, but one member of the leadership team is against it. The rest of the team is for it, but he's convinced them that we should find other organisations in the Joplin, MO area who are using Asterisk first because, we don't want to be the first in our area. This is such a dangerous thing - that one team member must have some influence or control. I would find out what his concerns are, and attend to those first, before finding a relevant organisation. Otherwise he/she may just be using them as ammunition on why _not_ to use Asterisk. Julian I can't fathom how that's relevant, since I have shown them case studies from much larger organisations than us around the country, and personally talked with other organisations using Asterisk, and found one small business with about 10 phones in Joplin, MO and talked to them, but he's insistent we find an organisation with 50-75 or more phones in the Joplin, MO area using Asterisk before we go ahead. So, is there anyone out there from the Joplin, MO area using Asterisk, or do you know of someone who is? If so, please contact me so I can go forward with this project. Thanks, Bryson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] anyone from Joplin, MO
Tell your Employer to have a little faith. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bryson Medlock Sent: Wednesday, May 14, 2008 3:40 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] anyone from Joplin, MO I'm trying to convince my employer to deploy an Asterisk based system, but one member of the leadership team is against it. The rest of the team is for it, but he's convinced them that we should find other organisations in the Joplin, MO area who are using Asterisk first because, we don't want to be the first in our area. I can't fathom how that's relevant, since I have shown them case studies from much larger organisations than us around the country, and personally talked with other organisations using Asterisk, and found one small business with about 10 phones in Joplin, MO and talked to them, but he's insistent we find an organisation with 50-75 or more phones in the Joplin, MO area using Asterisk before we go ahead. So, is there anyone out there from the Joplin, MO area using Asterisk, or do you know of someone who is? If so, please contact me so I can go forward with this project. Thanks, Bryson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail not sending emails
Roberto Milani wrote: Roberto - I noticed in your original email you had the lines something like mailcmd=/opt/local/bin/msmtp -t ; --from blah AND serveremail=from=blah In mailcmd everything after the ; will be ignored as a comment In serveremail - well - it should throw an error... I would probably test by adding the --debug to the mailcmd and watch the logs. I also don't know mstmp but does it have a '-t' option? Brett Hi Brett msmtp is a stand-in for sendmail (using another SMTP server) so it has a -t option the real problem is that it never get called. even if I use the test mode: mailcmd=cat \ /tmp/astvm-mail to send the output to a file. Ciao Roberto ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Are you relaying the mail through your isp? Are you using a system wide /etc/msmtprc or for user asterisk ~.msmtprc -- Powered by Gentoo GNU/Linux http://linuxcrazy.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about SS7
SS7 does NOT play a roll in VoIP. The SS7 signaling that you are describing is not really SS7 but signaling over a PRI using ISDN that your provider uses to exchange information via SS7 to the other carriers. To be blunt and I do not mean to be condescending in any way, but, if you are using Asterisk and do not know what SS7 is, you don't need to worry about it. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mark morreny Sent: Wednesday, May 14, 2008 3:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Question about SS7 Hi, I have read about SS7 recently and learnt that it is a signalling protocol used in PSTN for call management, setup, etc. The thing that I don't understand is how SS7 plays a role in VOIP. When I make calls between landline and Asterisk via PSTN, I don't need to do anything with SS7. Is it because the SS7 signalling is already done by Asterisk already? From the prespective of implementing Asterisk, what kind of SS7 support is needed? Is SS7 something needs to be concerned about when using Asterisk with T1/E1? I hope someone can help me to clearify these doubts that I am having. Thanks, Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail not sending emails
does the /tmp directory need to have some specific kind of mode/
ownership?
mine is linked to /private/tmp and is lrwxr-xr-x root admin
Ciao
Roberto
On May 14, 2008, at 8:34 PM, Roberto Milani wrote:
That's what I thought,
and my voicemail.conf is:
[general]
format=wav
attach=yes
serveremail= [EMAIL PROTECTED]
emailsubject=New message from ${VM_CALLERID}
emailbody=Hi, ${VM_NAME}!\n\nYou have a new message from $
{VM_CALLERID} in mailbox ${VM_MAILBOX}.
fromstring=My Telephone System
;max and min length of a message
maxmessage = 180
maxlogins = 3
[default]
100 = 4711,Front Desk,[EMAIL PROTECTED],,attach=yes
the voicemail works, I get also the MWI working perfectly
but no email
Roberto
On May 14, 2008, at 6:37 PM, Tilghman Lesher wrote:
On Wednesday 14 May 2008 19:45:13 Roberto Milani wrote:
Good hint but I tested that too
I sent the command line to the link called sendmail and I got my
mail
just right
is there any other configuration in asterisk that might prevent it
to
send mails?
The only reason why it wouldn't send an email is if an email address
is
not configured (third field in voicemail.conf, email column in
realtime):
123 = 456,Firstname Lastname,[EMAIL PROTECTED],[EMAIL PROTECTED]
The exact command that is run is:
sh -c ( /usr/sbin/sendmail -t /tmp/astmail-123456 ;
rm -f /tmp/astmail-123456 )
Or whatever you've substituted for /usr/sbin/sendmail -t.
--
Tilghman
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Re: [asterisk-users] Portability in Asterisk
Aadil, If the answers are not suitable for you, you might want to check out freeswitch. Thanks, Steve Totaro On Wed, May 14, 2008 at 11:30 PM, Paul Hales [EMAIL PROTECTED] wrote: Dear Aadil, You asked this question about 1 month ago, and received several response. Were you unhappy with the responses you received? PaulH On Wed, 2008-04-30 at 10:50 +0530, Aadilkhan Maniyar wrote: Hi All, I have a query with respect to Asterisk Portability. I would like to know about the different OS and Hardware that Asterisk can be run on. Any info regarding the above would be very much helpful. Regards, Aadil ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail not sending emails
First of all thanks to everybody I feel the need to clarify the configuration. from the command line msmtp works, this means that ~.msmtrc is configured properly I removed the mailcmd line from voicemail.conf , renamed sendmail to sendmail.orig and created a link to msmtp called sendmail in /usr/sbin/ lrwxr-xr-x1 root wheel 20 May 14 07:28 sendmail - /opt/local/bin/msmtp the command: echo hello. | sendmail --debug --account=sbcglobal [EMAIL PROTECTED] [EMAIL PROTECTED] sends the email I do have voicemail configured in voicemail.conf with valid email addresses, voicemails work fine this is the my tmp lrwxr-xr-x@ 1 root admin11 Nov 15 06:54 tmp - private/tmp and I have no emails no error messages, no logs, nothing any idea on how to debug this? On May 14, 2008, at 9:01 PM, Jose Flores Galicia wrote: That's right. msmtp behave different depending on the user that invokes de command. I suppose you are running asterisk like root or asterisk user so the config file must be /root/.msmtprc or /home/asterisk/.msmtprc. Also, as david notice, in the voicemail.conf line mailcmd=/opt/local/bin/msmtp -t ; --from blah all that comes after ; are ignored when parsed by asterisk. I suggest you to configure a default account on your System configuration file for msmtp, you can found the path to the file if you make msmtp --version from command line. This is how my configuration file looks like: #Config file for msmpt #Default values for all accounts defaults logfile /var/log/msmtp.log # Main Account account aspinet host mail.megamailservers.com from [EMAIL PROTECTED] auth MD5 user [EMAIL PROTECTED] password mostseecretpassword # Set a default account account default : aspinet Adn in voicemail.conf add mailcmd=/usr/bin/msmtp -t Also you can try to configure sendmail for smtp relay with your ISP This doc was very useful when I try it. http://cri.ch/linux/docs/sk0009.html Regards 2008/5/14 Tilghman Lesher [EMAIL PROTECTED]: On Wednesday 14 May 2008 17:19:09 Roberto Milani wrote: I do have a mail transport agent configured It is msmtp and it is working just fine I tested it on the command line and I receive the test email I have a link from sendmail pointing to msmtp. but it never get called. I've noted that the times that you've tested this, you've used msmtp on the command line. Some commands behave differently if you call them with a different name. For example, if sh is linked to bash and you call it, bash drops some of its features to more closely match sh. Could it be that msmtp acts the same way, and you need to test that behavior (calling it as sendmail)? -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jose Flores Galicia [EMAIL PROTECTED] BriefCode Code Based Training ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Listen And Talk mode differentiation of meetme() conference
Hello users, i am trying to setup a conference system and i have following requirement 1)some users are only in listen mode 2)some users are only in talk mode 3)some users are able to do both talk and listen how to diffrentiate them when they enter into a particular mode? meaning do i have to give a separate access number in my extensions.conf file so that i will bridge them all together in once coference using meetme() or is there any separate way to do that my idea is like this one 1)all listen only users can call on 123 exten = 123,1,MeetMe(|Mm) exten = 123,2,Hangup() 2)all talkers can call on 456 exten = 456,1,MeetMe(|Mt) exten = 456,2,Hangup() 3)both talk and listen users can call on 789 exten = 789,1,MeetMe(|M) exten = 789,2,Hangup() does this setup only works? or is there any other method of doing the things just enlighten me so that i can finalize my setup thanks in advance regards srinvias antarvedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk for larg
gmail wrote: Does anybody know how to off-load an Asterisk Box so that to distribute its functions like IVR and VoiceMail or its PTSN gateway function into different servers? in this case , will the installation of Asterisk on each server differe and how these different servers will interact as a single logical -vs physical- server? thx alot What do you mean how? Are you asking if Asterisk has a built-in clustering mechanism somewhere in its own application stack? The answer is no. Otherwise, the answer to how to distribute Asterisk is to ... distribute Asterisk. No, seriously; split up the users according to multiple servers and/or assign them dynamically, route using a SIP proxy functioning as a load balancer, and other things you do when setting up some sort of farm without extensive built-in parallelisation options. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday May 16th @1 Noon EDT: VoIP Users Conference is about Click 2 Call
Basic info site: http://VoipUsersConference.org Hi, What if you could connect people or businesses without having them require hardware or software of any kind? I've always been interested in this idea and now it's a reality with several choices to investigate. We've spoken to Yusuf Motiwala, TringMe's CEO before on a call. TringMe has recently announced an expanded system and API to make unified messaging something anyone with asterisk can accomplish. Here's the latest on what TringMe is up to: http://blog.tringme.com/tringme-announces-the-availability-of-mobilevoip-application/ Ted Gibson of 1EZPhone.com has also been on previous calls and they too are planning on offering solutions in this area. http://1ezphone.com/ is where you see what they're up to. Friday's call would be a good place to ask any questions you may have about C2C (Click to Call), beginning with Why do we care? and moving forward with How can we implement it? PSTN: Call (724) 444-7444 and enter 22622# 1# (or your PIN or set callerid to your PIN) SIP asterisk exten = 1234,1, NoOp(Calling Talkshoe conf bridge) exten = 1234,n, Dial(SIP/[EMAIL PROTECTED],60,D22622#1#)) or phone sip:[EMAIL PROTECTED] More info IRC channel on Freenode.net #voip-users-conference RSS Feed for past conferences http://feeds.feedburner.com/AstUser Forums, blogs, scheduling, social network: http://food4wine.ning.com Short URL to post to friends who may be interested: http://x2z.eu Finally, you can look at all conference archives on a single page: http://food4wine.ning.com/conference ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
