[Asterisk-Users] PHP Manager
Hi all, I have a small problem to execute Asterisk Commands in Asterisk Manager using PHP. I am able to run all Asterisk Manager command but the problem is comming with asterisk command. here is the code i am trying to run. ?php $socket = fsockopen(localhost,5038, $errno, $errstr, $timeout); fputs($socket, Action: Login\r\n); fputs($socket, UserName: 1212\r\n); fputs($socket, Secret: 1212\r\n\r\n); fputs($socket, Action: Command\r\n); fputs($socket, Command: reload\r\n\r\n); #Working well fputs($socket, Command: show channels\r\n\r\n); #Not working Working well fputs($socket, Command: 'show channels'\r\n\r\n); #Not working Working well $wrets=fgets($socket,128); ? If you see in my code when i am calling only reload command working but when i am trying to call piar command it is just prompting : == Manager '1212' logged off from localhost without showing channels Please advice me to solve this problem. -- Thank You, Code Lover ___ --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] Maximizing audio quality
Wolfgang Borgon wrote: A RAW file I created after converting from MP3 and WAV, sounded raspy. Does anyone have any tips for creating the best quality voice recordings? Generally you'd use a good-quality microphone for your recordings. The adage Garbage in = garbage out couldn't be more true in this instance. If you're looking for studio-quality recordings, use studio-quality equipment. Those $5 mics won't be satisfactory :) Then there's issues of sibilance, which isn't that apparent when you're recording at a higher rate, but is really pronounced when you downsample to 8k for the GSM files. The raspiness you encountered was probably sibilance, where words that have the ess sound in them are boosted due to the position of the microphone relative to the person being recorded. If you're going the budget route, at least get a decent quality sound card to record with. Another important factor to consider is your recording location -- try and record in as quiet a place as you can find. Some audio processing software (Goldwave, Audacity et al) have filters that can knock out background noise, alter volume, apply equalization etc. You can use these effects to enhance the recording. But again, if your original recording already sounds bad there's not much you can do to make it sound nice. Cheers, Flynn ___ --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] Asterisk Hosting
We've been trying Unison (http://www.cis.upenn.edu/~bcpierce/unison/) on a 1 minute cron job. There are some theoretical issues but it has been great so far. We use it to synch prompts as well as messages. SimonOn 12/27/05, BILL GITONGA [EMAIL PROTECTED] wrote: What is the best method of storing voice main messagesso that they are accessible to different asteriskservers in a hosted environment? I have consideredAsterisk real time but I don't think it stores theactual voice mail folder in the database. I'm thinking of using NFS for this and put my voice mail folders onthe NFS so that it is accessible by the differentservers. Is this a good way to do it or is there abetter way of doing this?__ Yahoo! for Good - Make a difference this year.http://brand.yahoo.com/cybergivingweek2005/___--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] Bad Checksum answering inbound call
Could anybody please help me with problem.. Outbound calls work fine, however inbound calls ring the phone, then answering the call, the service provider doesnt receive the picked up message from asterisk. We have narrowed it down to an incorrect checksum in the packets being sent back from asterisk after answering an inbound call. Regards, Darren Younger National Solutions Architect Nightfire Technologies Pty Ltd [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
[Asterisk-Users] billing system
Hello All, Have anybody test ISP BILLING SYSTEM ? http://ibs.sourceforge.net/index.html Regards Harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.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
[Asterisk-Users] Re: Voicemail as other format?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Some users with Blackberry's cant play .wav files, is there a way to save the voicemail to save as another format like mp3? -Kerry In voicemail.conf edit this line. [general] format=wav49|gsm|wav P.S. Please stop replaying to mesage. If you plan to open thread do so by writing mail to this address asterisk-users@lists.digium.com -- Tomislav Parcina [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
[Asterisk-Users] Re: Re: Grandstream Budge Tone 102
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Did you set the timezone correctly? All my phones are 1 hour behind because I still have to login to all phones to alter daylight saving time setting. It isn't just time, he misses the year! :))) Acording to Budgetone now is 1900. ;)) Configuration is quite strait-forward, I just need to enter IP address of NTP server (10.0.0.20), right? Phone IP address is 10.0.0.137, they are on same subnet. Cisco phones work with this NTP server!?! I realy don't know what could be the problem. -- Tomislav Parcina [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
[Asterisk-Users] Re: Does broadvoice modify caller ID name?
Please stop replaying to mesage. If you plan to open thread do so by writing mail to this address asterisk-users@lists.digium.com -- Tomislav Parcina [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
[Asterisk-Users] Re: Transfer
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I am aware of the possibility to add the option t or T to dial, so #33 transfers the call to extension 33. It needs to be deined in feautres.conf file. So when you dial #1 you'll hear transfer and than you enter extension. Is there any use of this command in the dialplan? If I want to redirekt a call because of the choices of a caller goto() or dial() does the job. In dialplan you need only to enter t and/or T. -- Tomislav Parcina [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
Re: [Asterisk-Users] Re: 4-port external sip fxo which doesnt suck?
I'd agree with you on the quality of the audio. But, unless they've changed a lot in the last year, the 1204's sip support and mgmt leaves a lot to be desired. For a box that has very poor reviews, it sure is great to use a box that you can throw in the closet and just forget about it. They just always work and sound great. The first time you configure one is a bit of a pain, but after that it is cruz time. I use a linux mib browser (mbrowse) because I work in an usoft free environment. I can drop ship a unit and have them plug it into the pbx lan and then configure it remotely. I find snmp more convenient than a browser interface. I have deployed quite a few Mediatrix 1204 and have never gone back and looked at any of them again. They just work. I'm looking for a 4-port external sip fxo which doesn't suck. o) Clipcomm CG-410. Poor reviews. o) Mediatrix 1204. Very poor reviews. o) Audiocodes MP104. Poor reviews. o) DLink DVG-3004S. Doesnt seem to exist yet. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ --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 ---End of Original Message- ___ --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] Automatic logoff of all agents at set time
On 18:06, Tue 27 Dec 05, Bud Bach wrote: But, if the agents don't log out for some reason, they will still be logged in the next time the queue opens even if they aren't there right? yes. What you can do is 2 things: * you can set the autologoff time in agents.conf. This can give you some trouble when agents go to the toilet or grab a cup of coffee. * set persistant agents to off and restart asterisk at midnight. This will logoff the agents :) Hope this helps -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D 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
[Asterisk-Users]CLI execute extensions
What I can doing execute extension (es.200) from CLI? Thanks Yahoo! Mail: gratis 1GB per i messaggi, antispam, antivirus, POP3___ --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] Stay away from Grandstream!
Brian Capouch wrote: They don't perform as well as the expensive Ciscos and Polycoms, but many of us are using them in a variety of circumstances quite happily. I have 4 of them in a small office (GXP2000) running 1.0.12 and they're just fine for our purposes. As Brian said, YMMV. For our 60-person office in Sydney, I'm probably going to use a mix of Polycom/Grandstream and softphones. cYa, Avi -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 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
[Asterisk-Users] Wrong Password?????
I've been working with asterisk with public ip but since I change this and put asterisk behind NAT, get this error when my hard phones try to register Dec 28 11:43:33 NOTICE[8716]: chan_sip.c:10817 handle_request_register: Registration from 'sip:[EMAIL PROTECTED]:5060' failed for '...' - Wrong password [general] section of sip.conf file looks like this: [general] bindport=5060 bindaddr=0.0.0.0 dtmfmode=rfc2833 context=incoming ; Default for incoming calls externip=... localnet=10.0.2.0/255.255.255.0 Regards, Rafael Ledesma Serrano Administrador de Sistemas Palmanet Networking Services [EMAIL PROTECTED] http://www.palmanet.net Tel +34 957649199 Fax +34 957644926 ___ --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] asterisk AVM C2 again
On Tue, 2005-12-27 at 22:39 +0100, Armin Schindler wrote: On Tue, 27 Dec 2005, Dave Cotton wrote: On Tue, 2005-12-27 at 19:27 +0100, Armin Schindler wrote: It looks like the call is signaled on both ports !? On another installation in France I'm also getting this, but with 2 Fritz! cards, the call is signalled on both cards. Is this some feature of the line configuration/protocol? I never heard of this before. I really don't know. It only happens at one location, another with exactly the same setup runs normally. Unfortunately it's virtually impossible to find anyone to discuss this with at France Telecom. -- Dave Cotton [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
[Asterisk-Users] Re: Why no sound from festival?
Please stop replaying to mesage. If you plan to open thread do so by writing mail to this address asterisk-users@lists.digium.com -- Tomislav Parcina [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
Re: [Asterisk-Users] asterisk AVM C2 again
Dave Cotton a écrit : On Tue, 2005-12-27 at 22:39 +0100, Armin Schindler wrote: On Tue, 27 Dec 2005, Dave Cotton wrote: On Tue, 2005-12-27 at 19:27 +0100, Armin Schindler wrote: It looks like the call is signaled on both ports !? On another installation in France I'm also getting this, but with 2 Fritz! cards, the call is signalled on both cards. Is this some feature of the line configuration/protocol? I never heard of this before. I really don't know. It only happens at one location, another with exactly the same setup runs normally. Unfortunately it's virtually impossible to find anyone to discuss this with at France Telecom. france telecom send call on both T0, it's the reson why * can't andle more than 2 chan. i have some access in france telecom, i try to have more informations. happy new year -- Stephane Plichon | HASGARD tel: +33 (0)472529881 fax: +33 (0)472177764 web: http://www.hasgard.net email: [EMAIL PROTECTED] jabber: [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
[Asterisk-Users] Re: Asterisk Christmas Help request
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... 4) Is it possible to make a routing, as follows Dial 8 go to Internet Call Dial 9 go to TelCo. Call Read this: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf 5) How do I change the time zone for Asterisk? Currently the system time is correct but when I dial *60 it reports a different time (out by many hours). I'm not familiar with this option. Can you please tell me more or send me some link. -- Tomislav Parcina [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
[Asterisk-Users] Re: channel monitoring whisper mode?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I am thinking to develop one. Thank you! (in advance :)) -- Tomislav Parcina [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
Re: [Asterisk-Users] ISDN/CAPI outgoing calls - weirdness with ringing
Armin, Please find attached the log file for 4 x test calls: 1. Call to Orange GSM mobile phone (effectively ISDN all the way) - works as expected 2. Call to BT analogue phone (we still have a lot of analogue in the UK) - no ringing problem (stuck in Session Progress) 3. Call to CW analogue phone (local loop unbundling operator) - double ringing problem 4. Call to SIPgate/Magrethea number (complex call routing) - works as expected (I am slightly suprised :o) Set up is: - Fedora Core 4, on P4 3.2GHz, 1Gb RAM - AVM C4 with: - 2 x ISDN2e in P2P mode on 01905756700/01905755777/0190475289x - 1 x ISDN2e in P2MP for other bits the main number is 01905756700 - Asterisk 1.2.1 - Chan-capi-cm-0.6.1 all cleanly compiled and re-installed for testing. I use a macro for placing the outgoing calls (copy at the start of the log) which aides with debugging and ensures the correct '6700' number (main PBX number) is used. All tests dialled with the macro provided and using just the 'b' flag. Look forward to your comments with interest. Regards Mike - Original Message - From: Armin Schindler [EMAIL PROTECTED] To: Michael J. Tubby G8TIC [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, December 27, 2005 6:37 PM Subject: Re: [Asterisk-Users] ISDN/CAPI outgoing calls - weirdness with ringing On Sun, 25 Dec 2005, Michael J. Tubby G8TIC wrote: On Sat, 24 Dec 2005, Michael J. Tubby G8TIC wrote: I changed the dial-string to include flags 'ob' as you mentioned (below) and now I get the following when I dial a BT phone number - dial number, get: Proceeding (in 100) briefly - after a second or so: Ringng Destination (in 180) - double ringing tone: BT style ringing generated by the exhange Cisco phone US-style ringing (generated by the phone) these are overlaid on each other (mixed together) My hunch is that there's something not right with the call set up sequence and CAPI handling. This is not a problem of CAPI. When you specify 'b' for early-b3, you will get the tones from the switch. If your phone adds its own tone, even when it receives progress tones, then it is incorrect (maybe wrong setup). Armin However the difference that I see looking at the Cisco 7960 phone which shows a version of the SIP messages on its status line is: 100 Proceeding 183 Session Progress 180 Ringng Destination the order of which varies and depends on the dialled number. Some dialled numbers go: 100-183-180 and these produce one set of alerting/ringing correctly. Some dialled numbers go: 100-183 and stay in state 183 until the called party answers - these are the ones that produce no ringing. Can you provide a verbose log level 5 with 'capi debug' ? I would like to compare the capi messages. Maybe the switch just send an alerting message. If I add the 'o' to the existing 'b' flag then dial it appears to change the behaviour so that the phone goes 100-180 for all calls but some give me a single (phone generated US style ring) while others give the 'double ringing'. The ones that produce double ringing are the ones that would have rung before, while the ones that now produce ringing (from the exchange) are the ones that used to be silent. When using 'o', chan_capi is doing early-b3 from the beginning before sending any digits and you will get b3-data in each case. Please send me a debug log of a connection with double ring-tone (no 183) as well. Armin isdn_testing.log Description: Binary data ___ --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] agent logs
Have a look at /var/log/asterisk/queue_log It has to be enabled on logger.conf (queue_log=yes on the [genera] section). - Original Message - From: Hall, Eric M. To: asterisk-users@lists.digium.com Sent: Tuesday, December 27, 2005 6:25 PM Subject: [Asterisk-Users] agent logs I'm looking for a ay to track when an agent logs inand logs out. Best if it could be put in a mysql db but a text file will be ok for now.. Any help would be great ! Thanks ___--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] spandsp fax
I am using Red Hat 9, but I don't think this changes the procedure - Original Message - From: Carlos Alperin [EMAIL PROTECTED] To: 'Dov Bigio' [EMAIL PROTECTED]; 'Asterisk Users Mailing List -Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, December 27, 2005 8:24 PM Subject: RE: [Asterisk-Users] spandsp fax Don, The previous question I believe was what linux are you using? By the way, I would like to know that too, just I was trying to make this work for weeks with no success. Thanks, Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dov Bigio Sent: Tuesday, December 27, 2005 10:54 AM To: Kristof Hardy; Asterisk Users Mailing List - Non-CommercialDiscussion Subject: Re: [Asterisk-Users] spandsp fax Hi BJ, Kristof, It worked! I am using the version at http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.2pre21c/asterisk-1. 2.x/. I think I had bad symlinks on /usr/local/lib and by reading the tutorial on AsteriskGuru I found that... (The previously installed version of spandsp has been 0.0.3, but now you have installed version 0.0.2. The problem is that the installation of version 0.0.3 creates a symlink, which is not replaced by installation of version 0.0.2. So the symlink points to the library of version 0.0.3, which actually does not exist.). I simply deleted all files related to spandsp from this directory and installed it again! Thank you Dov - Original Message - From: Kristof Hardy [EMAIL PROTECTED] To: Dov Bigio [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-CommercialDiscussion asterisk-users@lists.digium.com Sent: Tuesday, December 27, 2005 12:59 PM Subject: Re: [Asterisk-Users] spandsp fax Dov Bigio wrote: I am using Asterisk 1.2.1 and followed instructions on http://www.asteriskguru.com/tutorials/spandsp.html to install faxing capability on my server. what platform are you running on? (wich distro?) Does the make of the app_txfax and app_rxfax work out well? ___ --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
[Asterisk-Users] UK, Disconnect supervision
Jonathan, many thanks for your reply. The adapter has firmware version 3.1.3(GWa). Does that version have problems with diconnect tones? Would you recommend I should upgrade? Can you give me some reasons or point me to resources (apart from google) where I can research further? What would you say are the risks of upgrading? I am usually a bit anxious about firmware upgrade because I have that fixed idea that EITHER the new firmware may break other features (like - registration problems with SIP provider, connectivity issues and so on) OR there may be some problem during firmware upgrade which damages the device in question. For example, for the Grandstream Budgetone 100 phone, power outage during firmware upgrade from TFTP will damage the device(1). And I can't fix it once it's broken; it's not like a computer where I simply reinstall the OS / put in a new component etc. Once it's gone, it's gone. My fears are probably totally unfounded, but better safe than sorry. So I wouldn't upgrade unless there are good reasons to do so (if it ain't broke, don't fix it). But thanks very much for that hint. I actually have two other adapters, and they may be way out of date: 2.0.13(GWg) - so they may really need updating. Peter -- (1)BudgeTone-100 User Manual, version 1.0.5.11, section 6.1: Upgrade with TFTP, warning: The device WILL get damaged if there is a power outage during firmware upgrade. Grandstream STRONGLY recommend customer maintain UNINTERRUPTED POWER SUPPLY during firmware upgrade. This damage is NOT covered by the manufacture warranty. Grandstream will NOT take any responsibility for this kind of damage. Please be very CAREFUL when doing firmware upgrade. Which firmware version are you using on your spa3000? Peter Hoppe wrote: || Hello! || || This is actually less a question than some information, if anyone else || struggles with the same issue. || || I am located in the UK and use a Sipura-3000 adapter to connect to a BT || line (via fxo port). One problem I had was that disconnect supervision || didn't work: || || Some caller phones me (my adapter) || adapter goes off-hook (answers call) || caller hangs up || adapter doesn't realize and stays off hook. -- dyslexics of the world - untie ! ___ --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] unable to make calls out using [EMAIL PROTECTED]
I am setting up a phone system using [EMAIL PROTECTED], version 1.5. It runs Asterisk 1.0.9 built by [EMAIL PROTECTED] on a i686 running Linux (Asterisk info). I had some bigger problems: In AmpPortal / Setup/ Extensions: When I added new SIP devices and then looked at the resulting sip.conf I saw that the file got messed up - per extension settings were duplicated. As result the SIP devices didn't register anymore. I then hand edited my sip.conf and devices did register successfully. I then added estensions, but when I tried to initiate phone calls, no phone rang. So I hand edited extensions.conf as well, and lo-and-behold it worked! Since I have some tighter deadline I decided that it wasn't woth trying to use the AMP-portal way of things and simply scrapped the config files which were offered and to use hand edited files instead. System works very well now (except some features I still have to implement). Despite of all this I am NOT disappointed about [EMAIL PROTECTED] - I think it's a great software package, and I am very grateful that there are people who take the trouble of setting all that up and to offer it in such an easy-to-install package. The most likely reason that it didn't work is probably my own ignorance. There would probably be thousands of people who successfully used [EMAIL PROTECTED] as well. I just didn't have time to fiddle with it. But the system works now fine with hand edited files. Peter ___ --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] Stay away from Grandstream!
Stay away from Grandstream and AddPac. These are some of the companies with undereducated software developers that have problems with understanding written english, mainly the SIP RFC documents. I learned this the hard way, wasting half a year with helping them fix problems which shouldn't be there if they have had read/implemented the RFC correctly. Basically, they sell beta quality hardware and then you co-share their final firmware development costs by providing free testing/QA. I blame their sales management for pushing developers to release without proper testing. GXP2000 is much more buggy echo-can wise than the earlier models. For now, I'm back to more expensive equipment. We're not that rich to pay twice. HTH, Vahan Avi Miller wrote: Brian Capouch wrote: They don't perform as well as the expensive Ciscos and Polycoms, but many of us are using them in a variety of circumstances quite happily. I have 4 of them in a small office (GXP2000) running 1.0.12 and they're just fine for our purposes. As Brian said, YMMV. For our 60-person office in Sydney, I'm probably going to use a mix of Polycom/Grandstream and softphones. cYa, Avi ___ --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] 4-port external sip fxo which doesnt suck?
Go with SPA-3000. While it's much more awkward to maintain, they're rock stable and provide the features they advertise for. I'd also add AddPac VoiceFinder series as being not 100% asterisk compatible, expensive and not worth your time (learned this the hard way). It took me 6 months to persuade AddPac that each FXO/FXS has to use unique Call-ID on the same gateway device to work properly with Asterisk and other properly written SIP proxies etc. HTH, Vahan [EMAIL PROTECTED] wrote: I'm looking for a 4-port external sip fxo which doesn't suck. o) Clipcomm CG-410. Poor reviews. o) Mediatrix 1204. Very poor reviews. o) Audiocodes MP104. Poor reviews. o) DLink DVG-3004S. Doesnt seem to exist yet. Is anyone actually using a 4 port external sip fxo which doesn't suck? It almost seems better to buy a pile of SPA-3000 and use them for just SIP FXO. -Dan ___ --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
[Asterisk-Users] UK, Disconnect supervision
I found some more information on UK settings with sipura-3000 on http://www.voip-info.org/wiki/view/Sipura+3000 They point to a document with UK specific settings for spa-3000. This document is at http://www.provu.co.uk/pdf/sipura/sipura_uk_regional_settings.pdf Hope this helps others. These resources handle disconnect settings as well. I'll be trying the disconnect tone settings they suggest. Peter Jonathan Attwood jmattwood at gmail.com Tue Dec 27 16:26:59 CST 2005 Which firmware version are you using on your spa3000? Peter Hoppe wrote: || Hello! || || This is actually less a question than some information, if anyone else || struggles with the same issue. || || I am located in the UK and use a Sipura-3000 adapter to connect to a BT || line (via fxo port). One problem I had was that disconnect supervision || didn't work: || || Some caller phones me (my adapter) || adapter goes off-hook (answers call) || caller hangs up || adapter doesn't realize and stays off hook. ___ --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] Stay away from Grandstream!
My only experience is with their Budgetone 102. You basically get you pay for. I have since purchased a pair of Aastra 480i. Much much better. I am going to put the Budgetone on ebay, no point dealing with all the hassle. The main issue for me was actually not sofware but rather the design of the handset. Vahan Yerkanian wrote: Stay away from Grandstream and AddPac. These are some of the companies with undereducated software developers that have problems with understanding written english, mainly the SIP RFC documents. I learned this the hard way, wasting half a year with helping them fix problems which shouldn't be there if they have had read/implemented the RFC correctly. Basically, they sell beta quality hardware and then you co-share their final firmware development costs by providing free testing/QA. I blame their sales management for pushing developers to release without proper testing. GXP2000 is much more buggy echo-can wise than the earlier models. For now, I'm back to more expensive equipment. We're not that rich to pay twice. HTH, Vahan Avi Miller wrote: Brian Capouch wrote: They don't perform as well as the expensive Ciscos and Polycoms, but many of us are using them in a variety of circumstances quite happily. I have 4 of them in a small office (GXP2000) running 1.0.12 and they're just fine for our purposes. As Brian said, YMMV. For our 60-person office in Sydney, I'm probably going to use a mix of Polycom/Grandstream and softphones. cYa, Avi ___ --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] Stay away from Grandstream!
Hmmm... I feel that this is a little unfair towards GrandStream and other like vendors. Any vendor on the market has issues with their firmware, I can list many: Sipura/LinkSys SPA 841 (Latest firmware): 1. Phone doesn't re-register upon network loss 2. Phone firware becomes stalled, without any indication of an error while all functions continue working 3. Transfer function doesn't work as it should 4. MWI doesn't always work correctly 5. I can really go on and on... WellTech (Latest firmware): 1. Support for g729 is buggy 2. Echo cancel is buggy and causes ATA to crash 3. IP phones have no ability to re-configure the function keys on the box 4. Transfer/Conference buttons don't do anytning I can go on and on with other vendors, including Cisco, Nortel and more. The thing I'm saying is that any phone you'd test would run into issues at some time or other - claiming to stay away from one or another causes you to not even consider alternatives, thus at the end, you reach the Microsoft way of thinking. Last week, I got a phone to test with called a MicroNet. Actually, I got 3 phones, all from Micronet. I started them up, found out that 2 of them were actually WellTech phones (well, the shape told me, I hoped the firmware will be different, but I found out wrong). The third phone was different. It's called a Micronet SP5106 which to my surprise, worked almost flawlessly out of the box. It took me a while to configure the network correctly, and to understand the logic of the menu, but after that, the rest was easy. Transfer, 3-Way conference, Forward, DND, VoiceMail button, everything worked. What didn't work was configurable from the web backend - in other words: I couldn't find a flaw (yet). The only flaw I did find was this: the phone has the ability to connect to 3 SIP accounts at the same time. Upon defining a new account, you need to physically RESET the phone, other than that, the phone works just fine. I'll be posting a full review on my blog at http://www.net-gurus.net Regards, Nir S Vahan Yerkanian wrote: Stay away from Grandstream and AddPac. These are some of the companies with undereducated software developers that have problems with understanding written english, mainly the SIP RFC documents. I learned this the hard way, wasting half a year with helping them fix problems which shouldn't be there if they have had read/implemented the RFC correctly. Basically, they sell beta quality hardware and then you co-share their final firmware development costs by providing free testing/QA. I blame their sales management for pushing developers to release without proper testing. GXP2000 is much more buggy echo-can wise than the earlier models. For now, I'm back to more expensive equipment. We're not that rich to pay twice. HTH, Vahan Avi Miller wrote: Brian Capouch wrote: They don't perform as well as the expensive Ciscos and Polycoms, but many of us are using them in a variety of circumstances quite happily. I have 4 of them in a small office (GXP2000) running 1.0.12 and they're just fine for our purposes. As Brian said, YMMV. For our 60-person office in Sydney, I'm probably going to use a mix of Polycom/Grandstream and softphones. cYa, Avi ___ --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] UK, Disconnect supervision
Peter, I'm using the firmware 3.1.5(GWb) and was wondering if your suggestions would be of any benefit to me. Incidentally, I've never had an issue upgrading or downgrading the firmware in 2 spa-3000s, I just had to make sure the unit had only just been powered up when initiating the upgrade. (YMMV) Anyway, if you're wanting somewhere else to read ask questions have a look at http://voxilla.com/forum-viewforum-f-14.html for Sipura/Linksys adapters or http://voxilla.com/PNphpBB2-viewforum-f-17.html for asterisk. Fantastic resources with helpful knowledgeable respondents. Peter Hoppe wrote: || Jonathan, || || many thanks for your reply. The adapter has firmware version || 3.1.3(GWa). || || Does that version have problems with diconnect tones? Would you || recommend I should upgrade? Can you give me some reasons or point me to || resources (apart from google) where I can research further? What would || you say are the risks of upgrading? || || I am usually a bit anxious about firmware upgrade because I have that || fixed idea that EITHER the new firmware may break other features || (like - registration problems with SIP provider, connectivity issues || and so on) OR there may be some problem during firmware upgrade || which damages the device in question. For example, for the Grandstream || Budgetone 100 phone, power outage during firmware upgrade from TFTP || will damage the device(1). And I can't fix it once it's broken; it's || not like a computer where I simply reinstall the OS / put in a new || component etc. Once it's gone, it's gone. || My fears are probably totally unfounded, but better safe than sorry. || So I wouldn't upgrade unless there are good reasons to do so (if it || ain't broke, don't fix it). || || But thanks very much for that hint. I actually have two other || adapters, and they may be way out of date: 2.0.13(GWg) - so they may || really need updating. || || Peter || || -- || (1)BudgeTone-100 User Manual, version 1.0.5.11, section 6.1: || Upgrade with TFTP, warning: The device WILL get damaged if there is a || power outage during firmware upgrade. Grandstream STRONGLY recommend || customer maintain UNINTERRUPTED POWER SUPPLY during firmware upgrade. || This damage is NOT covered by the manufacture warranty. Grandstream || will NOT take any responsibility for this kind of damage. Please be || very CAREFUL when doing firmware upgrade. || || || ||| Which firmware version are you using on your spa3000? ||| ||| Peter Hoppe wrote: | Hello! | | This is actually less a question than some information, if anyone | else struggles with the same issue. | | I am located in the UK and use a Sipura-3000 adapter to connect to | a BT line (via fxo port). One problem I had was that disconnect | supervision didn't work: | | Some caller phones me (my adapter) | adapter goes off-hook (answers call) | caller hangs up | adapter doesn't realize and stays off hook. || || || -- || dyslexics of the world - untie ! || ___ || --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] Automatic logoff of all agents at set time
Hi, I have not been able to find anything about persistent agents in any wiki? Where does this command go and what is its syntax? Thanks Michiel van Baak wrote: On 18:06, Tue 27 Dec 05, Bud Bach wrote: But, if the agents don't log out for some reason, they will still be logged in the next time the queue opens even if they aren't there right? yes. What you can do is 2 things: * you can set the autologoff time in agents.conf. This can give you some trouble when agents go to the toilet or grab a cup of coffee. * set persistant agents to off and restart asterisk at midnight. This will logoff the agents :) Hope this helps ___ --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] Automatic logoff of all agents at set time
It is set in the queues.conf file. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chuck Bunn Sent: Wednesday, December 28, 2005 8:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Automatic logoff of all agents at set time Hi, I have not been able to find anything about persistent agents in any wiki? Where does this command go and what is its syntax? Thanks Michiel van Baak wrote: On 18:06, Tue 27 Dec 05, Bud Bach wrote: But, if the agents don't log out for some reason, they will still be logged in the next time the queue opens even if they aren't there right? yes. What you can do is 2 things: * you can set the autologoff time in agents.conf. This can give you some trouble when agents go to the toilet or grab a cup of coffee. * set persistant agents to off and restart asterisk at midnight. This will logoff the agents :) Hope this helps ___ --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] Dialling out with clone X100P board
Hi Ryan:
Christmas intervened!
Got it working. It turned out not to be the ww that did it, but the
toneduration parameter in the zapata.conf file.
Setting
toneduration=200
did the trick.
Thanks for the help, hope this tip helps someone else later on.
Happy New Year!
Roger
[EMAIL PROTECTED] wrote:
I had the same problem at first. Try adding a w or two before the
${EXTEN}. That makes it wait a little bit before sending the DTMF numbers.
Here is the dial() I'm using:
Dial(ZAP/1/ww${EXTEN})
Try it out and see. Let us know if it works.
Ryan
Hi all :
I need a little help please.
I have a clone X100P board. I have it all set up and working (just
testing so far) for incoming calls from PSTN.
For outgoing to PSTN I have a strange problem.
I dial out OK, the Zap channel answers the SIP channel ok, (But I do not
see a Call bridged message, and the call has some strange charateristics.
If I call 123, I can connect to and hear the time clock provided by BT
(I'm in the UK) Is this 'audio before answer'?)
If I call any other external number, eg my cellphone, it never rings,
and after 30 secs or so the Zap channel hangs up.
I have been testing this with a very simple Dial(ZAP/1/${EXTEN}) command.
What should I be looking for in my setup?
Many thanks, and happy Christmas to all.
Roger
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--
Roger Hill 07739 707 180
Perseverance is the hard work you do after you get
tired of doing the hard work you already did.
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Re: [Asterisk-Users] Re: Why no sound from festival?
Why don't you send this to the offender instead of the list? From: Tomislav Parcina [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Why no sound from festival? Date: Wed, 28 Dec 2005 12:07:52 +0100 To: asterisk-users@lists.digium.com Please stop replaying to mesage. If you plan to open thread do so by writing mail to this address asterisk-users@lists.digium.com -- Tomislav Parcina [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 ---End of Original Message- ___ --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] Stay away from Grandstream!
I think the unfairness stems from Grandstreams generally being people's first IP phone - it seems like a cheap entry point to try things out. They then falsely assume everything else has to be better, especially if it has a higher price tag. Wrong. The standard for VoIP phones is total crap. Anything rising even slightly above that level wins awards for excellence. :-) Steve Nir Simionovich wrote: Hmmm... I feel that this is a little unfair towards GrandStream and other like vendors. Any vendor on the market has issues with their firmware, I can list many: Sipura/LinkSys SPA 841 (Latest firmware): 1. Phone doesn't re-register upon network loss 2. Phone firware becomes stalled, without any indication of an error while all functions continue working 3. Transfer function doesn't work as it should 4. MWI doesn't always work correctly 5. I can really go on and on... WellTech (Latest firmware): 1. Support for g729 is buggy 2. Echo cancel is buggy and causes ATA to crash 3. IP phones have no ability to re-configure the function keys on the box 4. Transfer/Conference buttons don't do anytning I can go on and on with other vendors, including Cisco, Nortel and more. The thing I'm saying is that any phone you'd test would run into issues at some time or other - claiming to stay away from one or another causes you to not even consider alternatives, thus at the end, you reach the Microsoft way of thinking. Last week, I got a phone to test with called a MicroNet. Actually, I got 3 phones, all from Micronet. I started them up, found out that 2 of them were actually WellTech phones (well, the shape told me, I hoped the firmware will be different, but I found out wrong). The third phone was different. It's called a Micronet SP5106 which to my surprise, worked almost flawlessly out of the box. It took me a while to configure the network correctly, and to understand the logic of the menu, but after that, the rest was easy. Transfer, 3-Way conference, Forward, DND, VoiceMail button, everything worked. What didn't work was configurable from the web backend - in other words: I couldn't find a flaw (yet). The only flaw I did find was this: the phone has the ability to connect to 3 SIP accounts at the same time. Upon defining a new account, you need to physically RESET the phone, other than that, the phone works just fine. I'll be posting a full review on my blog at http://www.net-gurus.net Regards, Nir S Vahan Yerkanian wrote: Stay away from Grandstream and AddPac. These are some of the companies with undereducated software developers that have problems with understanding written english, mainly the SIP RFC documents. I learned this the hard way, wasting half a year with helping them fix problems which shouldn't be there if they have had read/implemented the RFC correctly. Basically, they sell beta quality hardware and then you co-share their final firmware development costs by providing free testing/QA. I blame their sales management for pushing developers to release without proper testing. GXP2000 is much more buggy echo-can wise than the earlier models. For now, I'm back to more expensive equipment. We're not that rich to pay twice. HTH, Vahan Avi Miller wrote: Brian Capouch wrote: They don't perform as well as the expensive Ciscos and Polycoms, but many of us are using them in a variety of circumstances quite happily. I have 4 of them in a small office (GXP2000) running 1.0.12 and they're just fine for our purposes. As Brian said, YMMV. For our 60-person office in Sydney, I'm probably going to use a mix of Polycom/Grandstream and softphones. ___ --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] Sipura 2002 codec preferences
I am about to sent some Sipura 2002 ATAs out to a call center. I want to use the dual line capability of the units, but I realize that the second channel will not be able to use G729 simultaneously. What do you think would the best option be for that channel? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [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
[Asterisk-Users] UK, Disconnect supervision
Thanks very much! I'll definitely sook at these resources as well if other problems come up. I also found some more info on sipura setup un the UK, see http://lists.digium.com/pipermail/asterisk-users/2005-December/140037.html Thank you very much again for your comments. Good to hear that you had no issues with firmware upgrades. I feel a bit more encouraged about it. God bless, Peter Peter, I'm using the firmware 3.1.5(GWb) and was wondering if your suggestions would be of any benefit to me. Incidentally, I've never had an issue upgrading or downgrading the firmware in 2 spa-3000s, I just had to make sure the unit had only just been powered up when initiating the upgrade. (YMMV) Anyway, if you're wanting somewhere else to read ask questions have a look at http://voxilla.com/forum-viewforum-f-14.html for Sipura/Linksys adapters or http://voxilla.com/PNphpBB2-viewforum-f-17.html for asterisk. Fantastic resources with helpful knowledgeable respondents. ___ --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] Stay away from Grandstream!
On Tuesday 27 December 2005 21:52, Erick Baum wrote: that, there is now a bad echo if one of the GXP users turns their volume up too high, the other party can hear an echo. If the GXP user turns their I'm afraid you're going to find this with pretty much *every* phone. Normal POTS phones just don't have this problem because the delay is low. I would imagine that Polycom would have the least problem with this since they are known for their superior audio, but as I do not own any and have not used any, I cannot say for certainty. Honestly you said it yourself though... they are turning it up too high and pushing the audio beyond what its design specifications are. This is perhaps the fault of the software guys, as they allow you to go beyond what what the acoustic coupling was good for, but then again I am pretty sure they allowed the volume to be increased due to customer complaints of the phones being too quiet. :-) we were forced to use the AC adapter. And many of the other phones suffer from all kinds of stupid little intermittent issues such as dropped calls, reboots and strange ticking and static on the line, even on internal calls. We discovered that quite a few of the network cables that came with the phones seemed to be faulty, which when replacing them seemed to solve some of our dropped calls and spontaneous reboot problems. Some of the phones had bad handset cables. Replacing some of those seemed to get rid of some of the static issues. We've replaced several of the really troublesome phones with Cisco's or Polycom's, and what do you know, no problems whatsoever. Well when you buy on price alone... -A. ___ --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] ipVolution
Hi, Anybody have some experience and did some testing with ipVolution E1/T1 cards? goran ___ --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] Automatic logoff of all agents at set time
Hi, When I add 'persistentmembers=no' in queues.conf and reload I get a message in the message log file saying unknown keyword 'persistentmembers'. I got the syntax from http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue under the notes section. Thanks Alexander Lopez wrote: It is set in the queues.conf file. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chuck Bunn Sent: Wednesday, December 28, 2005 8:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Automatic logoff of all agents at set time Hi, I have not been able to find anything about persistent agents in any wiki? Where does this command go and what is its syntax? Thanks Michiel van Baak wrote: On 18:06, Tue 27 Dec 05, Bud Bach wrote: But, if the agents don't log out for some reason, they will still be logged in the next time the queue opens even if they aren't there right? yes. What you can do is 2 things: * you can set the autologoff time in agents.conf. This can give you some trouble when agents go to the toilet or grab a cup of coffee. * set persistant agents to off and restart asterisk at midnight. This will logoff the agents :) Hope this helps ___ --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] 4-port external sip fxo which doesnt suck?
Just a small word of caution on the spa3k... It too has issues with handling echo cancellation on long loops. Or, maybe I should say on loops with long echo tails. Sipura/Linksys tech support has suggested downgrading from the current 3.1.7 firmware to 3.1.3a to improve the issue. I've not heard of anyone having issues on shorter pstn loops, and I've not been able to define long vs short loops. Go with SPA-3000. While it's much more awkward to maintain, they're rock stable and provide the features they advertise for. I'd also add AddPac VoiceFinder series as being not 100% asterisk compatible, expensive and not worth your time (learned this the hard way). It took me 6 months to persuade AddPac that each FXO/FXS has to use unique Call-ID on the same gateway device to work properly with Asterisk and other properly written SIP proxies etc. HTH, Vahan [EMAIL PROTECTED] wrote: I'm looking for a 4-port external sip fxo which doesn't suck. o) Clipcomm CG-410. Poor reviews. o) Mediatrix 1204. Very poor reviews. o) Audiocodes MP104. Poor reviews. o) DLink DVG-3004S. Doesnt seem to exist yet. Is anyone actually using a 4 port external sip fxo which doesn't suck? It almost seems better to buy a pile of SPA-3000 and use them for just SIP FXO. -Dan ___ --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 ---End of Original Message- ___ --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] Sipura 2002 codec preferences
I am about to sent some Sipura 2002 ATAs out to a call center. I want to use the dual line capability of the units, but I realize that the second channel will not be able to use G729 simultaneously. What do you think would the best option be for that channel? You might double check that. I thought someone commented the 2002 was a hardware upgrade that did support both ports running g729 now. I don't have one to test though. ___ --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] Automatic logoff of all agents at set time
Chuck Bunn wrote: When I add 'persistentmembers=no' in queues.conf and reload I get a message in the message log file saying unknown keyword 'persistentmembers'. I got the syntax from http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue under the notes section. You haven't told us what version of Asterisk you are using, but you are probably using 1.0.x, if it doesn't support that option. Regardless, that option won't do what you want anyway, since you are using agents and not dynamic queue members. The 'persistentagents' option in agents.conf could do it, but that's still an ugly way to handle it. Since agents can be logged off using CLI commands or manager interface actions, it would be quite simple to write a script to run via a cron job late at night to forcibly log off all your agents. ___ --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] oh323 configuration
It's possible to register oh323 with gnugk ? Any one knows one good oh323 how to? Regards, -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.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] voip-info: Asterisk record calls
On this page http://www.voip-info.org/wiki-Asterisk+record+calls there is Example by Mojo. I have done everything he said and I have sox package installed. [EMAIL PROTECTED] recordings]# sox -help sox: Version 12.17.7 ... When I open this web page http://10.0.0.26/recordings/index.php I get this: No Recordings Found And there are recordings in /var/spool/asterisk/monitor Do I have to do something more? Does it work for anybody else? Is there any other way to combine in and out soundfile when I use automon option? -- Tomislav Parcina [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
Re: [Asterisk-Users] oh323 configuration
Sorry... the question is related with ooh323 It's possible to register ooh323 with gnugk ? Any on knows one good ooh323 how to? On Wed, 2005-12-28 at 09:48 -0500, Guillermo Salas M wrote: It's possible to register oh323 with gnugk ? Any one knows one good oh323 how to? Regards, -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.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
Re: [Asterisk-Users] Automatic logoff of all agents at set time
Hi, Oh sorry I am using asterisk 1.2.1 Thanks Kevin P. Fleming wrote: Chuck Bunn wrote: When I add 'persistentmembers=no' in queues.conf and reload I get a message in the message log file saying unknown keyword 'persistentmembers'. I got the syntax from http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue under the notes section. You haven't told us what version of Asterisk you are using, but you are probably using 1.0.x, if it doesn't support that option. Regardless, that option won't do what you want anyway, since you are using agents and not dynamic queue members. The 'persistentagents' option in agents.conf could do it, but that's still an ugly way to handle it. Since agents can be logged off using CLI commands or manager interface actions, it would be quite simple to write a script to run via a cron job late at night to forcibly log off all your agents. ___ --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
[Asterisk-Users] who is online
hi all i use asdterisk in my company with Flash Panel Operator to know who is talking or ringing. But i dont know any web application to know who is online or offline. any body know any webapp for that ? -- thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: voip-info: Asterisk record calls
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Do I have to do something more? Does it work for anybody else? Is there any other way to combine in and out soundfile when I use automon option? My error. Everything works fine... Sorry. -- Tomislav Parcina [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
Re: [Asterisk-Users] PHP Manager
AFAIK you need to use different actions for each command., sending 3 commands in the same action wont work. I have no problems to issue commands, originates etc.On 12/28/05, Code Lover [EMAIL PROTECTED] wrote: Hi all,I have a small problem to execute Asterisk Commands in AsteriskManager using PHP. I am able to run all Asterisk Manager command but the problem iscomming with asterisk command.here is the code i am trying to run.?php $socket = fsockopen(localhost,5038, $errno, $errstr, $timeout); fputs($socket, Action: Login\r\n); fputs($socket, UserName: 1212\r\n); fputs($socket, Secret: 1212\r\n\r\n); fputs($socket, Action: Command\r\n); fputs($socket, Command: reload\r\n\r\n); #Working well fputs($socket, Command: show channels\r\n\r\n); #Not working Working well fputs($socket, Command: 'show channels'\r\n\r\n); #Not working Working well $wrets=fgets($socket,128); ?If you see in my code when i am calling only reload command workingbut when i am trying to call piar command it is just prompting :== Manager '1212' logged off from localhost without showing channelsPlease advice me to solve this problem.--Thank You,Code Lover___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___ --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] Sipura 2002 codec preferences
You can use both channel as G.726/32 at the same time, or lower than 32. Is the best solution we found. Regards, Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) Sent: Wednesday, December 28, 2005 8:44 AM To: Asterisk-Users Subject: [Asterisk-Users] Sipura 2002 codec preferences I am about to sent some Sipura 2002 ATAs out to a call center. I want to use the dual line capability of the units, but I realize that the second channel will not be able to use G729 simultaneously. What do you think would the best option be for that channel? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [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] who is online
Pablo Allietti wrote: hi all i use asdterisk in my company with Flash Panel Operator to know who is talking or ringing. But i dont know any web application to know who is online or offline. any body know any webapp for that ? Flash Operator Panel _is_ a web application. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call transfer
I'm not sure how this is suppose to work. But I want to be able to call people from a SIP phone and transfer them into a conference room. If I call another extension that is a SIP phone I can hit # and then enter the conference room number. If I call from the PSTN to the SIP extension phone I can transfer by hitting # too. But if I call from the SIP phone extension to a PSTN number it doesn't do anything when I hit the #. I'm using [EMAIL PROTECTED] and under general settings I have tTrwW for Asterisk Dial Command Settings. Can you call through a Zap trunk from a SIP phone and do a call transfer? -- Michael Sampson Information Systems Manager Customer Contact Services [EMAIL PROTECTED] 952-936-4000 ___ --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: who is online
On Wed, Dec 28, 2005 at 09:15:15AM -0600, Kevin P. Fleming wrote: Pablo Allietti wrote: hi all i use asdterisk in my company with Flash Panel Operator to know who is talking or ringing. But i dont know any web application to know who is online or offline. any body know any webapp for that ? Flash Operator Panel _is_ a web application. sure. but dont have the online and offline applet? or maybe have and i dont know how to configure it? ___ --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 ---end quoted text--- -- .- Pablo Allietti E-mail: [EMAIL PROTECTED] | LACNIC Phone : +598 2 604 | http://LACNIC.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] Stay away from Grandstream!
I agree, GrandStream does seem to become the poor man's VoIP solution - making the bar for other VoIP phones very low to pass. I believe that GrandStream have a very good chance to basically being bought by a bigger company, like what happened to Sipura. What would happen then would be that people would say: Oh, GrandStream, Very good - after all bought them. I found that sometimes the most surprising hardware comes from non-known companies, like PerfecTone or Micronet. I think the main thing is the try things out, and find out what is the best suited IPhone for you. Nir S Steve Underwood wrote: I think the unfairness stems from Grandstreams generally being people's first IP phone - it seems like a cheap entry point to try things out. They then falsely assume everything else has to be better, especially if it has a higher price tag. Wrong. The standard for VoIP phones is total crap. Anything rising even slightly above that level wins awards for excellence. :-) Steve Nir Simionovich wrote: Hmmm... I feel that this is a little unfair towards GrandStream and other like vendors. Any vendor on the market has issues with their firmware, I can list many: Sipura/LinkSys SPA 841 (Latest firmware): 1. Phone doesn't re-register upon network loss 2. Phone firware becomes stalled, without any indication of an error while all functions continue working 3. Transfer function doesn't work as it should 4. MWI doesn't always work correctly 5. I can really go on and on... WellTech (Latest firmware): 1. Support for g729 is buggy 2. Echo cancel is buggy and causes ATA to crash 3. IP phones have no ability to re-configure the function keys on the box 4. Transfer/Conference buttons don't do anytning I can go on and on with other vendors, including Cisco, Nortel and more. The thing I'm saying is that any phone you'd test would run into issues at some time or other - claiming to stay away from one or another causes you to not even consider alternatives, thus at the end, you reach the Microsoft way of thinking. Last week, I got a phone to test with called a MicroNet. Actually, I got 3 phones, all from Micronet. I started them up, found out that 2 of them were actually WellTech phones (well, the shape told me, I hoped the firmware will be different, but I found out wrong). The third phone was different. It's called a Micronet SP5106 which to my surprise, worked almost flawlessly out of the box. It took me a while to configure the network correctly, and to understand the logic of the menu, but after that, the rest was easy. Transfer, 3-Way conference, Forward, DND, VoiceMail button, everything worked. What didn't work was configurable from the web backend - in other words: I couldn't find a flaw (yet). The only flaw I did find was this: the phone has the ability to connect to 3 SIP accounts at the same time. Upon defining a new account, you need to physically RESET the phone, other than that, the phone works just fine. I'll be posting a full review on my blog at http://www.net-gurus.net Regards, Nir S Vahan Yerkanian wrote: Stay away from Grandstream and AddPac. These are some of the companies with undereducated software developers that have problems with understanding written english, mainly the SIP RFC documents. I learned this the hard way, wasting half a year with helping them fix problems which shouldn't be there if they have had read/implemented the RFC correctly. Basically, they sell beta quality hardware and then you co-share their final firmware development costs by providing free testing/QA. I blame their sales management for pushing developers to release without proper testing. GXP2000 is much more buggy echo-can wise than the earlier models. For now, I'm back to more expensive equipment. We're not that rich to pay twice. HTH, Vahan Avi Miller wrote: Brian Capouch wrote: They don't perform as well as the expensive Ciscos and Polycoms, but many of us are using them in a variety of circumstances quite happily. I have 4 of them in a small office (GXP2000) running 1.0.12 and they're just fine for our purposes. As Brian said, YMMV. For our 60-person office in Sydney, I'm probably going to use a mix of Polycom/Grandstream and softphones. ___ --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] Wrong Password?????
Does adding the line nat=yes into your sip.conf file help? Leah Newmark Capalon www.capalon.com Message: 21 Date: Wed, 28 Dec 2005 11:48:25 +0100 From: Rafael Ledesma [EMAIL PROTECTED] Subject: [Asterisk-Users] Wrong Password? To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii I've been working with asterisk with public ip but since I change this and put asterisk behind NAT, get this error when my hard phones try to register Dec 28 11:43:33 NOTICE[8716]: chan_sip.c:10817 handle_request_register: Registration from 'sip:[EMAIL PROTECTED]:5060' failed for '...' - Wrong password [general] section of sip.conf file looks like this: [general] bindport=5060 bindaddr=0.0.0.0 dtmfmode=rfc2833 context=incoming ; Default for incoming calls externip=... localnet=10.0.2.0/255.255.255.0 ___ --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: who is online
you need to set the extensions paramters to qualify=yes or qualify=integer and then FOP (flash operator panel) will reflect the status of the extensions. Pablo Allietti wrote: On Wed, Dec 28, 2005 at 09:15:15AM -0600, Kevin P. Fleming wrote: Pablo Allietti wrote: hi all i use asdterisk in my company with Flash Panel Operator to know who is talking or ringing. But i dont know any web application to know who is online or offline. any body know any webapp for that ? Flash Operator Panel _is_ a web application. sure. but dont have the online and offline applet? or maybe have and i dont know how to configure it? ___ --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 ---end quoted text--- -- Adrian Carter Technical Manager Leading Edge Internet Web http://www.lei.net.au http://support.lei.net.au Direct+61 2 6163 6162 Support 1 300 662 415 E-mail[EMAIL PROTECTED] -- Adrian Carter Technical Manager Leading Edge Internet Web http://www.lei.net.au http://support.lei.net.au Direct+61 2 6163 6162 Support 1 300 662 415 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
Re: [Asterisk-Users] Re: who is online
qualify=yes in both sip.conf and iax.conf, seems to highlight both the users and trunks who are currently available in FOP Bails Pablo Allietti wrote: On Wed, Dec 28, 2005 at 09:15:15AM -0600, Kevin P. Fleming wrote: Pablo Allietti wrote: hi all i use asdterisk in my company with Flash Panel Operator to know who is talking or ringing. But i dont know any web application to know who is online or offline. any body know any webapp for that ? Flash Operator Panel _is_ a web application. sure. but dont have the online and offline applet? or maybe have and i dont know how to configure it? ___ --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 ---end quoted text--- ___ --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: who is online
The 'status' is only as goodas the frequency of the qualify periodand you can say hello to a LOT of SIP OPTIONS messages being sent from Asterisk to each phone. -Original Message-From: Adrian Carter [mailto:[EMAIL PROTECTED]Sent: Wednesday, December 28, 2005 8:36 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Re: who is onlineyou need to set the extensions paramters to qualify=yes or qualify=integer and then FOP (flash operator panel) will reflect the status of the extensions.Pablo Allietti wrote: On Wed, Dec 28, 2005 at 09:15:15AM -0600, Kevin P. Fleming wrote: Pablo Allietti wrote: hi all i use asdterisk in my company with Flash Panel Operator to know who is talking or ringing. But i dont know any web application to know who is online or offline. any body know any webapp for that ? Flash Operator Panel _is_ a web application. sure. but dont have the online and offline applet? or maybe have and i dont know how to configure it? ___ --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 ---end quoted text--- -- Adrian Carter Technical Manager Leading Edge Internet Web http://www.lei.net.au http://support.lei.net.au Direct+61 2 6163 6162 Support 1 300 662 415 E-mail[EMAIL PROTECTED] -- Adrian Carter Technical Manager Leading Edge Internet Web http://www.lei.net.au http://support.lei.net.au Direct+61 2 6163 6162 Support 1 300 662 415 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
Re: [Asterisk-Users] Blackberry SIM card
My thanks to all the great people that have submitted replys. I have the information I need now. Bob Rawlinson Robert Rawlinson wrote: I acquired a Blackberry 7100T over Christmas. I had heard it will work with * and that is what I want to do with it. But I think it needs a SIM card to make it work. If this is true how do I go about getting a SIM card for it and how to set it up? Thanks for any help you can offer. Bob Rawlinson ___ --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] Stay away from Grandstream!
Having tried EVERY single product from Grandstream, I don't think it's fair to judge Grandstream the way people do. I'm very happy with Grandstream products. As long as you upgrade the firmware they work fine. In fact they sometimes handle NAT better than any other device that I've tried (including ALL Sipura products). Grandstream is also one of very few to support ILBC codec, and BLF-support for Asterisk. If someone has tried the same, please comment. Bjorn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nir Simionovich Sent: Wednesday, December 28, 2005 4:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Stay away from Grandstream! I agree, GrandStream does seem to become the poor man's VoIP solution - making the bar for other VoIP phones very low to pass. I believe that GrandStream have a very good chance to basically being bought by a bigger company, like what happened to Sipura. What would happen then would be that people would say: Oh, GrandStream, Very good - after all bought them. I found that sometimes the most surprising hardware comes from non-known companies, like PerfecTone or Micronet. I think the main thing is the try things out, and find out what is the best suited IPhone for you. Nir S Steve Underwood wrote: I think the unfairness stems from Grandstreams generally being people's first IP phone - it seems like a cheap entry point to try things out. They then falsely assume everything else has to be better, especially if it has a higher price tag. Wrong. The standard for VoIP phones is total crap. Anything rising even slightly above that level wins awards for excellence. :-) Steve Nir Simionovich wrote: Hmmm... I feel that this is a little unfair towards GrandStream and other like vendors. Any vendor on the market has issues with their firmware, I can list many: Sipura/LinkSys SPA 841 (Latest firmware): 1. Phone doesn't re-register upon network loss 2. Phone firware becomes stalled, without any indication of an error while all functions continue working 3. Transfer function doesn't work as it should 4. MWI doesn't always work correctly 5. I can really go on and on... WellTech (Latest firmware): 1. Support for g729 is buggy 2. Echo cancel is buggy and causes ATA to crash 3. IP phones have no ability to re-configure the function keys on the box 4. Transfer/Conference buttons don't do anytning I can go on and on with other vendors, including Cisco, Nortel and more. The thing I'm saying is that any phone you'd test would run into issues at some time or other - claiming to stay away from one or another causes you to not even consider alternatives, thus at the end, you reach the Microsoft way of thinking. Last week, I got a phone to test with called a MicroNet. Actually, I got 3 phones, all from Micronet. I started them up, found out that 2 of them were actually WellTech phones (well, the shape told me, I hoped the firmware will be different, but I found out wrong). The third phone was different. It's called a Micronet SP5106 which to my surprise, worked almost flawlessly out of the box. It took me a while to configure the network correctly, and to understand the logic of the menu, but after that, the rest was easy. Transfer, 3-Way conference, Forward, DND, VoiceMail button, everything worked. What didn't work was configurable from the web backend - in other words: I couldn't find a flaw (yet). The only flaw I did find was this: the phone has the ability to connect to 3 SIP accounts at the same time. Upon defining a new account, you need to physically RESET the phone, other than that, the phone works just fine. I'll be posting a full review on my blog at http://www.net-gurus.net Regards, Nir S Vahan Yerkanian wrote: Stay away from Grandstream and AddPac. These are some of the companies with undereducated software developers that have problems with understanding written english, mainly the SIP RFC documents. I learned this the hard way, wasting half a year with helping them fix problems which shouldn't be there if they have had read/implemented the RFC correctly. Basically, they sell beta quality hardware and then you co-share their final firmware development costs by providing free testing/QA. I blame their sales management for pushing developers to release without proper testing. GXP2000 is much more buggy echo-can wise than the earlier models. For now, I'm back to more expensive equipment. We're not that rich to pay twice. HTH, Vahan Avi Miller wrote: Brian Capouch wrote: They don't perform as well as the expensive Ciscos and Polycoms, but many of us are using them in a variety of circumstances quite happily. I have 4 of them in a small office (GXP2000) running 1.0.12 and they're just fine for our purposes. As Brian said, YMMV. For our 60-person office in
[Asterisk-Users] Re: [Solved] who is online
On Thu, Dec 29, 2005 at 02:36:05AM +1100, Adrian Carter wrote: you need to set the extensions paramters to qualify=yes or qualify=integer and then FOP (flash operator panel) will reflect the status of the extensions. Pablo Allietti wrote: yep. this solve my problem Thanks!! On Wed, Dec 28, 2005 at 09:15:15AM -0600, Kevin P. Fleming wrote: Pablo Allietti wrote: hi all i use asdterisk in my company with Flash Panel Operator to know who is talking or ringing. But i dont know any web application to know who is online or offline. any body know any webapp for that ? Flash Operator Panel _is_ a web application. sure. but dont have the online and offline applet? or maybe have and i dont know how to configure it? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: [1]http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- Adrian Carter Technical Manager Leading Edge Internet Web [2]http://www.lei.net.au [3]http://support.lei.net.au Direct+61 2 6163 6162 Support 1 300 662 415 E-mail[EMAIL PROTECTED] -- Adrian Carter Technical Manager Leading Edge Internet Web [5]http://www.lei.net.au [6]http://support.lei.net.au Direct+61 2 6163 6162 Support 1 300 662 415 E-mail[EMAIL PROTECTED] References 1. http://lists.digium.com/mailman/listinfo/asterisk-users 2. http://www.lei.net.au/ 3. http://support.lei.net.au/ 4. mailto:[EMAIL PROTECTED] 5. http://www.lei.net.au/ 6. http://support.lei.net.au/ 7. mailto:[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 ---end quoted text--- -- .- Pablo Allietti E-mail: [EMAIL PROTECTED] | LACNIC Phone : +598 2 604 | http://LACNIC.NET pgpVCiTqSPhRv.pgp Description: PGP signature ___ --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] Cisco 7912G through NAT, problems with tones detection.
Diego Mariano Velo wrote: Hi, i have a cisco 7912G with SIP firmware, its connect to the asterisk through nat. The only problems is in the voice mailasterisk not detect the tones, therefore i cant access to my voice mail extension. Check the DTMF settings... http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf dtmfmode: inband | info | rfc2833 ___ --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] CallerID info needed
Hey everybody,
I'm trying to figure out a problem with Caller-ID info coming in from
one of our facilities. The Caller-ID name is all that comes across. I
figured out that I probably could do a database lookup against the name
and set the Caller-ID number to their extension. I'm using Asterisk
SVN-trunk-r7230 on a PRI connected to a Definity PBX.
When testing, my Polycom IP501 still shows unknown. A bit from the log
below:
-- Accepting call from '' to '4288' on channel 0/2, span 1
-- Executing DBget(Zap/2-1, CIDINFO=name/Ballard, Lance) in new
stack
-- DBget: varname=CIDINFO, family=name, key=Ballard, Lance
-- DBget: set variable CIDINFO to 4300
-- Executing NoOp(Zap/2-1, Setting CallerID Number to: 4300) in
new stack
-- Executing Set(Zap/2-1, CALLERID(Name)=Ballard, Lance) in new
stack
-- Executing Set(Zap/2-1, CALLERID(Number)=4300) in new stack
-- Executing SetGroup(Zap/2-1, Max_Calls) in new stack
-- Executing NoOp(Zap/2-1, Active Calls: 1) in new stack
-- Executing GotoIf(Zap/2-1, 0?103) in new stack
-- Executing Dial(Zap/2-1,
IAX2/bc.asterisk:[EMAIL PROTECTED]/4288||t) in new stack
-- Called bc.asterisk:[EMAIL PROTECTED]/4288
-- Call accepted by 192.168.102.15 (format gsm)
-- Format for call is gsm
-- IAX2/liv.asterisk-2 is ringing
And my code snip:
exten = _42XX,1,Dbget(CIDINFO=name/${CALLERIDNAME})
exten = _42XX,2,NoOp(Setting CallerID Number to: ${CIDINFO})
exten = _42XX,3,Set(CALLERID(Name)=${CALLERIDNAME})
exten = _42XX,4,Set(CALLERID(Number)=${CALLERIDNUM}${CIDINFO})
; Both variables with never have data at the same
time
exten = _42XX,5,SetGroup(Max_Calls)
exten = _42XX,6,NoOP(Active Calls: ${GROUP_COUNT(Max_Calls)})
exten = _42XX,7,GotoIf($[ ${GROUP_COUNT(Max_Calls)} 4 ]?103)
exten =
_42XX,8,Dial(IAX2/bc.asterisk:[EMAIL PROTECTED]/${EXTEN},,t)
exten = _42XX,103,Congestion()
Any suggestions on a fix (if it's possible)?
Thanks!
Doug
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[Asterisk-Users] call test
Hello, I need to test my configuration please to dial sip:[EMAIL PROTECTED] . Your call will be sent to a queue . Regards Harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.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
[Asterisk-Users] MWI problem
Sorry if I am always here asking for MWI, but I do not know how to solve this issue, I have my ATAs (Azatel 200 and Fritz!Box) that they think that I have a message waiting. Anyone knows how to solve this issue? Thank you ___ --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] What setup
I am an Asterisk newbie and don't have any telecom experience. I do know some about Linux and Windows as a sysadmin of Windows servers. I need to know what hardware to buy to replace a broken PBX. I have currently: -CBeyond as my carrier -16 port Cisco router with analog termination (not sure on terminology) into the building -Broken PBX: analog ATT Merlin system (not sure on model #, but could get it) -Second PBX: analog ATT Merlin system -8 extensions (~14 if you include the second ATT Merlin system serving our other business) -2 DIDs (~4 if you include our second business) -14 analog phones So, what are my options? I am looking for the cheapest/best solution. I could switch to a digital PRI or CAS line from my telco as another option, but I assume I would need to switch both PBXs and all phones to digital as well in that case. I need auto-attendant and music on hold, especially. Are these easy to set up? I installed Asterisk @ Home and it is running, but I need to RTFM to configure it. I could lose 1 DID for the faxing if we did fax-to-email. Please let me know. Thanks in advance! Sent by Go2net 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
[Asterisk-Users] BUG? AGI stuck in ast_waitfor_nandfds()
I just have upgraded from Asterisk 1.0.7 to 1.2.1 and im having problems with my AGI script that takes care about routing the calls. It worked perfectly for the last year with 1.0.7, now is getting stuck when the is launched. I have agi debug enabled and this is the output: -- Launched AGI Script /var/lib/asterisk/agi-bin/agi_cdr.php AGI Tx agi_request: agi_cdr.php AGI Tx agi_channel: SIP/25-63c8 AGI Tx agi_language: en AGI Tx agi_type: SIP AGI Tx agi_uniqueid: 1135787079.1 AGI Tx agi_callerid: 25 It just hangs there, does not finish sending the initial vars. From 'show channel SIP/25-63c8' i can see that is blocking in: Application: AGI Data: agi_cdr.php Blocking in: ast_waitfor_nandfds Any pointings? should i post this in the development list? should i open a bug in mantis? Distro: Gentoo-Linux Linux chewbacca 2.6.14.2 #1 SMP PREEMPT Wed Dec 28 09:09:20 CST 2005 i686 AMD Athlon(tm) XP 2400+ AuthenticAMD GNU/Linux Best Regards-- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___ --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] What setup
I am an Asterisk newbie and don't have any telecom experience. I do know some about Linux and Windows as a sysadmin of Windows servers. I need to know what hardware to buy to replace a broken PBX. I have currently: -CBeyond as my carrier -16 port Cisco router with analog termination (not sure on terminology) into the building -Broken PBX: analog ATT Merlin system (not sure on model #, but could get it) -Second PBX: analog ATT Merlin system -8 extensions (~14 if you include the second ATT Merlin system serving our other business) -2 DIDs (~4 if you include our second business) -14 analog phones So, what are my options? I am looking for the cheapest/best solution. I could switch to a digital PRI or CAS line from my telco as another option, but I assume I would need to switch both PBXs and all phones to digital as well in that case. I need auto-attendant and music on hold, especially. Are these easy to set up? I installed Asterisk @ Home and it is running, but I need to RTFM to configure it. I could lose 1 DID for the faxing if we did fax-to-email. Please let me know. Thanks in advance! Sent by Go2net 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
[Asterisk-Users] What setup
I am an Asterisk newbie and don't have any telecom experience. I do know some about Linux and Windows as a sysadmin of Windows servers. I need to know what hardware to buy to replace a broken PBX. I have currently: -CBeyond as my carrier -16 port Cisco router with analog termination (not sure on terminology) into the building -Broken PBX: analog ATT Merlin system (not sure on model #, but could get it) -Second PBX: analog ATT Merlin system -8 extensions (~14 if you include the second ATT Merlin system serving our other business) -2 DIDs (~4 if you include our second business) -14 analog phones So, what are my options? I am looking for the cheapest/best solution. I could switch to a digital PRI or CAS line from my telco as another option, but I assume I would need to switch both PBXs and all phones to digital as well in that case. I need auto-attendant and music on hold, especially. Are these easy to set up? I installed Asterisk @ Home and it is running, but I need to RTFM to configure it. I could lose 1 DID for the faxing if we did fax-to-email. Please let me know. Thanks in advance! Sent by Go2net 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
[Asterisk-Users] CALLERIDNUM
is it possible rewrite CALLERIDNUM in the ZAP channel? I use
[int-transfer]
exten = _00.,1,SetVar(CALLERIDNUM=${CALLNR})
exten = _00.,2,MYSQL(Connect connid localhost webcdr ser91623 cdr)
exten = _00.,3,MYSQL(Query resultid ${connid} select\
if((floor(u.credit/p.cost))1\,ceil((u.credit)/p.cost)*60\,0)\ as\
sekund\ from\ user\ u\,\ sip\ s\,\ pricelist\ p\ where\
u.iduser=s.iduser\ and\ s.idsip=\'${CALLERIDNUM}\'\ and\
p.acode=s.acode\ and\ u.currency=p.currency\ and\
right(left(\'${EXTEN}\'\,CHAR_LENGTH(p.ccode)+2)\,CHAR_LENGTH(p.ccode))\
like\ concat(p.ccode\,\'%\')\ order\ by\ p.ccode\ desc\ limit\ 1)
exten = _00.,4,MYSQL(Fetch foundRow ${resultid} sekund) ; fetch row
..
..
without success. At row 3 have var ${CALLERIDNUM} original value,
not value from ${CALLNR}.
--
[EMAIL PROTECTED]
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[Asterisk-Users] Driver not configuring correctly on TE210P for CCS
Hi all, It sas been a while since I have been on the mailing list but am really hoping someone can help me. Using Dell SC1420, Fedora 4, Asterisk 1.2.1 and a TE210P I cant seem to get the card to configure itself properly for CCS, it always appears as ESF in the /var/log/messages file, although ztcfg reports correctly. No compiles errors that I saw and OS has been updated etc. I am not new to asterisk but this has me stumped (googled and also emailed digium but probably due to the holidays not had a reply.) If I try to run asterisk it complains about not finding the bchannel on 24, note it should be configured to 16 so again I assume its not paying attention to the /etc/zaptel.cong Note: This exact config worked perfectly on asterisk 1.0 and a E100P card. var/log/messages: Dec 28 12:51:55 caudi_apx1 kernel: Zapata Telephony Interface Registered on major 196 Dec 28 12:51:55 caudi_apx1 kernel: ACPI: PCI Interrupt :03:0d.0[A] - GSI 49 (level, low) - IRQ 201 Dec 28 12:51:55 caudi_apx1 kernel: Found TE2XXP at base address ddfbff80, remapped to f882ef80 Dec 28 12:51:55 caudi_apx1 kernel: TE2XXP version c01a0164, burst OFF, slip debug: OFF Dec 28 12:51:55 caudi_apx1 kernel: FALC version: 0005, Board ID: 00 Dec 28 12:51:55 caudi_apx1 kernel: Reg 0: 0x32c0e400 Dec 28 12:51:55 caudi_apx1 kernel: Reg 1: 0x32c0e000 Dec 28 12:51:55 caudi_apx1 kernel: Reg 2: 0x Dec 28 12:51:55 caudi_apx1 kernel: Reg 3: 0x Dec 28 12:51:55 caudi_apx1 kernel: Reg 4: 0x0001 Dec 28 12:51:55 caudi_apx1 kernel: Reg 5: 0x Dec 28 12:51:55 caudi_apx1 kernel: Reg 6: 0xc01a0164 Dec 28 12:51:55 caudi_apx1 kernel: Reg 7: 0x1000 Dec 28 12:51:55 caudi_apx1 kernel: Reg 8: 0x Dec 28 12:51:55 caudi_apx1 kernel: Reg 9: 0x00ff Dec 28 12:51:55 caudi_apx1 kernel: Reg 10: 0x Dec 28 12:51:55 caudi_apx1 kernel: TE2XXP: Launching card: 0 Dec 28 12:51:55 caudi_apx1 kernel: TE2XXP: Setting up global serial parameters Dec 28 12:51:56 caudi_apx1 kernel: Found a Wildcard: Wildcard TE210P Dec 28 12:51:56 caudi_apx1 kernel: About to enter spanconfig! Dec 28 12:51:56 caudi_apx1 kernel: Done with spanconfig! Dec 28 12:51:56 caudi_apx1 kernel: Registered tone zone 4 (United Kingdom) Dec 28 12:51:56 caudi_apx1 kernel: About to enter startup! Dec 28 12:51:56 caudi_apx1 kernel: TE2XXP: Span 1 configured for ESF/B8ZS Dec 28 12:51:56 caudi_apx1 kernel: wct2xxp: Setting yellow alarm on span 1 Dec 28 12:51:56 caudi_apx1 kernel: SPAN 1: Primary Sync Source Dec 28 12:51:56 caudi_apx1 kernel: 2G: Got interrupt, status = 000f, GIS = 0081 Dec 28 12:51:56 caudi_apx1 kernel: 2G: Got interrupt, status = 000a, GIS = 0080 Dec 28 12:51:56 caudi_apx1 kernel: 2G: Got interrupt, status = 000a, GIS = 0080 Dec 28 12:51:56 caudi_apx1 kernel: VPM: Not Present Dec 28 12:51:56 caudi_apx1 kernel: Completed startup! Dec 28 12:51:56 caudi_apx1 kernel: 2G: Got interrupt, status = 000a, GIS = 0080 Dec 28 12:51:56 caudi_apx1 last message repeated 15 times etc/zaptel.conf # Define the E210P span=1,1,0,ccs,hdb3,crc4 bchan=1-8 dchan=16 unused=9-15,17-31 ztcfg: [EMAIL PROTECTED] ~]# ztcfg -vvv -d 2 Line 154: loadzone=uk Line 162: defaultzone=uk Line 224: span=1,1,0,ccs,hdb3,crc4 Line 225: bchan=1-8 Line 226: dchan=16 Line 227: unused=9-15,17-31 End of File Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: Clear channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 16: D-channel (Default) (Slaves: 16) 9 channels configured. Thanks in advance. Alex PS. If this is in HTML I applogise am having to use the webmail interface of Exchange and cant see if I can change the email type. Information contained in this e-mail and any attachments are intended for the use of the addressee only, and may contain confidential information of Ubiquity Software Corporation. All unauthorized use, disclosure or distribution is strictly prohibited. If you are not the addressee, please notify the sender immediately and destroy all copies of this email. Unless otherwise expressly agreed in writing signed by an officer of Ubiquity Software Corporation, nothing in this communication shall be deemed to be legally binding. Thank you. ___ --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] What setup
John - You might consider getting a T1 and splitting it using some of the pipe for your voice traffic and some for your data traffic. You can set up a VLAN on your internal network for your phones if you want to migrate to SIP phones and Asterisk, or you could implement a channel bank out to your existing analog handsets. Cory Andrews Purchasing Manager ++ VOIPSupply.com A Division of b2 Technologies 454 Sonwil Drive Buffalo, NY 14225 direct - 716.250.3402 mobile - 716.907.4054 email - [EMAIL PROTECTED] AIM - b2Cory - Original Message - From: John Crew [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, December 28, 2005 11:46 AM Subject: [Asterisk-Users] What setup I am an Asterisk newbie and don't have any telecom experience. I do know some about Linux and Windows as a sysadmin of Windows servers. I need to know what hardware to buy to replace a broken PBX. I have currently: -CBeyond as my carrier -16 port Cisco router with analog termination (not sure on terminology) into the building -Broken PBX: analog ATT Merlin system (not sure on model #, but could get it) -Second PBX: analog ATT Merlin system -8 extensions (~14 if you include the second ATT Merlin system serving our other business) -2 DIDs (~4 if you include our second business) -14 analog phones So, what are my options? I am looking for the cheapest/best solution. I could switch to a digital PRI or CAS line from my telco as another option, but I assume I would need to switch both PBXs and all phones to digital as well in that case. I need auto-attendant and music on hold, especially. Are these easy to set up? I installed Asterisk @ Home and it is running, but I need to RTFM to configure it. I could lose 1 DID for the faxing if we did fax-to-email. Please let me know. Thanks in advance! Sent by Go2net 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 ___ --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] CALLERIDNUM
in 1.2 and on (or CVS HEAD) you have to use: Set(CALLERID(num)=value)
On 12/28/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
is it possible rewrite CALLERIDNUM in the ZAP channel? I use
[int-transfer]
exten = _00.,1,SetVar(CALLERIDNUM=${CALLNR})
exten = _00.,2,MYSQL(Connect connid localhost webcdr ser91623 cdr)
exten = _00.,3,MYSQL(Query resultid ${connid} select\
if((floor(u.credit/p.cost))1\,ceil((u.credit)/p.cost)*60\,0)\ as\
sekund\ from\ user\ u\,\ sip\ s\,\ pricelist\ p\ where\
u.iduser=s.iduser\ and\ s.idsip=\'${CALLERIDNUM}\'\ and\
p.acode=s.acode\ and\ u.currency=p.currency\ and\
right(left(\'${EXTEN}\'\,CHAR_LENGTH(p.ccode)+2)\,CHAR_LENGTH(p.ccode))\
like\ concat(p.ccode\,\'%\')\ order\ by\ p.ccode\ desc\ limit\ 1)
exten = _00.,4,MYSQL(Fetch foundRow ${resultid} sekund) ; fetch row
..
..
without success. At row 3 have var ${CALLERIDNUM} original value,
not value from ${CALLNR}.
--
[EMAIL PROTECTED]
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RE: [Asterisk-Users] PRI: This number has been disconnected
I believe this behavior has nothing to do with the [EMAIL PROTECTED] Scripts. I think the problem is in the PRI signalization. I can see the zap hangup messages when trying to call a disconnected number. . -- Executing Dial(SIP/9349-1787, ZAP/g0/2514990) in new stack -- Called g0/2514990 -- Channel 0/2, span 1 got hangup -- Hungup 'Zap/2-1' == No one is available to answer at this time -- Executing Goto(SIP/9349-1787, s-NOANSWER|1) in new stack -- Goto (macro-dialout-trunk,s-NOANSWER,1) The telco says they are sending inband information with the status of the call, but Asterisk is hanging up the channel instead of connecting it to let hear the audio message. There is a post with a similar issue here: http://mailgate.supereva.com/comp/comp.dcom.isdn.capi/msg04138.html Is anyone experiencing the same behavior? -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Francesco Peeters (Asterisk) Enviado el: Martes, 27 de Diciembre de 2005 20:09 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] PRI: This number has been disconnected On Tue, December 27, 2005 23:37, Javier Ergas said: Hi, I'm running [EMAIL PROTECTED] 1.5 with TE110P E1 PRI in Chile. When calling an invalid number using, I expect to hear: We're sorry you have reached a number which has been disconnected ... And that is indeed what I hear when I dial out from [*] using analog FXO, or VoicePulse or NuPhone. When I dial that same number trough the T1 / PRI interface however, I only hear the allison7/all-circuits-busy-now message. There was another issue like this in an old post (http://lists.digium.com/pipermail/asterisk-users/2004-April/043597.html) but I think it isn't the same. SNIP I believe this has to do with the AMP macro's being used in [EMAIL PROTECTED] I am seeing similar things. For instance: One issue I have is that when a route has multiple trunks, and the first trunk after a while returns with 'NOANSWER', it merrily continues to the next trunk, which is not quite the behavior I'd expect. Especially as the primary trunk (IAX/VoipBuster) is *much* cheaper (ie free) as compared to the second trunk (Zap/g1), but the switch is made without any message. This could mean that you might be talking to someone on a different trunk, and instead of a free call, be paying normal fees. This could become expensive if you're calling the USA from Europe!... I am currently looking in to ways to enhance those macro's to respond more reliably, as well as return more useful information (busy tone on busy and no-answer, number disconnected info, etc.) when needed. If I do get to a satifactory set of macro's, I will put them up on the Wiki and let the list know... (I'm just starting on doing manual configuring, so it will be a tough job to crack, but also a learning experience...) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --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
Tr: Re: [Asterisk-Users] call test
Remarque : message transféré en pièce jointe. ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com---BeginMessage--- Harry, Here's what I am, getting -- Executing Dial(SIP/JacquesDesk-f2d7, Sip/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] Dec 28 11:26:09 NOTICE[25361]: chan_sip.c:9514 handle_response_invite: Failed to authenticate on INVITE to 'Jacques Desk sip:[EMAIL PROTECTED];tag=as5137aeb6' -- SIP/nxs.yi.org-851e is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) Best Jacques [EMAIL PROTECTED] wrote: Hello, I need to test my configuration please to dial sip:[EMAIL PROTECTED] . Your call will be sent to a queue . Regards Harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.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 ---End Message--- ___ --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] Most Stable Version of Asterisk
Anyone know what version of Asterisk is the most stable running Real-time queues and agents ? I am setting up a 200 phone call center and the first test run caused the system to crash 3 time in 3 days with only about 100 calls an hour. I used the same build that I have used in prior stable installations the only difference is I was not running real-time. Any help would be appreciated. Thanks John Bittner Simlab.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] CALLERIDNUM
I use 1.0.9 and 1.0.10 in 1.2 and on (or CVS HEAD) you have to use: Set(CALLERID(num)=value) -- [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
Re: [Asterisk-Users] What setup
Sending the message 3 times within 7 minutes wont get you responses any faster than just sending it once. Welcome to asterisk. Put your seat belts on and get ready for a few weeks of reading, testing, and caffeine. Here are my recommendations (assuming you want to set it up alone, otherwise just get the Business edition from digium.com): 1. Read, read, and read again 2. Test, test, and test again 3. Don't give up. 4. When everything fails, start at 1 again (or try this list :) 5. Get a T1 card anyhow for your system, even if you will stay with POTS, use a channel bank, or if you want you can use SIP gateways. 6. Try not using [EMAIL PROTECTED] or the like, it will just make sure that you never know asterisk (I know some people here are going to be all over me for this one), well sort of. In any case this only applies if you want to *know* asterisk. 7. A few days/weeks/months down the road when you have made it far enough to be able to jump into an argument of [EMAIL PROTECTED] vs asterisk from source, come back here and help other people :), and setup your system for production. Make sure: A. Your dialplan is clean (no duplicate extensions, no overlapping extensions, secure contexts, and easy to add, delete extensions, easy to modify extensions in just one place, like a macro) B. Your Linux distro is one that you know well enough to handle if anything outside asterisk goes wrong. C. Your Linux distro is one that *you* trust for being up as long as there is a power failure. Here are some URLs to get you started: http://www.asterisk.org/ ; well the asteirsk site http://www.voip-info.org/ ;the wiki http://www.asteriskdocs.org/ ;the asterisk docs project http://www.digium.com/ ;digiums site http://lists.digium.com/ ;the list archive http://bugs.digium.com/ ;the bug tracker for asterisk, I find this very helpful to see what an app is suppose to do to get it working before the docs are out, or to write up the wiki for an app :) in addition to the above you can search the lists using google, by entering site:lists.digium.com as part of your search term. I hope this helps On 12/28/05, John Crew [EMAIL PROTECTED] wrote: I am an Asterisk newbie and don't have any telecom experience. I do know some about Linux and Windows as a sysadmin of Windows servers. I need to know what hardware to buy to replace a broken PBX. I have currently: -CBeyond as my carrier -16 port Cisco router with analog termination (not sure on terminology) into the building -Broken PBX: analog ATT Merlin system (not sure on model #, but could get it) -Second PBX: analog ATT Merlin system -8 extensions (~14 if you include the second ATT Merlin system serving our other business) -2 DIDs (~4 if you include our second business) -14 analog phones So, what are my options? I am looking for the cheapest/best solution. I could switch to a digital PRI or CAS line from my telco as another option, but I assume I would need to switch both PBXs and all phones to digital as well in that case. I need auto-attendant and music on hold, especially. Are these easy to set up? I installed Asterisk @ Home and it is running, but I need to RTFM to configure it. I could lose 1 DID for the faxing if we did fax-to-email. Please let me know. Thanks in advance! Sent by Go2net 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 ___ --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] CALLERIDNUM
In 1.0.x the command is SetCIDNum http://www.voip-info.org/wiki-asterisk+cmd+setcidnum On 12/28/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I use 1.0.9 and 1.0.10 in 1.2 and on (or CVS HEAD) you have to use: Set(CALLERID(num)=value) -- [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] call transfer
I got this to work by editing the line
exten = s,14,Dial(${OUT_${ARG1}}/${OUTNUM})
to say
exten = s,14,Dial(${OUT_${ARG1}}/${OUTNUM},,Tt)
in extensions.conf
Do you know of anyway to set it up through AMP, so it works with all calls?
Michael Sampson
Information Systems Manager
Customer Contact Services
[EMAIL PROTECTED]
952-936-4000
Michael Sampson wrote:
I'm not sure how this is suppose to work. But I want to be able to
call people from a SIP phone and transfer them into a conference room.
If I call another extension that is a SIP phone I can hit # and then
enter the conference room number. If I call from the PSTN to the SIP
extension phone I can transfer by hitting # too. But if I call from
the SIP phone extension to a PSTN number it doesn't do anything when I
hit the #. I'm using [EMAIL PROTECTED] and under general settings I have
tTrwW for Asterisk Dial Command Settings.
Can you call through a Zap trunk from a SIP phone and do a call transfer?
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Re: [Asterisk-Users] CallerID info needed
Look at this post:
http://lists.digium.com/pipermail/asterisk-users/2005-December/139952.html
On 12/28/05, Doug Lytle [EMAIL PROTECTED] wrote:
Hey everybody,
I'm trying to figure out a problem with Caller-ID info coming in from
one of our facilities. The Caller-ID name is all that comes across. I
figured out that I probably could do a database lookup against the name
and set the Caller-ID number to their extension. I'm using Asterisk
SVN-trunk-r7230 on a PRI connected to a Definity PBX.
When testing, my Polycom IP501 still shows unknown. A bit from the log
below:
-- Accepting call from '' to '4288' on channel 0/2, span 1
-- Executing DBget(Zap/2-1, CIDINFO=name/Ballard, Lance) in new
stack
-- DBget: varname=CIDINFO, family=name, key=Ballard, Lance
-- DBget: set variable CIDINFO to 4300
-- Executing NoOp(Zap/2-1, Setting CallerID Number to: 4300) in
new stack
-- Executing Set(Zap/2-1, CALLERID(Name)=Ballard, Lance) in new
stack
-- Executing Set(Zap/2-1, CALLERID(Number)=4300) in new stack
-- Executing SetGroup(Zap/2-1, Max_Calls) in new stack
-- Executing NoOp(Zap/2-1, Active Calls: 1) in new stack
-- Executing GotoIf(Zap/2-1, 0?103) in new stack
-- Executing Dial(Zap/2-1,
IAX2/bc.asterisk:[EMAIL PROTECTED]/4288||t) in new stack
-- Called bc.asterisk:[EMAIL PROTECTED]/4288
-- Call accepted by 192.168.102.15 (format gsm)
-- Format for call is gsm
-- IAX2/liv.asterisk-2 is ringing
And my code snip:
exten = _42XX,1,Dbget(CIDINFO=name/${CALLERIDNAME})
exten = _42XX,2,NoOp(Setting CallerID Number to: ${CIDINFO})
exten = _42XX,3,Set(CALLERID(Name)=${CALLERIDNAME})
exten = _42XX,4,Set(CALLERID(Number)=${CALLERIDNUM}${CIDINFO})
; Both variables with never have data at the same
time
exten = _42XX,5,SetGroup(Max_Calls)
exten = _42XX,6,NoOP(Active Calls: ${GROUP_COUNT(Max_Calls)})
exten = _42XX,7,GotoIf($[ ${GROUP_COUNT(Max_Calls)} 4 ]?103)
exten =
_42XX,8,Dial(IAX2/bc.asterisk:[EMAIL PROTECTED]/${EXTEN},,t)
exten = _42XX,103,Congestion()
Any suggestions on a fix (if it's possible)?
Thanks!
Doug
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RE: [Asterisk-Users] What setup
John, I just switched from an old Merlin system myself! (haven't looked back). I was in the same situation as you (Windows server sysadmin with minimal *nix experience). Here's the setup I have currently in production: -Custom built server: Gigabyte motherboard with AMD Sempron 2800+ (I think), 1GB ram, 80GB IDE hard drive. Built for around $400. -Digium TDM04B in the server--supports 4 analog POTS lines (regular analog phone lines). If you have more lines, I think the 'preferred method' is to get a T1 or channel bank, then put a Digium T1 card in the server (instead of having a bunch of TDM04B's to connect on the analog lines). -19 Grandstream GXP-2000's (cheap and work well) -2 Polycom 601's (for the executives who think they need fancier _looking_ phones) -2 UTStarCom F1000's (uses WiFi; for myself and the custodian) When we changed phone systems, I made the decision to scrap all the analog phones and just replace them with the GXP-2000's. For ease of administration, I wanted to 'do it right' and not have a mish-mosh of analog and SIP phones. In our situation, it wasn't that much more money (the GXP-2000's are pretty inexpensive) and ensures everyone has the features they need. I have to confessI have all my stuff running on [EMAIL PROTECTED] I know I'll get flamed for saying this, because I know I should probably use Asterisk a-la-carte, but [EMAIL PROTECTED] is so easy I just couldn't resist!! I setup a system at my house first using this tutorial: http://mundy.org/blog/index.php?p=81 Then Googled and customized from there. Are you looking to get rid of both Merlin systems? I think after you get Asterisk setup at the one location, everyone else on the other Merlin system will get jealous :P The telecommuters love the remote capabilities of VOIP (using a softphone or the like) to work from home just like they're at work. Music on hold and auto-attendants are super easy to setup using [EMAIL PROTECTED] and/or AMP. [EMAIL PROTECTED] comes with music on hold by default. Browse through that tutorial above; it has all the commonly used features spelled out in 1,2,3 steps. I haven't been at this for too terribly long and I'm by no means an expert or anything, but I'd be happy to answer any questions I can. -ross -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Crew Sent: Wednesday, December 28, 2005 10:46 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] What setup I am an Asterisk newbie and don't have any telecom experience. I do know some about Linux and Windows as a sysadmin of Windows servers. I need to know what hardware to buy to replace a broken PBX. I have currently: -CBeyond as my carrier -16 port Cisco router with analog termination (not sure on terminology) into the building -Broken PBX: analog ATT Merlin system (not sure on model #, but could get it) -Second PBX: analog ATT Merlin system -8 extensions (~14 if you include the second ATT Merlin system serving our other business) -2 DIDs (~4 if you include our second business) -14 analog phones So, what are my options? I am looking for the cheapest/best solution. I could switch to a digital PRI or CAS line from my telco as another option, but I assume I would need to switch both PBXs and all phones to digital as well in that case. I need auto-attendant and music on hold, especially. Are these easy to set up? I installed Asterisk @ Home and it is running, but I need to RTFM to configure it. I could lose 1 DID for the faxing if we did fax-to-email. Please let me know. Thanks in advance! Sent by Go2net 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 ___ --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] PRI outgoing caller ID stopped working
Will this help: http://lists.digium.com/pipermail/asterisk-users/2005-December/140074.html On 12/27/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Saturday 24 December 2005 16:40, Kevin P. Fleming wrote: Interestingly, some systems I manage also began exhibiting this behavior in the past ten days or so. I have been working with the telco and they too show the Calling Number being received as expected over the PRI, but yet the far end receives 'Unknown' or 'Out of Area' depending on their CLID display device. I will continue to try to debug it, but I can't back down the code on that box to an older version for comparison of the PRI traffic; if you can do so, that would be most helpful. rev 5552 of asterisk, rev 208 of libpri, rev 877 (current) of zaptel... I still have this problem. Now this code is significantly older (with the exception of zaptel) than what I was running about a month ago when it was known to work... No great news yet, but at least it's a datapoint. :-( -A. ___ --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
[Asterisk-Users] SIP to SIP calls
hello, Is this so difficult to call an ip phone towards another via sip ? Does ser and asterisk projects are dedicated to the telephony or mail servers ? Harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.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] CallerID info needed
C F wrote: Look at this post: http://lists.digium.com/pipermail/asterisk-users/2005-December/139952.html Actually, I don't think the caller ID number is being sent in my situation, I am wondering what I can't manually set it. Thanks for the reply! Doug ___ --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: 26. RE: Stay away from Grandstream! (Bjorn Asmul)
The Grandstream certainly has issues, but it seems most of the SIP phones do. I continue to have excellent results with the Aastra 9133i. The latest firmware (1.3) supports busy lamps with Asterisk 1.2.x. I think that dollar for dollar, it is a fine phone and works better than most. Again, YMMV but its a good phone for my office needs. 26. RE: Stay away from Grandstream! (Bjorn Asmul) -- Message: 26 Date: Wed, 28 Dec 2005 10:49:16 -0500 From: Bjorn Asmul [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Stay away from Grandstream! To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Having tried EVERY single product from Grandstream, I don't think it's fair to judge Grandstream the way people do. I'm very happy with Grandstream products. As long as you upgrade the firmware they work fine. In fact they sometimes handle NAT better than any other device that I've tried (including ALL Sipura products). Grandstream is also one of very few to support ILBC codec, and BLF-support for Asterisk. If someone has tried the same, please comment. Bjorn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nir Simionovich Sent: Wednesday, December 28, 2005 4:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Stay away from Grandstream! I agree, GrandStream does seem to become the poor man's VoIP solution - making the bar for other VoIP phones very low to pass. I believe that GrandStream have a very good chance to basically being bought by a bigger company, like what happened to Sipura. What would happen then would be that people would say: Oh, GrandStream, Very good - after all bought them. I found that sometimes the most surprising hardware comes from non-known companies, like PerfecTone or Micronet. I think the main thing is the try things out, and find out what is the best suited IPhone for you. Nir S Steve Underwood wrote: I think the unfairness stems from Grandstreams generally being people's first IP phone - it seems like a cheap entry point to try things out. They then falsely assume everything else has to be better, especially if it has a higher price tag. Wrong. The standard for VoIP phones is total crap. Anything rising even slightly above that level wins awards for excellence. :-) Steve Nir Simionovich wrote: Hmmm... I feel that this is a little unfair towards GrandStream and other like vendors. Any vendor on the market has issues with their firmware, I can list many: Sipura/LinkSys SPA 841 (Latest firmware): 1. Phone doesn't re-register upon network loss 2. Phone firware becomes stalled, without any indication of an error while all functions continue working 3. Transfer function doesn't work as it should 4. MWI doesn't always work correctly 5. I can really go on and on... WellTech (Latest firmware): 1. Support for g729 is buggy 2. Echo cancel is buggy and causes ATA to crash 3. IP phones have no ability to re-configure the function keys on the box 4. Transfer/Conference buttons don't do anytning I can go on and on with other vendors, including Cisco, Nortel and more. The thing I'm saying is that any phone you'd test would run into issues at some time or other - claiming to stay away from one or another causes you to not even consider alternatives, thus at the end, you reach the Microsoft way of thinking. Last week, I got a phone to test with called a MicroNet. Actually, I got 3 phones, all from Micronet. I started them up, found out that 2 of them were actually WellTech phones (well, the shape told me, I hoped the firmware will be different, but I found out wrong). The third phone was different. It's called a Micronet SP5106 which to my surprise, worked almost flawlessly out of the box. It took me a while to configure the network correctly, and to understand the logic of the menu, but after that, the rest was easy. Transfer, 3-Way conference, Forward, DND, VoiceMail button, everything worked. What didn't work was configurable from the web backend - in other words: I couldn't find a flaw (yet). The only flaw I did find was this: the phone has the ability to connect to 3 SIP accounts at the same time. Upon defining a new account, you need to physically RESET the phone, other than that, the phone works just fine. I'll be posting a full review on my blog at http://www.net-gurus.net Regards, Nir S Vahan Yerkanian wrote: Stay away from Grandstream and AddPac. These are some of the companies with undereducated software developers that have problems with understanding written english, mainly the SIP RFC documents. I learned this the hard way, wasting half a year with helping them fix problems which shouldn't be there if they have had read/implemented the RFC correctly. Basically, they sell beta quality hardware and then you co-share their
Re: [Asterisk-Users] Re: Voicemail as other format?
is there an mp3 format for voicemail? what's the difference between wav49 and wav?On 12/28/05, Tomislav Parcina [EMAIL PROTECTED] wrote:In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Some users with Blackberry's cant play .wav files, is there a way to save the voicemail to save as another format like mp3? -KerryIn voicemail.conf edit this line. [general]format=wav49|gsm|wavP.S.Please stop replaying to mesage. If you plan to open thread do so bywriting mail to this addressasterisk-users@lists.digium.com --Tomislav Parcina[EMAIL PROTECTED]___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards,Mark Quitoriano, CCNAFan the flame...http://www.spreadfirefox.com/?q=user/registerr=19441 ___ --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] GSM-gateway setup
Merry Christmas List, Any body with experience on the GSM-gatewas that Cyber-telecom.net sell? The thing keeps on asking for a PASS and ... pretty much that's all. Help anyone? benchev ___ --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] subscription
hello ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.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] Sipura 2002 codec preferences
Rich Adamson wrote: I am about to sent some Sipura 2002 ATAs out to a call center. I want to use the dual line capability of the units, but I realize that the second channel will not be able to use G729 simultaneously. What do you think would the best option be for that channel? You might double check that. I thought someone commented the 2002 was a hardware upgrade that did support both ports running g729 now. I don't have one to test though. I think that might have been me, and I need to correct myself. The SPA-2100 supports two simultaneous g729 sessions, while the SPA-2002 only supports one. The chart at: http://www.sipura.com/products/spa2002.htm Kind of clears it up... -- Kristian Kielhofner ___ --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: who is online
On Wed, December 28, 2005 16:38, bails said: qualify=yes in both sip.conf and iax.conf, seems to highlight both the users and trunks who are currently available in FOP Bails Note that some IAX clients do not seem to like qualify=yes. I use DIAX, and when I use Qualify=yes, it becomes unavailable after a while... Also see http://www.voip-info.org/wiki-Asterisk+config+iax.conf and scroll halfway down to the qualify header -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --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] Polycom Asterisk 1.2.1 and Sipura SPA3000 strange problem
I have the follwoing setup: Asterisk SVN-tag-1.2.1-r7367 6 Polycom 500 Sip version 1.5.x 4 Sipura SPA3000 (not sure what build) (FXO port) All on flat single network, no NAT, and no gateways to reach each other. Sometimes (happens around 3 times a day, but sometimes far more often), while on the phone to an outside caller (on the PSTN using the FXO on the spa3k), the call dissconects from the polycom and goes thru the incoming extension for the sipura. In other words, astrisk at least as far as I can see from what gets executed in the DP (and maybe spa3k) sees this as if the follwoing has happened: 1. The polycom user hungup, 2. A new call came in on the spa3k. The follwoing is part of the log that I think might help: Dec 28 01:28:24 DEBUG[3368] channel.c: Didn't get a frame from channel: SIP/201-8ba1 Dec 28 01:28:24 DEBUG[3368] channel.c: Bridge stops bridging channels SIP/201-8ba1 and SIP/804-fd83 SIP/201 is the Polycom, while SIP/804 is the spa3k. If I'm losing a frame, is there a way to configure asterisk not to drop the channel? Or is this something the Polycom/Sipura are doing? FYI, asterisk is running on a VIA/MPIA platform. Thank You ___ --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] Iterfacing with a Mitel PBX
I am testing [EMAIL PROTECTED] V2.2 I want to interface with our PBX via a FXO card (TDM400P). I have one extension hooked up right now, and I can call into the Asterisk system from both a PBX connected phone, or through a DID number, but I can't dial from an IP phone out to our PBX system or out through a PSTN line (9 on the extension in the PBX gets an outside line). I can call other extensions that are set up within Asterisk. I have configured the Zap trunk, and set up an outbound route, but the best results I have gotten so far is 'connected' to dead air, or a fast busy tone. Are there good instructions posted somewhere for using a PBX extension? Thanks, Tom C ___ --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] how to alter cdr dst info?
Here is a reference cdr: 29. 2005-12-28 13:02:01 Zap/1-1... 8103970196 81039701968103970196 5100 ANSWERED 78 30. 2005-12-28 12:59:54 Zap/1-1... 8104590192 81045901928104590192 5128 ANSWERED 23 31. 2005-12-28 12:56:01 Zap/1-1... 8102320369 8102320369 5162 ANSWERED 165 32. 2005-12-28 12:54:19 Zap/2-1... 2489220303 2489220303 5162 ANSWERED 40 33. 2005-12-28 12:54:16 Zap/71-1... 2480365108 Mika2480365108 15860360822 NO ANSWER 25 34. 2005-12-28 12:51:11 Zap/71-1... 2480365123 Gary2480365123 12402240667 NO ANSWER 24 I am using a PBX behind our asterisk server, so calls coming from the PBX may be extensions or forwarded calls from outside. To add consistency, I have set all of the callerID info for asterisk extensions to be 10 digit numbers. Outbound calls from asterisk start with a 9, but there is no 9 from the PBX, so I have 2 sets of outbound rules. (1 for each context) What I would like to do: When DID calls come in, I would like to record a 10 digit number in the "dst" cdr field. When outbound calls are made, I would like to store them as 10 digits as well. either stripping off the 1 from PBX calls or the 91 from asterisk calls. I would also have to take International calls into account. removing the 9 from the 011X. from the PBXor just leaving the 011X. in tactfrom asterisk calls This alteration would have to after the call is made, to make sure that we are still dialing the number correctly. I will figure out the rules if someone can just point me in the right direction for altering the "dst" field in the cdr, or at least tell me that it can not be done. If not, I will have to see if I can do it via SQL after the fact. please advise. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past.--- - --- - - - - - - - -- - - - --- - -- - - --- - - -- - - - -- - - - ___ --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] Asterisk in Spanish
I was looking at the asterisk in spanish webpage. The register form is giving timeouts. On 9/19/05, Sergio Serrano [EMAIL PROTECTED] wrote: Try in www.asterisk-es.org -Mensaje original- De: Sebastian Milioto [mailto:[EMAIL PROTECTED] Enviado el: lunes, 19 de septiembre de 2005 15:08 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [Asterisk-Users] Asterisk in Spanish Hi all, I've been installing [EMAIL PROTECTED] and (of course) all the answering machine (I don't sure that's the right word in english, preatendedora in spanish) speech is in enlgish languaje. Is there anyway to download all those .gsm files speaked in spanish? Or may be another site which contain this kind of stuff (.wav, .gsm files for answering machines in spanish)? Thank you very much, Regards, Sebastian Milioto Telecommunications Engineer IM: [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] Mobile: 549 3571 543658 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.1/104 - Release Date: 16/09/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --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] Polycom Asterisk 1.2.1 and Sipura SPA3000 strange problem
I have the follwoing setup: Asterisk SVN-tag-1.2.1-r7367 6 Polycom 500 Sip version 1.5.x 4 Sipura SPA3000 (not sure what build) (FXO port) All on flat single network, no NAT, and no gateways to reach each other. Sometimes (happens around 3 times a day, but sometimes far more often), while on the phone to an outside caller (on the PSTN using the FXO on the spa3k), the call dissconects from the polycom and goes thru the incoming extension for the sipura. In other words, astrisk at least as far as I can see from what gets executed in the DP (and maybe spa3k) sees this as if the follwoing has happened: 1. The polycom user hungup, 2. A new call came in on the spa3k. The follwoing is part of the log that I think might help: Dec 28 01:28:24 DEBUG[3368] channel.c: Didn't get a frame from channel: SIP/201-8ba1 Dec 28 01:28:24 DEBUG[3368] channel.c: Bridge stops bridging channels SIP/201-8ba1 and SIP/804-fd83 SIP/201 is the Polycom, while SIP/804 is the spa3k. If I'm losing a frame, is there a way to configure asterisk not to drop the channel? Or is this something the Polycom/Sipura are doing? FYI, asterisk is running on a VIA/MPIA platform. Pure guess is that something happened (unknown what) and the error messages posted above are the result of that, and not the root cause. Finding the root cause may require you to implement the syslog server and debug server options in the spa3k, and compare those log entries to what * records for log messages during a failure. Implementing the log functions on the spa3k does require a reboot. Their log messages are rather cryptic, but looking at keywords and timestamps might identify which box(es) are involved with the dropped calls. ___ --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] Problems with multiple outbound calls going to PSTN - Wildcard TE405P
Hello everyone, Im having an outbound calling issue with our SIP phones. When one call is made to the PSTN another person trying to call receives a 404 error on the SIP phone. If we call the PSTN using SIP phone A and also calling from SIP phone B to SIP phone C everything works. The only problem were seeing is multiple calls going to the PSTN. Please let me know if anyone has any suggestions or recommendations. Here are the specifications of our server. 1 Digium Wildcard TE405P Asterisk 1.0.9 zapata.conf ; Zapata telephony interface ; ; Configuration file [trunkgroups] ; ; Trunk groups are used for NFAS or GR-303 connections. ; ; Group: Defines a trunk group. ; group = trunkgroup,dchannel[,backup1...] ; ; trunkgroup is the numerical trunk group to create ; dchannel is the zap channel which will have the ; d-channel for the trunk. ; backup1 is an optional list of backup d-channels. ; ;trunkgroup = 1,24,48 ; ; Spanmap: Associates a span with a trunk group ; spanmap = zapspan,trunkgroup[,logicalspan] ; ; zapspan is the zap span number to associate ; trunkgroup is the trunkgroup (specified above) for the mapping ; logicalspan is the logical span number within the trunk group to use. ; if unspecified, no logical span number is used. ; ;spanmap = 1,1,1 ;spanmap = 2,1,2 ;spanmap = 3,1,3 ;spanmap = 4,1,4 [channels] ; ; Default language ; ;language=en signalling = pri_cpe Switchtype=dms100 group=1 context=default channel = 1-23 ;busydetect=1 ;busycount=5 ;relaxdtmf=yes ;callwaiting=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callreturn=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes echocancelwhenbridged=yes group=1 callgroup=1 pickupgroup=1 immediate=no ; SOS context ;context=SOS ;usecallerid=yes ;group=1 ;callerid=SOS XX-5000 ;channel = 14-18 ; D-D context context=D-D usecallerid=yes group=1 callerid=D2D XX5010 ;CHANNELs may be associated with account codes 4 billing ;accountcode=DD5010 channel = 1-23 ___ --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] Polycom Asterisk 1.2.1 and Sipura SPA3000 strange problem
For somereason I think it's the polycom, which means I need logging for the Polycom and not the spa. On 12/28/05, Rich Adamson [EMAIL PROTECTED] wrote: I have the follwoing setup: Asterisk SVN-tag-1.2.1-r7367 6 Polycom 500 Sip version 1.5.x 4 Sipura SPA3000 (not sure what build) (FXO port) All on flat single network, no NAT, and no gateways to reach each other. Sometimes (happens around 3 times a day, but sometimes far more often), while on the phone to an outside caller (on the PSTN using the FXO on the spa3k), the call dissconects from the polycom and goes thru the incoming extension for the sipura. In other words, astrisk at least as far as I can see from what gets executed in the DP (and maybe spa3k) sees this as if the follwoing has happened: 1. The polycom user hungup, 2. A new call came in on the spa3k. The follwoing is part of the log that I think might help: Dec 28 01:28:24 DEBUG[3368] channel.c: Didn't get a frame from channel: SIP/201-8ba1 Dec 28 01:28:24 DEBUG[3368] channel.c: Bridge stops bridging channels SIP/201-8ba1 and SIP/804-fd83 SIP/201 is the Polycom, while SIP/804 is the spa3k. If I'm losing a frame, is there a way to configure asterisk not to drop the channel? Or is this something the Polycom/Sipura are doing? FYI, asterisk is running on a VIA/MPIA platform. Pure guess is that something happened (unknown what) and the error messages posted above are the result of that, and not the root cause. Finding the root cause may require you to implement the syslog server and debug server options in the spa3k, and compare those log entries to what * records for log messages during a failure. Implementing the log functions on the spa3k does require a reboot. Their log messages are rather cryptic, but looking at keywords and timestamps might identify which box(es) are involved with the dropped calls. ___ --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] Polycom Asterisk 1.2.1 and Sipura SPA3000 strange problem
In any case I'm trying to figure out if maybe someone else has seen this problem. Or if they know what it might be. On 12/28/05, C F [EMAIL PROTECTED] wrote: For somereason I think it's the polycom, which means I need logging for the Polycom and not the spa. On 12/28/05, Rich Adamson [EMAIL PROTECTED] wrote: I have the follwoing setup: Asterisk SVN-tag-1.2.1-r7367 6 Polycom 500 Sip version 1.5.x 4 Sipura SPA3000 (not sure what build) (FXO port) All on flat single network, no NAT, and no gateways to reach each other. Sometimes (happens around 3 times a day, but sometimes far more often), while on the phone to an outside caller (on the PSTN using the FXO on the spa3k), the call dissconects from the polycom and goes thru the incoming extension for the sipura. In other words, astrisk at least as far as I can see from what gets executed in the DP (and maybe spa3k) sees this as if the follwoing has happened: 1. The polycom user hungup, 2. A new call came in on the spa3k. The follwoing is part of the log that I think might help: Dec 28 01:28:24 DEBUG[3368] channel.c: Didn't get a frame from channel: SIP/201-8ba1 Dec 28 01:28:24 DEBUG[3368] channel.c: Bridge stops bridging channels SIP/201-8ba1 and SIP/804-fd83 SIP/201 is the Polycom, while SIP/804 is the spa3k. If I'm losing a frame, is there a way to configure asterisk not to drop the channel? Or is this something the Polycom/Sipura are doing? FYI, asterisk is running on a VIA/MPIA platform. Pure guess is that something happened (unknown what) and the error messages posted above are the result of that, and not the root cause. Finding the root cause may require you to implement the syslog server and debug server options in the spa3k, and compare those log entries to what * records for log messages during a failure. Implementing the log functions on the spa3k does require a reboot. Their log messages are rather cryptic, but looking at keywords and timestamps might identify which box(es) are involved with the dropped calls. ___ --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
