Re: [asterisk-users] 12 port FXx PCI card
Yusuf, I am using this card and it works very well for me. To use it you need to download a driver addition and recompile Zaptel. Not a big problem really. In all other aspects it works like the Digium card. Peter Yusuf wrote: Hi, http://www.openvox.com.cn/products_detail.php?genre_id=17id=45 The A1200P is a 12 port card, that used the same modules as a TDM400P. I have been looking at this card, and I want to know if anybody has used this card and what their experiences were? thanks, yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] two SIP phones as one line
The quirk of your old PBX is in fact exactly what happens when you put any two analog phones on the same line. The easiest way to duplicate this is to connect another analog phone to your ATA. Some analog phones can indicate when the other is on the line and can put a call on hold locally. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Hi, I am looking to replace a quirk of our old PBX system functionality with asterisk but after searching, archives, wiki, etc.. I cannot figure out how. Here is what I would like to do: PhoneA is a SIP hard phone, phoneB is an analogue phone connected to a SIP ATA. When an incoming call comes in, I would like to ring both phones, but if phoneA is answered first, I would like phoneB to be answered as well and left in a off hook state so that when someone picks up the receiver of phoneB, they can hear and participate in the conversation between the calling party and phoneA. I believe I would have to put both phones in a MeetMe conference, but how to I auto-answer phoneB when phoneA has answered the call? I suspect that this may not be possible with asterisk, but would like confirmation of that. Thanks in advance. -m ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] two SIP phones as one line
The quirk of your old PBX is in fact exactly what happens when you put any two analog phones on the same line. The easiest way to duplicate this is to connect another analog phone to your ATA. Some analog phones can indicate when the other is on the line and can put a call on hold locally. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Hi, I am looking to replace a quirk of our old PBX system functionality with asterisk but after searching, archives, wiki, etc.. I cannot figure out how. Here is what I would like to do: PhoneA is a SIP hard phone, phoneB is an analogue phone connected to a SIP ATA. When an incoming call comes in, I would like to ring both phones, but if phoneA is answered first, I would like phoneB to be answered as well and left in a off hook state so that when someone picks up the receiver of phoneB, they can hear and participate in the conversation between the calling party and phoneA. I believe I would have to put both phones in a MeetMe conference, but how to I auto-answer phoneB when phoneA has answered the call? I suspect that this may not be possible with asterisk, but would like confirmation of that. Thanks in advance. -m ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] two SIP phones as one line
On Sun, 2006-15-10 at 05:09 -0400, Henry.L.Coleman wrote: The quirk of your old PBX is in fact exactly what happens when you put any two analog phones on the same line. The easiest way to duplicate this is to connect another analog phone to your ATA. Some analog phones can indicate when the other is on the line and can put a call on hold locally. In fact no, I should have explained better, but in the old system one phone was analogue and the other was a multi-line digital Nortel Meridian phone. The one phone has to be analogue because it interfaces with a radio broadcast phone patch. -m Hi, I am looking to replace a quirk of our old PBX system functionality with asterisk but after searching, archives, wiki, etc.. I cannot figure out how. Here is what I would like to do: PhoneA is a SIP hard phone, phoneB is an analogue phone connected to a SIP ATA. When an incoming call comes in, I would like to ring both phones, but if phoneA is answered first, I would like phoneB to be answered as well and left in a off hook state so that when someone picks up the receiver of phoneB, they can hear and participate in the conversation between the calling party and phoneA. I believe I would have to put both phones in a MeetMe conference, but how to I auto-answer phoneB when phoneA has answered the call? I suspect that this may not be possible with asterisk, but would like confirmation of that. Thanks in advance. -m signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail issue
Check for the existence of INBOX and OLD folders in the VM folders in /var/spool/asterisk/voicemailI messed around with something and found this out the hard way.Let me know if this works, as I am curious.-Rob.On Oct 10, 2006, at 12:58 PM, stan ford wrote:the last thing i was trying to do was change the default password to same as voicemail. i also tried reversing these changes but doesnt work. this is my log. i should probably mention that im running trixbox 1.21. when i connect to the voicemail system remotely, i enter the username, then a password and thats when this comes up. Core debug is at least 1 -- Executing Macro("Local/[EMAIL PROTECTED],2", "hangupcall") in new stack -- Executing ResetCDR("Local/[EMAIL PROTECTED],2", "w") in new stack -- Executing NoCDR("Local/[EMAIL PROTECTED],2", "") in new stack -- Executing Wait("Local/[EMAIL PROTECTED],2", "5") in new stack -- Executing Hangup("Local/[EMAIL PROTECTED],2", "") in new stack How low will we go? Check out Yahoo! Messenger’s low PC-to-Phone call rates.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___
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Re: [asterisk-users] two SIP phones as one line
Are your sip phones capable of auto-answer?I can imagine you can terminate the incoming call into a meet-me conference (no pass code) and then trigger a script that creates a call file for each of the other participating phones. The auto-answer part seems like the sticky part. On 10/15/06, Marc Heckmann [EMAIL PROTECTED] wrote: On Sun, 2006-15-10 at 05:09 -0400, Henry.L.Coleman wrote: The quirk of your old PBX is in fact exactly what happens when you put any two analog phones on the same line. The easiest way to duplicate this is to connect another analog phone to your ATA. Some analog phones can indicate when the other is on the line and can put a call on hold locally.In fact no, I should have explained better, but in the old system one phone was analogue and the other was a multi-line digital NortelMeridian phone. The one phone has to be analogue because it interfaceswith a radio broadcast phone patch.-m Hi, I am looking to replace a quirk of our old PBX system functionality with asterisk but after searching, archives, wiki, etc.. I cannot figure out how. Here is what I would like to do: PhoneA is a SIP hard phone, phoneB is an analogue phone connected to a SIP ATA. When an incoming call comes in, I would like to ring both phones, but if phoneA is answered first, I would like phoneB to be answered as well and left in a off hook state so that when someone picks up the receiver of phoneB, they can hear and participate in the conversation between the calling party and phoneA. I believe I would have to put both phones in a MeetMe conference, but how to I auto-answer phoneB when phoneA has answered the call? I suspect that this may not be possible with asterisk, but would like confirmation of that. Thanks in advance. -m ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] two SIP phones as one line
Hi Mark - PhoneA is a SIP hard phone, phoneB is an analogue phone connected to a SIP ATA. When an incoming call comes in, I would like to ring both phones, but if phoneA is answered first, I would like phoneB to be answered as well and left in a off hook state so that when someone picks up the receiver of phoneB, they can hear and participate in the conversation between the calling party and phoneA. I believe I would have to put both phones in a MeetMe conference, but how to I auto-answer phoneB when phoneA has answered the call? Two Questions: 1. On the SIP phone, will this special conference function be needed on both incoming and outgoing calls, or just one of those? 2. Does the analog phone have to do anything else? Should it work like a normal phone when it's not doing this special conference function? You should be able to do this, like you said, by dumping both phones into a meetme conference. There would be two tricky things here A) getting the analog phone to automatically go to a meetme conference whenever it is off-hook, B) getting outgoing calls from the sip phone into a meetme conference (incoming calls would be easy). I think A) is probably not possible, given that you are using an external ATA device. That device would somehow have to send the off-hook status back to asterisk via sip messages (I think there's actually a bounty for this). This should be possible if you were using an internal zaptel card rather than an external gateway. If the answer to question 2) above is yes, you would have other problems, too. A good compromise to the problems of both 2) and A) would be to put the analog phone into a special context where you'd have a one digit press for each function (e.g. press 1 for normal phone, press 2 for conference). Still, unless these users are really ornery, I'd probably just make them learn to transfer and dial into a conference. - Noah On 10/15/06, Marc Heckmann [EMAIL PROTECTED] wrote: On Sun, 2006-15-10 at 05:09 -0400, Henry.L.Coleman wrote: The quirk of your old PBX is in fact exactly what happens when you put any two analog phones on the same line. The easiest way to duplicate this is to connect another analog phone to your ATA. Some analog phones can indicate when the other is on the line and can put a call on hold locally. In fact no, I should have explained better, but in the old system one phone was analogue and the other was a multi-line digital Nortel Meridian phone. The one phone has to be analogue because it interfaces with a radio broadcast phone patch. -m Hi, I am looking to replace a quirk of our old PBX system functionality with asterisk but after searching, archives, wiki, etc.. I cannot figure out how. Here is what I would like to do: PhoneA is a SIP hard phone, phoneB is an analogue phone connected to a SIP ATA. When an incoming call comes in, I would like to ring both phones, but if phoneA is answered first, I would like phoneB to be answered as well and left in a off hook state so that when someone picks up the receiver of phoneB, they can hear and participate in the conversation between the calling party and phoneA. I believe I would have to put both phones in a MeetMe conference, but how to I auto-answer phoneB when phoneA has answered the call? I suspect that this may not be possible with asterisk, but would like confirmation of that. Thanks in advance. -m ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] two SIP phones as one line
On Sun, 2006-15-10 at 12:11 -0400, Noah Miller wrote: Hi Mark - PhoneA is a SIP hard phone, phoneB is an analogue phone connected to a SIP ATA. When an incoming call comes in, I would like to ring both phones, but if phoneA is answered first, I would like phoneB to be answered as well and left in a off hook state so that when someone picks up the receiver of phoneB, they can hear and participate in the conversation between the calling party and phoneA. I believe I would have to put both phones in a MeetMe conference, but how to I auto-answer phoneB when phoneA has answered the call? Two Questions: 1. On the SIP phone, will this special conference function be needed on both incoming and outgoing calls, or just one of those? just for incoming calls. 2. Does the analog phone have to do anything else? Should it work like a normal phone when it's not doing this special conference function? yes. it should work like a normal phone otherwise. You should be able to do this, like you said, by dumping both phones into a meetme conference. There would be two tricky things here A) getting the analog phone to automatically go to a meetme conference whenever it is off-hook, B) getting outgoing calls from the sip phone into a meetme conference (incoming calls would be easy). I think A) is probably not possible, given that you are using an external ATA device. That device would somehow have to send the off-hook status back to asterisk via sip messages (I think there's actually a bounty for this). This should be possible if you were using an internal zaptel card rather than an external gateway. If the answer to question Actually I think I might be able to get it to auto-dial into a conference when it is off hook. In any case, using a zaptel card is also an option. 2) above is yes, you would have other problems, too. ok, maybe it can check if the conference exists and if not simply act as a normal phone? A good compromise to the problems of both 2) and A) would be to put the analog phone into a special context where you'd have a one digit press for each function (e.g. press 1 for normal phone, press 2 for conference). Still, unless these users are really ornery, I'd probably just make them learn to transfer and dial into a conference. The users (there are many of them) have a hard time adapting to change. -m signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Generate Random Numbers in dialplan
Thanks everyone for your help. The agi script works as well as RAND (with the latest trunk version of asterisk). -Jon - Original Message - From: Steve Murphy [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, October 14, 2006 9:03 PM Subject: [asterisk-users] Re: Generate Random Numbers in dialplan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Where is the PlayDTMF command?
When I apply the patch, this is the error I obtain. [EMAIL PROTECTED] apps]# patch -p0 ../ast_trunk_manager_PlayDTMF.patch patching file [EMAIL PROTECTED] apps]# patch -p0 ../ast_trunk_manager_PlayDTMF.patch patching file app_senddtmf.c Hunk #1 succeeded at 41 (offset -2 lines). patch: malformed patch at line 28: ast_get_channel_by_name_locked(channel); I am not that familiar with patch and the source of app_senddtmf.c may have changed since you submitted it. Is there some way to interpret the instructions and apply it by hand? On 10/13/06, Moises Silva [EMAIL PROTECTED] wrote: Is there something else I am missing Learning how to patch? :) It seems you are applying incorrectly the patch, remember to use the -p option, as suggested by the error message. From reading the error you posted, it seems to me that you need to # cd asterisk-1.2.12.1/apps/ # patch -p0 ../../ast_trunk_manager_PlayDTMF.patch Regards On 10/13/06, Frank Church [EMAIL PROTECTED] wrote: When I try to apply this patch - ast_trunk_manager_PlayDTMF.patch - I receive the error below missing header for unified diff at line 3 of patch can't find file to patch at input line 3 Perhaps you used the wrong -p or --strip option? The text leading up to this was: -- |--- app_senddtmf.c~2006-05-04 15:27:41.0 -0500 |+++ app_senddtmf.c 2006-05-04 15:29:21.0 -0500 -- File to patch: Is there something else I am missing On 10/12/06, Frank Church [EMAIL PROTECTED] wrote: Hi Moises, I have looked on that page and there is no link to click on to download the patch. There are references to deletions to some patch files on the page but nothing to click on. I have seen a link at http://bugs.digium.com/view.php?id=6990, asterisk-svn-21231-DTMF_event.patch. Would that be the one? Frank On 10/12/06, Moises Silva [EMAIL PROTECTED] wrote: Hi Frank, I sent a patch updated here: http://bugs.digium.com/view.php?id=6082 But that was some months ago, I havent seen a bugmarshall for a while there, so I keep patching my own Asterisk for several stuff. New features are never added to release branches, so you need to patch 1.2.12.1 adapting the trunk patch. Dont worry, is an easy patch. Regards On 10/11/06, Frank Church [EMAIL PROTECTED] wrote: Hi Moises, Ignore my last reply about the presence of the DTMF in 1.4. Do you have the source for patching the DTMF event? There is no link to it on the bug6082 page, and I am not quite sure how it can be obtained from SVN. Regards Richard On 10/12/06, Frank Church [EMAIL PROTECTED] wrote: Hi Moises, does the you mentioned earlier at http://galileo.ivsol.net/play_dtmf-1.2.12.1.patch include the DTMF event, or is it for PlayDTMF and SendDTMF? Looking through the actions on bug6082 it is hard to tell whether the DTMF event patch is still in there when I last compiled that branch the DTMF event was not coming up. Is the DTMF patch incorporated into 1.2.12.1, or will I need to apply the original patch to 1.2.12.1? On 10/10/06, Moises Silva [EMAIL PROTECTED] wrote: Jan, im sorry to get back to you so late, ive been busy. It seems i sent you an incorrect patch I was testing, but I have found the correct patch in mantis: http://bugs.digium.com/view.php?id=6682 Please be aware that the patch I sent you initially used a funciton that received 1 or more DTMF digits, and thats why it fails, because the operation need to be fast enough to not lock the channel more time than allowed, so the patch you can find now in mantis, use a function that only accepts 1 DTMF digit at time, so PlayDTMF only accepts 1 digit to, you need to call it several times to send a DTMF stream. Regards On 10/9/06, Jan du Toit [EMAIL PROTECTED] wrote: So I patch my asterisk (version 1.2.12.1) with the patch given by Moises. http://galileo.ivsol.net/play_dtmf-1.2.12.1.patch Thanks Moises. When I type in show manager command PlayDTMF it is their. With the show manager commands it is not within the list containing all the commands. When I execute the manager PlayDTMF action, the manager response says DTMF successfully queued. I don't hear anything on the phone, when I look at the CLI I see the following warning message. Its produced everytime I execute the PlayDTMF action. Oct 6 09:31:06 WARNING[3449]: channel.c:1610 ast_waitfor_nandfds: Thread 294931 Blocking 'SIP/Jan-081ba140', already blocked by thread 360468 in procedure ast_waitfor_nandfds Am I doing something wrong? Is this a bug? Please help, I need this to work as soon as possible... Thanks
Re: [asterisk-users] How do you like TrixBox?
Yes but they will never understand the configs. They need to learn step by step. - Original Message - From: joe, at j4computers [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, October 13, 2006 4:11 PM Subject: Re: [asterisk-users] How do you like TrixBox? Dovid B[EMAIL PROTECTED] Wrote on: 10/13/2006 9:51 AM: . . . A)If something goes wrong they wont know where to start. They only know the GUI. B)They will never know the real way of working asterisk.. . . But, can't it be one way of learning? Can't one setup and modify a Trixbox setup, then peruse the conf files, to get familiar with (almost) all things Asterisk? Spoke as one who was not very pleased with their own foray into Trixbox and is still creeping up to speed on Asterisk. joe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Student Research - Asterisk H323 Video
I am currently doing my thesis on an implementation of Video into Asterisk using H323. So I know that they are various mailing lists that demonstrate that SIP is the way forward, but sometimes It helps to use old equipment that one already owns. so I am just looking for some simple ideas as to if possible Provide a quick and simple method of passing video through asterisk between 2 softphones. I currently have the following Channels installed on the various systems. Fedora Core 2 - Asterisk-0h323 on Asterisk 1.1.00 Fedora Core 5 - Asterisk h323 on Asterisk 1.2.12 Fedora Core 5 - Asterisk h323 on Asterisk 1.2.12 Fedora Core 5 - Asterisk h323 on Asterisk 1.4.Beta 2 Just looking for any pointers and decent directions. Hi, what softphone will you be using? Have you tried Ekiga! http://www.gnomemeeting.org/ I supports video also. thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] two SIP phones as one line
On Sun, Oct 15, 2006 at 11:24:00AM -0400, Marc Heckmann wrote: In fact no, I should have explained better, but in the old system one phone was analogue and the other was a multi-line digital Nortel Meridian phone. The one phone has to be analogue because it interfaces with a radio broadcast phone patch. You might want to re-evaluate your goal. Is your actual goal -- from an engineering standpoint -- to deliver the combined audio of both sides of the call to a broadcast-standard 600 ohm termination? Cause there may be better ways to do that than an analog phone and a hybrid... Cheers, -- jr 'drop back 10...' a -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ringtones won't work
I was hoping that someone may be able to shed some light on some issues I'm having on trying to get an Asterisk test server up and running. At the moment I have the basics, two Polycom hard phones (301 601 with expansion unit (which oddly will not power up)) that can call each other, log into voicemail (one touch) and have custom directories buddy lists. Unfortunately some of the seemingly simple things do not want to work for me: - Ringtones. Apparently the phones do not have any of the defaults on them as the Ring Type menu on each phone lists ms, Inc. beside each option, and will not play anything. I've placed several .wav files (from http://www.voipphreak.ca/index.php?serendipity%5Baction%5D=searchserendipity%5BsearchTerm%5D=ringtones ) and set up the sip.cfg as per what I've been able to find, a copy is below. The phones do download the .wav files on each boot, and list the filenames in the web browser config pages, but still show ms, Inc. under the Ring Type menu? - Busy indicators/presence. I have configured a buddy watcher on the 601 which will show the appropriate Online/On Phone status of the 301 through the Buddies menu, but it does not inicate the status from the directory key/button on the main screen. Should the indicator beside the contact name not show some sort of status update when the associated buddy is on the phone? - Voicemail. This one is just odd, and I have only found one search result that has the same issue but unfortunately no resolution. When either phone connects to voicemail they are presented with the voice prompts but any key I press is not recognized (ie. Press 1 for new messages and the voice prompts just continue like nothing was pressed). This happens through onetouch voicemail and by dialing the VM extension directly (I can't even log in if dialing the VM extension directly). If anyone can shed some light on these topics it would be greatly appreciated! Many thanks,MikeMY CURRENT SIP.CFG:--- ?xml version=1.0 standalone=yes?!-- SIP Application Configuration File --sip voIpProt local voIpProt.local.port=5060/ server voIpProt.server.1.address=10.215.100.1 voIpProt.server.1.port= voIpProt.server.1.transport=UDPonly voIpProt.server.1.expires=3600 voIpProt.server.1.register=1 voIpProt.server.1.retryTimeOut=0 voIpProt.server.1.retryMaxCount=0 voIpProt.server.1.expires.lineSeize=30/ SIP voIpProt.SIP.useRFC2543hold= 1 voIpProt.SIP.lcs=0 voIpProt.SIP.sendCompactHdrs=0 voIpProt.SIP.WM50=0 voIpProt.SIP.keepalive.sessionTimers=0 voIpProt.SIP.requestURI.E164.addGlobalPrefix= outboundProxy voIpProt.SIP.outboundProxy.address= voIpProt.SIP.outboundProxy.port=5060/ alertInfo voIpProt.SIP.alertInfo.1.value=AA voIpProt.SIP.alertInfo.1.class= 3/ alertInfo voIpProt.SIP.alertInfo.2.value=RA voIpProt.SIP.alertInfo.2.class=4/ requestValidation voIpProt.SIP.requestValidation.1.request= voIpProt.SIP.requestValidation.1.method= voIpProt.SIP.requestValidation.1.request.1.event= digest voIpProt.SIP.requestValidation.digest.realm= 10.215.100.1/ /requestValidation specialEvent voIpProt.SIP.specialEvent.lineSeize.nonStandard=1 voIpProt.SIP.specialEvent.checkSync.alwaysReboot=0/ conference voIpProt.SIP.conference.address=/ /SIP /voIpProt dialplan dialplan.impossibleMatchHandling=2 dialplan.removeEndOfDial=1 digitmap dialplan.digitmap=[2-9]11|0T|011xxx.T|[0-1][2-9]x|[2-9]x|[2-9]xxxT dialplan.digitmap.timeOut=3/ routing server dialplan.routing.server.1.address= dialplan.routing.server.1.port=5060/ emergency dialplan.routing.emergency.1.value=911 dialplan.routing.emergency.1.server.1=1/ /routing /dialplan sampled_audio saf.1=SoundPointIPWelcome.wav saf.2=RING_bennyhill.wav saf.3=RING_drwho.wav saf.4=RING_inspectorgadget.wav saf.5=RING_jamesbond.wav saf.6=RING_knightrider.wav saf.7=RING_macgyver.wav saf.8=RING_missionimpossible.wav saf.9=RING_nightcourt.wav saf.10=RING_ateam.wav/ HTTPD httpd.enabled= 1 httpd.cfg.enabled=1 httpd.cfg.port=80/ feature feature.1.name=presence feature.1.enabled=1/ logging level change log.level.change.sip=4 log.level.change.sip.obs=5/ /level /logging/sipEXCERPT EXTENSIONS.CONF: --- exten = 120,hint,SIP/120exten = 120,1,Macro(extensions,SIP/120,120)exten = 120,2,Dial(SIP/120)exten = 120,3,Answerexten = 120,4,Set(TIMEOUT(response)=10)exten = 120,5,Playback(NoAnswer_Extension) exten = 120,6,Voicemail(u120)exten = 120,n,Hangupexten = 158,hint,SIP/158exten = 158,1,Macro(extensions,SIP/158,158)exten = 158,2,Dial(SIP/158)exten = 158,3,Answerexten = 158,4,Set(TIMEOUT(response)=10) exten = 158,5,Playback(NoAnswer_Extension)exten = 158,6,Voicemail(u158)exten = 158,n,HangupEXCERPT SIP.CONF: --- [120]type=friendcontext=localusername=120password=12345host=dynamicdtmfmode=rfc2833[EMAIL
Re: [asterisk-users] How do you like TrixBox?
Dear All, I am have experimented asterisk long before any gui was available and also currently working with trixbox, ofcourse working with asterisk directly makes you more aware but when you start deploying the system you will face management issues for asterisk, as anyone who deals with asterisk must be experienced enough with it and that will make the people who support the users a few, while with trixbox those few people can be left as escalation points and through GUI you can make other less aware of asterisk administer the day to day tasks. Trixbox in my belief is making more people everyday depend on asterisk ofcourse knowing how to deal directly with asterisk will be a plus but yet this could come by time with trix box and everyday experience being gained will make them someday reach that level. Trixbox is a great start point to implement asterisk but learning asterisk configs must also be in schedule to maintain a persistent environment. Thx MAG Dovid B wrote: Yes but they will never understand the configs. They need to learn step by step. - Original Message - From: "joe, at j4computers" [EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.com> Sent: Friday, October 13, 2006 4:11 PM Subject: Re: [asterisk-users] How do you like TrixBox? Dovid B[EMAIL PROTECTED]> Wrote on: 10/13/2006 9:51 AM: >. . . A)If something goes wrong they wont know where to > start. They only know the GUI. B)They will never know the "real way" of > working asterisk.. . . > But, can't it be one way of "learning"? Can't one setup and modify a Trixbox setup, then peruse the conf files, to get familiar with (almost) all things Asterisk? Spoke as one who was not very pleased with their own foray into Trixbox and is still creeping up to speed on Asterisk. joe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thx MAG ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Codec swap (reinvite)
On 2006-10-14 20:00:30 -0700, Julian J. M. [EMAIL PROTECTED] said: Hi, I've finally given up on trying to fax over my Digium TDM400 card. I've found that fax over VoIP is quite more reliable (at least I can receive the faxes). My ITSP supports G729 and alaw/ulaw. As I won't be receiving faxes everyday (just ocasionally), i pretend on using g729, unless a fax is detected. Is there any way to force asterisk to make a reinvite, and swap the codec on the fly? Something like this would be great: exten = fax,1,CodecChange(ulaw) exten = fax,2,rxfax(blablabla) I think the answer is no. I am pretty sure this has been discussed multiple times and there is currently no way to change the codec once the call is established. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: SIP stuck channel soft hangup?
On 2006-10-14 13:15:55 -0700, Benny Amorsen [EMAIL PROTECTED] said: MJ == Martin Joseph [EMAIL PROTECTED] writes: MJ I added the rtptimeout=60 to my general section in sip.conf, and MJ now when the e60 goes out of wifi range, 61 seconds later, my MJ channels are clear! Sweet. Does this work with canreinvite=yes? (I can't see how it could, but I'd like to be surprised) Don't know, but that could be a problem. If the RTP stream is not going through the server I hope rtptimeout doesn't come into play? This isn't an issue for me, as the extension that is causing the issue is not allowed to do that anyhow... Good thought/question though. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID failover
Todd- Asterisk wrote: I'm setting up an asterisk server where an administrator will not always be available in case of problems. While I expect problems to be rare, I need to be prepared. We're thinking of VoIP DID's and SIP phones so it's an all TCP/IP network. We could get a second server to substitute - What is involved in 'transferring' or 're-registering' the DID incoming lines to a second server in case the primary is down? If there a better fall-over method? I'm looking for the easiest way for the un-educated sys-admin-apprentice to handle it. The system doesn't exist yet so any suggestions are appreciated. I recognize I'll need to modify the SIP phones- I'll figure that out later. thanks in advance One of the simplest ways to accomplish this is to use an APC power strip with SNMP control. (Each of the power outlets on the power strip can be turned on / off remotely via an snmp command. With this rough approach, stop the 'broken' asterisk server and start the backup server (via the power strip control), and wait for the system to come up. If both asterisk systems are configured absolutely the same (eg, same *.conf entries, ip addresses), then when the system comes up, it will 'register' with your sip or iax provider. The sip phones will likely take a little bit longer to come up due to arp cache timout values within the sip phones. I've not tested any of the sip phones to see what the default timeout values have to be, but it will vary by manufacturer. (Microsoft PC stuff is generally around two minutes.) As soon as that cache value timeouts out, the sip phone will register (with the new server) and should be totally functional. If at some future time you need a T1 or PRI on the system, someone manufacturers a T1 relay that will swap the T1 from one system to another. The downside to this approach is that you have to wait on each device's arp cache timeout value (including routers, dsl moems, sip phones, ATA boxes, and any other device that is required in you fully working system. Very few of the voip devices allow you to set the arp timeout value. In very general terms from a historical perspective, abruptly shutting down power to a linux/unix box is not is not an acceptable practice. However, the newer systems are far more tolerant, and for emergency purposes, its probably not that bad as the last step. If you read over some of the archives, there are other ways that involve redundant servers, heartbeats, load sharing, reserving a valid extension number that would kick of scripts (etc) to swap boxes. Each have their advantages, disadvantages, and costs. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SPA942 quality for a Bank
Before committing to about 50 of the spa942's, I like to take a last poll from those on the list to identify any negative issues that might be associated with the audio, functionality, early failures, etc, on the spa942. Expecting to deploy these using existing cat5 cabling and both rj45 jacks. Been using three of theme in a short term demo with the customer, but the demo systems has been purposefully configured with only basic telephony functions. Oh... someone mentioned the headset (no handset) pin jack is only for the microphone (and not the speaker) which would seem very odd. Anyone using a headset with the 942 where both the microphone and earpiece function fully? Any thoughts? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie question about meetme
On 22:48, Sat 14 Oct 06, Rajeev Natarajan wrote: yes to ztdummy: but you may have trouble when you try and run multiple simultaneous meetme sessions. On 10/5/06, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: omar parihuana wrote: Is possible use meetme feature without Zaptel card? (ztdummy will be the solution? ) Yup. :P I switched from app_meetme.so to app_conference.so couple of weeks ago and was stunned with the quality of app_conference.so No ztdummy needed so finally my OpenBSD boxen can run meetme and iax2 trunks :) If you dont have zaptel hardware and are experiencing trouble with multiple meetme running the same time (like me) try the app_conference.so I'm in no way involved in this project, it just made my life easier. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls being disconnected across VPN
Check the rpttimeout setting in sip.conf for the necessary extensions. As you are not making normal calls to outside parties, that setting may not applicable in your case. On 10/13/06, Jason Adams [EMAIL PROTECTED] wrote: Hey All, Sometimes we are running into issues with calls randomly being disconnected. We have a couple of phones is in a remote office. We have a PIX-to-PIX VPN setup between the offices. I know neither of these devices provide QOS but we usually don't run into any problems. I notice this problem when we are in a conference call with one of the employees at the remote office and a third party. They are disconnected after a minute. Here is the error: Chan_sip.c 11452 do_monitor: Disconnecting call 'SIP/103 for lack of RTP activity in 61 seconds. Any ideas? Jason Adams Sumo Systems 4694 Cemetery Road Suite 310 Hilliard, OH 43026 Phone | 614.433.9906 ext: 102 Fax | 614.433.9931 E-mail | [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA942 quality for a Bank
Rich Adamson wrote: Before committing to about 50 of the spa942's, I like to take a last poll from those on the list to identify any negative issues that might be associated with the audio, functionality, early failures, etc, on the spa942. Expecting to deploy these using existing cat5 cabling and both rj45 jacks. Been using three of theme in a short term demo with the customer, but the demo systems has been purposefully configured with only basic telephony functions. Oh... someone mentioned the headset (no handset) pin jack is only for the microphone (and not the speaker) which would seem very odd. Anyone using a headset with the 942 where both the microphone and earpiece function fully? Any thoughts? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users We use SPA 941's internally and are very happy with them.. Headset jack works great. Speaker premo.. Could use a backlight but other than that we are very happy.. You couldn't pry mine out of my cold dead hands.. -- VoIP Origination Termination Svcs. With Superior Customer Service! SIP, IAX, G711, GSM, G729 http://www.VoIPstreet.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID failover
Todd- Asterisk wrote: I'm setting up an asterisk server where an administrator will not always be available in case of problems. **SNIP** Not sure about your provider but within our control panel we offer our customers the ability to set multiple devices. Devices are basically setup as a one to one relationship to your Asterisk servers and or IP phones, soft phones etc. So you could setup 2 devices in our control panel or with your provider if they offer this. Device 1: Asterisk Server (Primary) Device 2: Asterisk Server (Secondary) Each device creates a SIP or IAX username and password and ability for each machine to register to us. When it comes to DIDs you can configure them in the control panel to all point to Device 1, in the event of a failure on device one you simply log in to our control panel and repoint your DIDs to Device 2. It takes approximately 5 minutes for our queue to run and all calls will now flow from your DIDs to Device 2. At any time you can push calls out through Device to and terminate to us regardless of when/if you repoint your DIDs. In the case of getting your internal phones to re-route to the correct system.. Maybe you could leave server 2 off, pre-configured with the IP of server 1. If you have a failure, simply turn off server 1 at IP X.X.X.X if it's not already dead and have someone hit the power switch to Server 2 which will now be at the same X.X.X.X IP. No IP phone re-configuration required. I am sure there is a more elegant solution using a router or something but, this is just one approach. So, there are a few ways to skin that cat, just up to you how you want to go about it. Hope this helps.. -- VoIP Origination Termination Svcs. With Superior Customer Service! SIP, IAX, G711, GSM, G729 http://www.VoIPstreet.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA942 quality for a Bank
At 02:30 PM 10/15/2006, you wrote: Before committing to about 50 of the spa942's, I like to take a last poll from those on the list to identify any negative issues that might be associated with the audio, functionality, early failures, etc, on the spa942. We have been using Cisco hard phones with Asterisk for over two years. Our longest experience has been with 7960g phones although recently I switched to the 7970g for evaluation. Several of us have tried the SPA942's for about one month. Personally I was real happy to go back to a Cisco IP phone. The sound quality is useable but just not as good on the SPA942. And the speaker phone on the SPA942 is poor enough quality that it is barely usable. We include the SPA-942 in our side by side demos for prospective customers. So far they are buying Cisco and willing to pay the higher price. We only have four of the SPA-942's and have not seen any failures in our limited use. Expecting to deploy these using existing cat5 cabling and both rj45 jacks. Been using three of theme in a short term demo with the customer, but the demo systems has been purposefully configured with only basic telephony functions. Oh... someone mentioned the headset (no handset) pin jack is only for the microphone (and not the speaker) which would seem very odd. Anyone using a headset with the 942 where both the microphone and earpiece function fully? We have used them with VXI headsets and the microphone works fine. Any thoughts? _ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] detecting the receivers voicemail
Hi, Is there any way asterisk can detect if the outgoing call is being received by a user or it has been forwarded to his voicemail. Actuallymy requirement is to proceed only if user picks up the phone otherwiseto hangup as soon as the call goes to voicemail. Is there anyway to do this? Thanks in advance. Nitin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA942 quality for a Bank
Tom wrote: At 02:30 PM 10/15/2006, you wrote: Before committing to about 50 of the spa942's, I like to take a last poll from those on the list to identify any negative issues that might be associated with the audio, functionality, early failures, etc, on the spa942. We have been using Cisco hard phones with Asterisk for over two years. Our longest experience has been with 7960g phones although recently I switched to the 7970g for evaluation. Several of us have tried the SPA942's for about one month. Personally I was real happy to go back to a Cisco IP phone. The sound quality is useable but just not as good on the SPA942. And the speaker phone on the SPA942 is poor enough quality that it is barely usable. Yes, we see about the same here comparing the old BT102, Cisco 79x), and the spa942. Given most of the demo's thus far are with small banks, price weights very heavily in their mind. We include the SPA-942 in our side by side demos for prospective customers. So far they are buying Cisco and willing to pay the higher price. The sales pitch tries to address the sip licensing costs on the Cisco 79x0's, and when that's added to the base refurb cost, the banks seem to move quickly to the 942's. We only have four of the SPA-942's and have not seen any failures in our limited use. Expecting to deploy these using existing cat5 cabling and both rj45 jacks. Been using three of theme in a short term demo with the customer, but the demo systems has been purposefully configured with only basic telephony functions. Oh... someone mentioned the headset (no handset) pin jack is only for the microphone (and not the speaker) which would seem very odd. Anyone using a headset with the 942 where both the microphone and earpiece function fully? We have used them with VXI headsets and the microphone works fine. Good. Who mentioned the 'mic only' must not have had the correct headset/plug for the spa942. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] detecting the receivers voicemail
Not really. Not unless you prompt the user to press a key and they have to do within x amount of seconds. - Original Message - From: Nitin Gupta To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, October 16, 2006 12:39 AM Subject: [asterisk-users] detecting the receivers voicemail Hi, Is there any way asterisk can detect if the outgoing call is being received by a user or it has been forwarded to his voicemail. Actuallymy requirement is to proceed only if user picks up the phone otherwiseto hangup as soon as the call goes to voicemail. Is there anyway to do this? Thanks in advance. Nitin ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID failover
On Sun, Oct 15, 2006 at 02:06:51PM -0500, Rich Adamson wrote: Todd- Asterisk wrote: I'm setting up an asterisk server where an administrator will not always be available in case of problems. While I expect problems to be rare, I need to be prepared. We're thinking of VoIP DID's and SIP phones so it's an all TCP/IP network. We could get a second server to substitute - What is involved in 'transferring' or 're-registering' the DID incoming lines to a second server in case the primary is down? If there a better fall-over method? I'm looking for the easiest way for the un-educated sys-admin-apprentice to handle it. The system doesn't exist yet so any suggestions are appreciated. I recognize I'll need to modify the SIP phones- I'll figure that out later. thanks in advance One of the simplest ways to accomplish this is to use an APC power strip with SNMP control. (Each of the power outlets on the power strip can be turned on / off remotely via an snmp command. With this rough approach, stop the 'broken' asterisk server and start the backup server (via the power strip control), and wait for the system to come up. If both asterisk systems are configured absolutely the same (eg, same *.conf entries, ip addresses), then when the system comes up, it will 'register' with your sip or iax provider. The sip phones will likely take a little bit longer to come up due to arp cache timout values within the sip phones. I've not tested any of the sip phones to see what the default timeout values have to be, but it will vary by manufacturer. (Microsoft PC stuff is generally around two minutes.) As soon as that cache value timeouts out, the sip phone will register (with the new server) and should be totally functional. [ ... ] If you read over some of the archives, there are other ways that involve redundant servers, heartbeats, load sharing, reserving a valid extension number that would kick of scripts (etc) to swap boxes. Each have their advantages, disadvantages, and costs. ...and what Rick describes is the little baby version of failover. Setting up the High-availaibility heartbeat daemons is *reportedly* not that bad, and will save you the ARP delay by the simple expedient of having the backup machine take over the primary's MAC address as well as IP address. I myself will be needing shortly to look into whether it's possible to carry other active-session state from one machine to the other (I'm hoping for transaction-mirrored PgSQL databases on the two machines) for a project I'm considering. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Increase VoiceMail Messages Recording Gain - AudioCalls are Ok
The problem is:Right now, and i'm referring only to calls directly handled by VoiceMail application, the users get their audio files in email but the audio is very very low. I've thought about changing RX gain on PRI interface between legacy pbx and asterisk, but until now no complaining with audio calls. There's a patch for this: http://bugs.digium.com/file_download.php?file_id=10824type=bug Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] two SIP phones as one line
On Sun, 2006-15-10 at 13:42 -0400, Jay R. Ashworth wrote: On Sun, Oct 15, 2006 at 11:24:00AM -0400, Marc Heckmann wrote: In fact no, I should have explained better, but in the old system one phone was analogue and the other was a multi-line digital Nortel Meridian phone. The one phone has to be analogue because it interfaces with a radio broadcast phone patch. You might want to re-evaluate your goal. Is your actual goal -- from an engineering standpoint -- to deliver the combined audio of both sides of the call to a broadcast-standard 600 ohm termination? yes, using a Telos One digital hybrid. Only catch is that the On-Air techs like to be able to 'stack' calls and place callers on hold using a multi-line phone (the former Meridian replaced by the new SIP phone) and then choose which ones to put on air. Of course, a proper solution would be to get one of these: http://www.telos-systems.com/1x6/default.htm but I suspect that I can avoid it with the right Asterisk config. -m Cause there may be better ways to do that than an analog phone and a hybrid... I'm open to other ideas... Cheers, -m signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do you like TrixBox?
I agree with Mohamed. TrixBox is an excellent way to start, but in the long run, if you attempt to use Asterisk in a business setting, you will probably want to be able to hardcode the conf files yourself. I have only recently changed over to TrixBox from a standard installation on a debian system. Honestly, I really don't use FreePBX much at all. I use it to create new extentions for VMB's since I need to create a number of them every day for users on my website, but overall I code my own dialplan. I don't really understand FreePBX well enough, nor do I really want to put in the effort of learning it since I can already hand-code. TrixBox just has a couple of nifty features that I enjoy to make daily life a tad easier. On 10/15/06, Mohamed A. Gombolaty [EMAIL PROTECTED] wrote: Dear All, I am have experimented asterisk long before any gui was available and also currently working with trixbox, ofcourse working with asterisk directly makes you more aware but when you start deploying the system you will face management issues for asterisk, as anyone who deals with asterisk must be experienced enough with it and that will make the people who support the users a few, while with trixbox those few people can be left as escalation points and through GUI you can make other less aware of asterisk administer the day to day tasks. Trixbox in my belief is making more people everyday depend on asterisk ofcourse knowing how to deal directly with asterisk will be a plus but yet this could come by time with trix box and everyday experience being gained will make them someday reach that level. Trixbox is a great start point to implement asterisk but learning asterisk configs must also be in schedule to maintain a persistent environment. Thx MAG Dovid B wrote: Yes but they will never understand the configs. They need to learn step by step. - Original Message - From: joe, at j4computers [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, October 13, 2006 4:11 PM Subject: Re: [asterisk-users] How do you like TrixBox? Dovid B[EMAIL PROTECTED] Wrote on: 10/13/2006 9:51 AM: . . . A)If something goes wrong they wont know where to start. They only know the GUI. B)They will never know the real way of working asterisk.. . . But, can't it be one way of learning? Can't one setup and modify a Trixbox setup, then peruse the conf files, to get familiar with (almost) all things Asterisk? Spoke as one who was not very pleased with their own foray into Trixbox and is still creeping up to speed on Asterisk. joe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --ThxMAG ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- www.AsteriskBlog.comYour home for easy to learn Asterisk stuff. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] two SIP phones as one line
On Sun, Oct 15, 2006 at 10:52:27PM -0400, Marc Heckmann wrote: On Sun, 2006-15-10 at 13:42 -0400, Jay R. Ashworth wrote: On Sun, Oct 15, 2006 at 11:24:00AM -0400, Marc Heckmann wrote: In fact no, I should have explained better, but in the old system one phone was analogue and the other was a multi-line digital Nortel Meridian phone. The one phone has to be analogue because it interfaces with a radio broadcast phone patch. You might want to re-evaluate your goal. Is your actual goal -- from an engineering standpoint -- to deliver the combined audio of both sides of the call to a broadcast-standard 600 ohm termination? yes, using a Telos One digital hybrid. Only catch is that the On-Air techs like to be able to 'stack' calls and place callers on hold using a multi-line phone (the former Meridian replaced by the new SIP phone) and then choose which ones to put on air. Of course, a proper solution would be to get one of these: http://www.telos-systems.com/1x6/default.htm but I suspect that I can avoid it with the right Asterisk config. I'm open to other ideas... Seems to *me* that what you need to do is able to dynamically pick the call they want to deal with, and hand it to a meetme where the other side's audio gets merged and fed to a 600ohm bridge, which goes directly into your board. IE: replace the hybrid. You might pop over to the Rivendell mailing list; some of those people might have something informative to say as well... I don't know if Call Commander is Asterisk compatible (yet) or not. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] eagi-sphinx-test how and why
Hi I see eagi-sphinx-test in agi-bin, anyone know how is it supposed to be used and what version of sphinx. Any help will be appraciated mawali ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip agent stuck in queue even after restarts
This is the first time I have used a sip device on a queue. Iogged in under the extension and now I can't logout. No kidding. I have restarted asterisk with persistant agents=off and also done a show agents. It shows no agents logged on and I am still receiving calls. To complicate things I am using Flash operator panel to see logged in agents and it has, at best, been sporadic. I have had no problems for the last six months and now I am in a hole. I beleive that the agent is stuck in the astdb somehow but every attempt I have made to remove it fails. Can any one see what I might be missing? winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Reception Console
We are currently writing a reception console for Asterisk - if anyone is interested in beta testing it, feel free to ask. Paul Hales -- Paul Hales Technical Manager AsteriskIT www.asteriskit.com.au bus: 03 8320 8106 mob: 0434 673 529 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reception Console
On 16/10/2006, at 2:32 PM, Paul Hales wrote: We are currently writing a reception console for Asterisk - if anyone is interested in beta testing it, feel free to ask. If it can handle multiple Asterisk servers -- ME, ME! PICK ME! PICK ME! :) Thanks, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.net .. / ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reception Console
Count me in. Paul Hales wrote: We are currently writing a reception console for Asterisk - if anyone is interested in beta testing it, feel free to ask. Paul Hales ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoicePulse Connect 4 Channel Limit?
Does anyone know what happens if you try to have 5 concurrent outgoing channels with VoicePulse Connect? Does it give you an error message or a reorder or something? I'm worried about using them as my primary carrier if this is the case. I noticed that they supposedly only allow 4 channels for free and then you have to pay $20 a month extra per channel. I'm guessing this is for inbound and outbound channels. If you wanted to be able to have 8 concurrent channels then this could get costly. Too costly in my opinion. I meanthat seems like a LOT to me, when you can go with other providers who don't limit you to 4 channels, like Voxee, NuFone or SixTel, for around the same price. I can understand the channel restrictions for inbound calls, but not for outbound calls. VoicePulse, I know you read these lists! You should be able to provide us VoicePulse Connect users with more than 4 concurrent channels for free! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reception Console
Hi Yes I am interested. Regards Jon --- Paul Hales [EMAIL PROTECTED] wrote: We are currently writing a reception console for Asterisk - if anyone is interested in beta testing it, feel free to ask. Paul Hales -- Paul Hales Technical Manager AsteriskIT www.asteriskit.com.au bus: 03 8320 8106 mob: 0434 673 529 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Farmer Telford, Shropshire, UK ___ All new Yahoo! Mail The new Interface is stunning in its simplicity and ease of use. - PC Magazine http://uk.docs.yahoo.com/nowyoucan.html ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] List of VoIP+RJ11 Phones
Hi friends,Thank you to all for response. At last, I got these below links which contains Ethernet port and RJ11 port. http://www.voipsupply.com/product_info.php?products_id=307 http://www.thechewtongroup.com/zultys-zip-4x5.phphttp://gigaset.siemens.com/shc/0,1935,hq_en_0_122378_rArNrNrNrN,00.html Thank you.Regards,Chandra. Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reception Console
Sure thing, count me in Paul Hales wrote: We are currently writing a reception console for Asterisk - if anyone is interested in beta testing it, feel free to ask. Paul Hales ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Reception Console
I'm interesting in testing this. Scott Higginbotham Systems / Network Operations Manager 215.259.2185 or 1.800.835.5710 ext 2185 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Paul Hales Sent: Monday, October 16, 2006 12:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Reception Console We are currently writing a reception console for Asterisk - if anyone is interested in beta testing it, feel free to ask. Paul Hales -- Paul Hales Technical Manager AsteriskIT www.asteriskit.com.au bus: 03 8320 8106 mob: 0434 673 529 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] eagi-sphinx-test how and why
http://turnkey-solution.com/asterisk-sphinx.html~VamsiOn 10/16/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: HiI see eagi-sphinx-test in agi-bin, anyone know how is it supposed to beused and what version of sphinx.Any help will be appraciatedmawali___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_bluetooth - one way audio
I have compiled chan_bluetooth and I have used it many times before... but now I have a problem and I can´t find a solution. I have a bluetooth dongle (MSI) connected to my asterisk and I use a V3 phone to make calls to another cell phones... the thing is that I just get one way audio.. I can hear people from my IAX softphone but they can´t hear me... I don´t know what could be.. anyone had the same problem? Thanks for your time, Danko This is my bluetooth.conf [general] ; Channel we listen on as a HS (Headset) rfchannel_ag = 7 ; hci interface to use (number - e.g '0') interface = 0 ;; Some phone example [00:14:9A:F8:EB:E7] name= V3; The name of the phone channel = 3 ; The channel used by the HS audio gateway profi le autoconnect = yes ; Tells chan_bluetooth to establish link at startup type= AG ~# sdptool search --bdaddr 00:14:9A:F8:EB:E7 0x111F Class 0x111F Searching for 0x111F on 00:14:9A:F8:EB:E7 ... Service Name: Hands-Free voice gateway Service Description: Hands-Free voice gateway Service Provider: /a/mobile/system/cl.gif Service RecHandle: 0x10007 Service Class ID List: Handfree Audio Gateway (0x111f) Generic Audio (0x1203) Protocol Descriptor List: L2CAP (0x0100) RFCOMM (0x0003) Channel: 7 Language Base Attr List: code_ISO639: 0x656e encoding:0x6a base_offset: 0x100 code_ISO639: 0x6672 encoding:0x6a base_offset: 0xd800 code_ISO639: 0x6573 encoding:0x6a base_offset: 0xd803 code_ISO639: 0x7074 encoding:0x6a base_offset: 0xd806 Profile Descriptor List: Handsfree (0x111e) Version: 0x0101 It didn´t worked with channel 7.. I can´t make them pair.. Asterisk initialization: NOTICE[3263]: /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.c:2227 try_connect: Initialised bluetooth link to device V3 [AG] V3 AT+BRSF=23 [AG] V3 +MBAN: Copyright 2000-2002 Motorola, Inc. WARNING[3272]: /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.c:622 sco_thread: SCO thread started on fd 21, pid 3249 WARNING[3263]: /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.c:2585 handle_rd_data: Device V3: Unhandled Unsolicited: +BRSF: 63 [AG] V3 +BRSF: 63 [AG] V3 OK [AG] V3 AT+CIND=? [AG] V3 +CIND: (Voice Mail,(0,1)),(service,(0,1)),(call,(0,1)),(Roam,(0-2)),(signal,(0-5)),(callsetup,(0-3)),(smsfull,(0,1)) [AG] V3 OK [AG] V3 AT+CIND? NOTICE[3263]: /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.c:510 set_cind: Audio Gateway V3 got signal [AG] V3 +CIND: 0,1,0,0,5,0,0 [AG] V3 OK [AG] V3 AT+CMER=3,0,0,1 [AG] V3 OK [AG] V3 AT+CLIP=1 [AG] V3 OK [AG] V3 AT+CGMI [AG] V3 +CGMI: Motorola CE, Copyright 2004 [AG] V3 OK [AG] V3 AT+CGMI [AG] V3 +CGMI: Motorola CE, Copyright 2004 [AG] V3 OK ERROR[3272]: /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.c:702 sco_thread: SCO connection error: Connection reset by peer (errno 104) Making a call -- Accepting AUTHENTICATED call from 192.168.0.102: requested format = gsm, requested prefs = (), actual format = gsm, host prefs = (), priority = mine -- Executing Dial(IAX2/danko-2, BLT/V3/1150246209) in new stack [AG] V3 ATD1150246209; -- Called V3 [AG] V3 OK [AG] V3 +CIEV: 6,2 [AG] V3 +CIEV: 5,4 WARNING[3290]: /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.c:622 sco_thread: SCO thread started on fd 26, pid 3249 [AG] V3 +CIEV: 6,3 -- BLT/V3 is ringing [AG] V3 +CIEV: 5,4 [AG] V3 +CIEV: 5,5 [AG] V3 +CIEV: 5,4 [AG] V3 +CIEV: 5,5 [AG] V3 +CIEV: 5,4 [AG] V3 +CIEV: 5,5 [AG] V3 D: VOICE [AG] V3 +CIEV: 6,0 [AG] V3 +CIEV: 5,4 WARNING[3290]: /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.c:686 sco_thread: Wrote 48 to sco WARNING[3290]: /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.c:686 sco_thread: Wrote 48 to sco WARNING[3290]: /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.c:686 sco_thread: Wrote 48 to sco and lots and lots of the same... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
