RE: [Asterisk-Users] The Smallest Asterisk Server Ever?
http://nlug.org/mail/nlug__2003_12/0094.html Kevin -Original Message- From: Panny Malialis [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 2:58 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] The Smallest Asterisk Server Ever? I cant wait to see the asterisk on an xbox page!!, but the link seems broken http://nlug.org/mail/nlugb2003_12/0094.html I've tried removing the b with no luck Anyone know what the link should be ? Thanks Panny - Original Message - From: David J Carter [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 8:31 PM Subject: RE: [Asterisk-Users] The Smallest Asterisk Server Ever? Hey I don't know, I paid ?100 ($170) for my XBox, couldn't get a PC for that. The Linux bit is all free, and only a couple of PCB work to disenable the protection. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Albertson Sent: 03 February 2004 18:01 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] The Smallest Asterisk Server Ever? I read a report of Asterisk running on a Microsoft X-Box. That's kind of a stunt as you could buy a decent PC for the price of a Linux-capable XBox. Id's still like to see Asterisk run on very low-end hardware The Snom IP phone runs Linux inside? I assume as Linux is GPL'd Snom will supply the source code? It would be fun to install an Asterisk server in a phone. --- Panny Malialis [EMAIL PROTECTED] wrote: Does anyone have it running on a Cyclades T100 ? same as used for ntop/nbox. I was thinking of using that as an IAX-sip translator for offices with NAT. CPU MPC855T (PowerPC Dual-CPU) Memory 32MB RAM / 4MB Flash (TS100) Interfaces1 Ethernet 10/100BT on RJ45 1 RS232 Console on RJ45 RS232 Serial Ports on RJ45 Looks like fun! Although a little lacking on memory. Any comments? Panny ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk / T-bird [THREAD NAME CHANGE]
Regarding using asterisk as a T-Bird replacement, that would be pretty cool, as asterisk could do some incredible monitoring and call generation tricks. But, bugs in the PRI driver software (especially beefed up error handling) would have to be fixed first, as asterisk doesn't handle high volume call setups very well. Framing errors etc Regards, Scott Stingel Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: scott at evtmedia.com URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Lopez Sent: Tuesday, February 03, 2004 1:18 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #2719 - 10 msgs How about a PCMCIA Zapata interface?? Asterisk and its strength kick ass as a test unit. Can't do some of the things a T-byrd can do but the Telco techs freak when you tell them its your PBX!!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [OT] Oldest Telephone
While the rest of you were chatting about the smallest * server, I was sitting her staring at the telephone hanging on the wall. It is a Western Electric set in a varnished pine box with an earpiece you hold in your left hand and a mouthpiece attached to the box. You crank the magneto with your right hand to signal the switchboard. The two dry cells inside are dated 1935, and I'm throwing them away because they're leaking acid. Wouldn't it be a kick to be able to ring this phone from an * server and have a conversation. There are several lugs inside: L1, GND, L2, COND (no condenser installed), E, and K. And of course the two battery lugs. Does anyone know anything at all about this set? I realize the CO supplies battery instead of the telephone, but can the phone be modified? Also, am I going to blow a FXS card if I don't get it right? Cheers, Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voip phones
What is the best inexpensive voip phone out there? I want to try a few with *, but don't want to go broke while I'm just playing around... Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 (888) 924-3728 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] The Smallest Asterisk Server Ever?
http://nlug.org/mail/nlug__2003_12/0094.html Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Panny Malialis [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 10:58 AM Subject: Re: [Asterisk-Users] The Smallest Asterisk Server Ever? I cant wait to see the asterisk on an xbox page!!, but the link seems broken http://nlug.org/mail/nlugb2003_12/0094.html I've tried removing the b with no luck Anyone know what the link should be ? Thanks Panny - Original Message - From: David J Carter [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 8:31 PM Subject: RE: [Asterisk-Users] The Smallest Asterisk Server Ever? Hey I don't know, I paid ?100 ($170) for my XBox, couldn't get a PC for that. The Linux bit is all free, and only a couple of PCB work to disenable the protection. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Albertson Sent: 03 February 2004 18:01 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] The Smallest Asterisk Server Ever? I read a report of Asterisk running on a Microsoft X-Box. That's kind of a stunt as you could buy a decent PC for the price of a Linux-capable XBox. Id's still like to see Asterisk run on very low-end hardware The Snom IP phone runs Linux inside? I assume as Linux is GPL'd Snom will supply the source code? It would be fun to install an Asterisk server in a phone. --- Panny Malialis [EMAIL PROTECTED] wrote: Does anyone have it running on a Cyclades T100 ? same as used for ntop/nbox. I was thinking of using that as an IAX-sip translator for offices with NAT. CPU MPC855T (PowerPC Dual-CPU) Memory 32MB RAM / 4MB Flash (TS100) Interfaces1 Ethernet 10/100BT on RJ45 1 RS232 Console on RJ45 RS232 Serial Ports on RJ45 Looks like fun! Although a little lacking on memory. Any comments? Panny ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sipphone dialing out problem
Hello
when i dial a toll free no using sipphone
i get this error message. How do i solve this?
Any help will be appreciated.
console message:
Starting simple switch on
'Zap/2-1' -- Executing SetCallerID("Zap/2-1",
"17473863282") in new stack -- Executing
SetCIDName("Zap/2-1", "Deepak JV") in new stack --
Executing Dial("Zap/2-1", "SIP/[EMAIL PROTECTED]")
in new stack -- Called [EMAIL PROTECTED]
-- SIP/proxy01.sipphone.com-8efc is making progress passing it to
Zap/2-1 -- SIP/proxy01.sipphone.com-8efc answered
Zap/2-1Feb 3 22:15:57 NOTICE[1218901440]: channel.c:1481
ast_set_read_format: Unable to find a path from G729A to ULAWFeb 3
22:15:57 NOTICE[1218901440]: channel.c:1451 ast_set_write_format: Unable to find
a path from ULAW to G729AFeb 3 22:15:57 WARNING[1218901440]:
chan_zap.c:3728 zt_write: Cannot handle frames in 256 format == Spawn
extension (internal, 18006526672, 3) exited non-zero on
'Zap/2-1' -- Hungup 'Zap/2-1'
Regards
Deepak
Re: [Asterisk-Users] [OT] Oldest Telephone
While the rest of you were chatting about the smallest * server, I was sitting her staring at the telephone hanging on the wall. It is a Western Electric set in a varnished pine box with an earpiece you hold in your left hand and a mouthpiece attached to the box. You crank the magneto with your right hand to signal the switchboard. That's really odd, I've got it's matching mate... the one where the earpiece is held in the right hand. ;) The two dry cells inside are dated 1935, and I'm throwing them away because they're leaking acid. Wouldn't it be a kick to be able to ring this phone from an * server and have a conversation. There are several lugs inside: L1, GND, L2, COND (no condenser installed), E, and K. And of course the two battery lugs. L1 and L2 are essentially Tip Ring, just like current stuff. Does anyone know anything at all about this set? I realize the CO supplies battery instead of the telephone, but can the phone be modified? Also, am I going to blow a FXS card if I don't get it right? If you do connect it and someone turns the crank on the phone, you'll be applying about 100 volts AC to the FXS card. No idea if your card would hold up to that. (Disconnect one of the wires on that magneto if not sure.) I've got one of the old Operator line/drop modules that use to be at the other end of that old telephone line. Neat conversation piece. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voip phones
Tim Sailer wrote: What is the best inexpensive voip phone out there? I want to try a few with *, but don't want to go broke while I'm just playing around... Tim AFAIK Grandstreams are still the cheapest.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] [OT] Oldest Telephone
Well, the nice things about telephones, in the US anyway, is that they are generally backward-compatible with each other. Why not, as a first step, connect a normal telco line to L1 and L2, and see if you get dial tone through the receiver? Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: [EMAIL PROTECTED] URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Welter Sent: Tuesday, February 03, 2004 1:57 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] [OT] Oldest Telephone While the rest of you were chatting about the smallest * server, I was sitting her staring at the telephone hanging on the wall. It is a Western Electric set in a varnished pine box with an earpiece you hold in your left hand and a mouthpiece attached to the box. You crank the magneto with your right hand to signal the switchboard. The two dry cells inside are dated 1935, and I'm throwing them away because they're leaking acid. Wouldn't it be a kick to be able to ring this phone from an * server and have a conversation. There are several lugs inside: L1, GND, L2, COND (no condenser installed), E, and K. And of course the two battery lugs. Does anyone know anything at all about this set? I realize the CO supplies battery instead of the telephone, but can the phone be modified? Also, am I going to blow a FXS card if I don't get it right? Cheers, Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voip phones
If you've got a Linux workstation, www.vovida.org offers sipset, a free softphone. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Sailer Sent: Tuesday, February 03, 2004 4:10 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] voip phones What is the best inexpensive voip phone out there? I want to try a few with *, but don't want to go broke while I'm just playing around... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] [OT] Oldest Telephone
ON second thought: I re-read your message - missed the part about the magneto. Maybe you shouldn't connect that phone to the PSTN after all! (or to a digium card) Magneto-generated ring detection is a bit beyond the digium card spec I'm sure! Cheers! Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: [EMAIL PROTECTED] URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Welter Sent: Tuesday, February 03, 2004 1:57 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] [OT] Oldest Telephone While the rest of you were chatting about the smallest * server, I was sitting her staring at the telephone hanging on the wall. It is a Western Electric set in a varnished pine box with an earpiece you hold in your left hand and a mouthpiece attached to the box. You crank the magneto with your right hand to signal the switchboard. The two dry cells inside are dated 1935, and I'm throwing them away because they're leaking acid. Wouldn't it be a kick to be able to ring this phone from an * server and have a conversation. There are several lugs inside: L1, GND, L2, COND (no condenser installed), E, and K. And of course the two battery lugs. Does anyone know anything at all about this set? I realize the CO supplies battery instead of the telephone, but can the phone be modified? Also, am I going to blow a FXS card if I don't get it right? Cheers, Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: RE: [Asterisk-Users] cdr mysql problem
I just followed those steps and its working wonderfully. I did have to download and compile asterisk-addons from CVS, as well as the steps illustrated above. Now I'm going to make some PHP Web Reports so the users can view the CDR. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomica Crnek Sent: Tuesday, February 03, 2004 2:54 PM To: [EMAIL PROTECTED] Subject: RE: RE: [Asterisk-Users] cdr mysql problem Thanks, I don't know what is different from all steps I have followed several times. I did all this before, believe me. Now, I said to myself that I'll do it once again, and it worked. Thanks once again! Tomica -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dipak0105 Sent: Tuesday, February 03, 2004 1:47 PM To: [EMAIL PROTECTED] Subject: Re: RE: [Asterisk-Users] cdr mysql problem Hi You have to follow this steps and obviously you got success because we have followed this steps and got success. Configure the cdr_mysql.conf file in the location /etc/asterisk. The configuration is as follows: [global] hostname=localhost ;This is the host name of MySql dbname=asteriskcdrdb;This is the database name of MySql password=password ;This is the password of user user=asteriskcdruser;This is the user in MySql port=3306 ;This is the port number running the MySql sock=/var/lib/mysql/mysql.sock ;This is the socket of MySql and the location Then follow the steps in the MySql server 1. Go to Start ApplicationSystemService Configuration. Check the mysqld box. Click the save button Then click the start button. 2. Open a kolsole. 3. Type: mysql -u root to enter into mysql. 4. Type: SET PASSWORD FOR root = PASSWORD(password); for setting the root password. 5. Type: GRANT ALL PRIVILEGES ON *.* TO [EMAIL PROTECTED] IDENTIFIED BY password WITH GRANT OPTION; 6. Type: \q; to Quit mysql. 7. Type: In konsole mysql -u asteriskcdruser -p and use the password password, to reenter mysql as asteriskcdruser user. 8. Type: CREATE DATABASE asteriskcdrdb; 9. Type: CREATE TABLE cdr ( calldate datetime NOT NULL default '-00-00 00:00:00', clid varchar(45) NOT NULL default '', src varchar(45) NOT NULL default '', dst varchar(45) NOT NULL default '', dcontext varchar(45) NOT NULL default '', channel varchar(45) NOT NULL default '', dstchannel varchar(45) NOT NULL default '', lastapp varchar(45) NOT NULL default '', lastdata varchar(45) NOT NULL default '', duration int(11) NOT NULL default '0', billsec int(11) NOT NULL default '0', disposition varchar(45) NOT NULL default '', amaflags int(11) NOT NULL default '0', accountcode varchar(45) NOT NULL default '', uniqueid varchar(45) NOT NULL default '' ); for create a table cdr. 10. To reload the configuration, type reload from the Asterisk command prompt. These are the steps of Configuration of MySql with Asterisk server. Contact me if you need any further clarifications. Dipak Kumar Paul. Sigmabit Technology India. [EMAIL PROTECTED] wrote: Hi, here it is... [EMAIL PROTECTED] asterisk]# cat cdr_mysql.conf ; ; Note - if the database server is hosted on the same machine as the ; asterisk server, you can achieve a local Unix socket connection by ; setting hostname=localhost ; ; port and sock are both optional parameters. If hostname is specified ; and is not localhost, then cdr_mysql will attempt to connect to the ; port specified or use the default port. If hostname is not specified ; or if hostname is localhost, then cdr_mysql will attempt to connect ; to the socket file specified by sock or otherwise use the default socket ; file. ; [global] hostname=localhost dbname=asteriskcdrdb password=** user=asteriskcdruser ;port=3306 ;sock=/tmp/mysql.sock sock=/var/lib/mysql/mysql.sock srwxrwxrwx1 mysqlmysql 0 Feb 2 19:37 /var/lib/mysql/mysql.sock Tomica -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Tuesday, February 03, 2004 12:32 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] cdr mysql problem On Monday 02 February 2004 15:27, Tomica Crnek wrote: Yes, I have checked the logs. There is nothing there. I think asterisk doesn't try to connect. Please paste the contents of /etc/asterisk/cdr_mysql.conf. Also, paste the output of: ls -l /tmp/mysql.sock /var/lib/mysql/mysql.sock ; locate mysql.sock -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
Re: [Asterisk-Users] voip phones
I prefer SIP Phones, Grandstream BT-100 IP-Phone or the HT-286 Analogue telephone adapter Why? - brilliant user interface, with or with out a web browser - cristal clear voice even with low band codecs - PPP over ethernet (PPPoE) aware - continual firmware improvement - plenty of tweak options - economically priced - protocol conform - made in china - fast shipping Retail from $39 to $245 .. google is your friend. Tim Sailer wrote: What is the best inexpensive voip phone out there? I want to try a few with *, but don't want to go broke while I'm just playing around... Tim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialling Hook Flash on Zaptel
On Tue, 3 Feb 2004 11:33:24 -0600, David Gomillion wrote
Steven Critchfield wrote:
On Tue, 2004-02-03 at 09:53, David Gomillion wrote:
Steven Critchfield wrote:
On Tue, 2004-02-03 at 06:59, Christopher Lee wrote:
[flames, non-flames etc. snipped]
what about something like this? NOTE: THIS IS NOT WORKING
CODE. It is an idea, a concept. If you want to try it to
make it work, then you will have to build on this.
exten = _*XX,1,Dial(Zap/1/1) (dials a 1 on the outgoing
zap interface, probably needs a short timeout)
exten = _*XX,2,Flash()
exten = _*XX,3,Dial(Zap/1/${EXTEN})
The flash is probably on the wrong side, as I look at it
more closely. This will probably send the flash to your
VoIP client. But maybe you could look into scripting with AGI.
I'm not sure it'll work as Dial doesn't return to the dialplan. It
seems like the only clean method to do this would be to extend
chan_zap to include flash as an extra digit (a'la W). Would a
copy-paste from app_flash do? I have no zaptel devices to begin with
so I can't really check.
Or maybe some voodoo with transfers, local channels etc. would work?
And with that, I bid the fair Asterisk-User's list a
farewell, at least for posting. I will now become one of
the countless other leaches who give nothing back to the
community. It was good to get help, and I tried to help
others out, but I have a lot better things to do than spend
my time helping others only to get flamed every time I turn around.
You need to remember that we're all volunteers. I will only
take it in the teeth so many times before I say goodbye.
Go ahead and rip me a new one. Have fun. Rant, rave, call
me stupid. Tell me I have no value, and that I contribute
nothing. The more you say it, the more accurate it becomes.
Hey, why take things personally? I think Steven is known for his
inflammatory posts. Sure it isn't nice to be flamed openly and IMHO
Steven is doing a fine job scaring people away from the list
(sometimes just *too* fine), but we (=the newbies, like me) all end up
learning something. In this way I find his posts doing more good than
harm. From a third-person perspective (not being flamed by him - yet?)
I'd describe his posts somewhere between informative and funny with an
occasional flamebait (yes, I do hang out on /.)
That being said, I'm mostly a lurker myself.
Regards and don't feel too offended :)
Greg
___
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Re: [Asterisk-Users] voip phones
and with the HT-286 you get a Chinaman in a box! :) - Original Message - From: Michael Koehler [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 5:45 PM Subject: Re: [Asterisk-Users] voip phones I prefer SIP Phones, Grandstream BT-100 IP-Phone or the HT-286 Analogue telephone adapter Why? - brilliant user interface, with or with out a web browser - cristal clear voice even with low band codecs - PPP over ethernet (PPPoE) aware - continual firmware improvement - plenty of tweak options - economically priced - protocol conform - made in china - fast shipping Retail from $39 to $245 .. google is your friend. Tim Sailer wrote: What is the best inexpensive voip phone out there? I want to try a few with *, but don't want to go broke while I'm just playing around... Tim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
I am a new asterisk user and I love what I see so far. I have a question about distinctive ring though. In my situation, we have 1 phone number for voice calls and one for faxes. They share the same line, and right now I use vgetty with mgetty+sendfax and VOCP to deal with calls and faxes. Vgetty detects the ring type and answers the phone with a hayes modem in fax mode and calls a script that I use to process the faxes. If it is a voice call then vgetty hands it over to vocp. I am replacing the vocp part of the process with asterisk, but I still need to be able to get faxes on the line. I need to have asterisk ignore the call if it has the fax ring and let vgetty deal with it. Of course, I also need vgetty to ignore the voice ring. I know how to do this in vgetty, but I am a little confused in asterisk. I found the stuff in zapata.conf and made the following changes: usedistinctiveringdetection=yes dring1=96,0,0 dring1context=inbound-analog dring2=336,95,0 dring2context=ignore-fax dring3=336,96,0 dring3context=ignore-fax dring4=338,98,0 dring4context=ignore-fax I have several ring types specified for the fax as it seemed that I got different ring patterns from time to time. Anyway, I put the inbound-analog and ignore-fax as contexts in my extensions.conf file. For the ignore-fax context I put the following hoping it would work, but it just answers the phone normally. [ignore-fax] exten = s,1,NoOp ; Ignore this one Any info/pokes in the right direction are greatly appreciated. Thanks, Cullen -- Cullen Simpson CompuCrew Inc. Tel: (407)349-2373 Cel: (407)721-8014 Fax: (407)349-2710 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #2711 - 15 msgs
you can do this with MeetMe, but you don't have to. you can also use Parking, which makes things a little simpler. in either case, the strategy is going to be something like: 1. Record the soundfile 2. Park the inbound caller 3. Use a .call file or the manager interface to initiate an outbound call to the mobile phone 4. play soundfile and prompt the mobile phone user to accept/reject the call 5a. if the call is accepted, transfer the parked call to the mobile user. 5b. if the call is not accepted, unpark the call to some extension and take whatever action you like. be warned, * will probably have no idea whether your outbound call is answered by a human, an answering machine, or hits a busy signal. be further warned, i'm fairly new to * too. however, i have successfully implemented the above. hopefully one of the more astute readers will jump in if it seems like i'm leading you too far astray. ;) hope this helps, tad --__--__-- Message: 7 Date: Tue, 3 Feb 2004 03:41:44 + From: Kris Edwards [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Playing announcement to called user prior to Confirmation Reply-To: [EMAIL PROTECTED] Hello all, As I'm sure is pretty common, I have some extensions that dial mobile numbers after a local timeout. I would like to prompt the caller to record their name after the local timeout and have the recipient be able to hear the name prior to accepting the call. Recording the message is easy enough, so I thought about doing something like dumping them into MeetMe after they record (change the empty conference room message to something more appropriate please wait while I try mobile.. blah blah.. even some nice music when they wait. Then, when the mobile is called I could dial some extension that plays the recorded name and decide whether or not I want to join the conf, but if I rejects the call, then the caller is stuck in the conference, right? I'm pretty new (in case it doesn't show) so if this has been covered I hope someone is kind enough to post a link (I've searched... nothing so far.) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Detecting answer supervison from an AGI app
I've got a dumb Western Electric payphone and some homebuilt hardware to control the coin relay which is accessible to Asterisk through the AGI interface. I'd like to be able to set the state of the coin relay to collect at the end of a call if a called party answers. [Hey, I admit this project is being persued just for the fun of it ] Looking through the documentation, there is a way to get channel state, but it only tells you that the channel is bridged and does not tell you the called party has answered. Is there another interface (unix sockets maybe?) which I can use to query the answer supervison state for a given channel? Steve. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dialling Hook Flash on Zaptel
Just for fun, try this: exten = 922,1,Flash(Zap/1) exten = 922,2,Dial(Zap/1/*022) exten = 922,3,Congestion exten = 922,4,Hangup and see if it gives the same error. I'd be interested to see if there's perhaps some strange variable swapping going on. I gave that a try, but the same problem on the console that this is not a Zap channel. Also tried David's Gomillion's approach with a dial then flash, but still the same problem about not being a zap channel. And indeed, flash doesn't accept an argument so there's not much point in me placing it there, I was just trying out a non-working example/food for thought that was posted previously... http://www.mail-archive.com/[EMAIL PROTECTED]/msg23426.html I've been toying with the extension a little more in the hope of perhaps doing a pseudo flash as per David's dial then flash suggestion but tweaked as follows:- exten = 922,1,Dial(Zap/1/*022,1) exten = 922,2,Dial(Zap/1/*022) exten = 922,3,Congestion exten = 922,4,Hangup (I've tried a few variations on the above, including a 1 in place of the *, and first line dialing 0 instead of the full command) Indeed, Asterisk picks up the line, dials (hopefully correctly) *022 then hangs up after 1000ms. Then hopefully hammers the line open again so the Norstar sees it as a flash and continues to dial *022 again. Unfortunately it doesn't work, Asterisk seems to be doing it correctly, but the Nortel can be a cantankerous beast with it's analog ports. I just get the rapid congestion tone from it that somethings not right with the way I dialled. As per the further suggestions, my speed dials rarely ever change, and I think I will relent and take this approach... basically I was wanting to not have to change the restriction filters on the Nortel for that analog port (since the speed dials override restrictions), but I think I'll fine grain the dialing restrictions through my Asterisk dial plans, should be the most pain free approach. The other thing was I just wanted to learn a little more about what can be done on the X100P, as there's many other commands that can be sent to the Norstar that are prefixed with a flash, although I doubt I'll really ever need to use any of them. Thanks very much for all the suggestions though, much appreciated. And David I hope you continue on the list, if only as a reader, as your input and contributions are definitely appreciated. Cheers, Chris Lee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] [OT] Oldest Telephone
On Tue, 3 Feb 2004, Scott Stingel wrote: Well, the nice things about telephones, in the US anyway, is that they are generally backward-compatible with each other. Why not, as a first step, connect a normal telco line to L1 and L2, and see if you get dial tone through the receiver? I'm sure that your LEC wouldn't minx 100 Volts AC getting sent to their switch ! ;) If you plan on doing that, might as well dust off the archives and check out this backwoods, inbred cousin of 802.3af: http://www.linuxsavvy.com/staff/jgotts/underground/boxes/blotto.html What is scary is that I was around when this originally circulated the BBS scene, circa 84/85. ;) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] [OT] Oldest Telephone
Yes, I hereby withdraw my suggestion (see my earlier, 2nd post)!! NOT advocating connecting a magneto to the telco circuit! and only to a digium card if you want to let off some steam (and probably fry the card) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Boehnlein Sent: Wednesday, February 04, 2004 12:51 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] [OT] Oldest Telephone On Tue, 3 Feb 2004, Scott Stingel wrote: Well, the nice things about telephones, in the US anyway, is that they are generally backward-compatible with each other. Why not, as a first step, connect a normal telco line to L1 and L2, and see if you get dial tone through the receiver? I'm sure that your LEC wouldn't minx 100 Volts AC getting sent to their switch ! ;) If you plan on doing that, might as well dust off the archives and check out this backwoods, inbred cousin of 802.3af: http://www.linuxsavvy.com/staff/jgotts/underground/boxes/blotto.html What is scary is that I was around when this originally circulated the BBS scene, circa 84/85. ;) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voip phones
not to mention, fortune cookies are included! :) Hey Chinaman... I was wondering if the following SIP phone is just a Grandstream's OEM or just a japanese copy... http://sipphone.livedoor.com/ What do you think? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Pictures of new multiport FXO/FXS from digum
Does the voicetronix card work with Asterisk? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, February 02, 2004 11:06 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Pictures of new multiport FXO/FXS from digum -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, 31 January 2004 8:56 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Pictures of new multiport FXO/FXS from digum Hello, I have heard that digium (mark) is going to introduce a new multiport FXO/FXS card next month. Does anyone have any details or some pictures of the card. The latest I know about the card is that it will have 16 ports and I wonder how these 16 ports will be connected to a PCI interface using RJ11? Any guess from the intelligent ones out there? Is there an external connector (from PCI card) going into a little box with bunch of RJ11? The voicetronix openline12 (12 fxo/fxs) does it vi RJ45 ports -i.e. 4 lines per RJ45 which connects to some sort of breakout box to split into 4 RJ11s. Using standard connectors means you can do things like run 1 standard cable from rhe server to the wall or another office rather than 4. Cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] [OT] Oldest Telephone
Well, the nice things about telephones, in the US anyway, is that they are generally backward-compatible with each other. Why not, as a first step, connect a normal telco line to L1 and L2, and see if you get dial tone through the receiver? I'm sure that your LEC wouldn't minx 100 Volts AC getting sent to their switch ! ;) If you plan on doing that, might as well dust off the archives and check out this backwoods, inbred cousin of 802.3af: http://www.linuxsavvy.com/staff/jgotts/underground/boxes/blotto.html What is scary is that I was around when this originally circulated the BBS scene, circa 84/85. ;) Oh yah!!! And I was around when our telephone number was 79M and the operator had to complete every call (and she was my Sunday School teacher too)! :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 8 lines - best approach
Will this product be available in the next few weeks? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Saturday, January 31, 2004 3:01 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 8 lines - best approach How about a 16 port FXO/FXS card (your choice of FXO/FXS modules or any mix of them) for $999. Will that be a good option? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Saturday, January 24, 2004 5:35 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 8 lines - best approach On Friday 23 January 2004 12:18, Paul Mahler wrote: On Friday, January 23, 2004 at 8:04 AM, Steven Critchfield wrote: On Fri, 2004-01-23 at 09:30, Darren Martz wrote: I have 8 lines coming into an existing PBX system and am looking for a cost effective way to replace the existing system with Asterisk. We need some of the features in Asterisk, including its ability to support remote offices (long distance savings). At first glance this appears to require a T100P card and a channel bank, but that seems rather expensive. My estimated price on that would be roughly $2600 for 8 lines given that system - perhaps my estimate is way off Is there another way that is more cost effective? That number sounds about right. It is likely that it will be less, but budgeting that much for hardware is a good start. Do you have to continue to use the existing handsets? You should look at replacing the existing phones with SIP phones. He did say cost-effective. Last I checked, 24 SIP phones (unless they are Grandstreams) will cost far more than a channel bank. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Pictures of new multiport FXO/FXS from digum
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gene Kochanowsky Sent: Wednesday, 4 February 2004 12:18 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Pictures of new multiport FXO/FXS from digum Does the voicetronix card work with Asterisk? There is a channel driver (chan_vpb) distributed with asterisk, I don't know how well it works, someone needs to do a report/review like Rich Adamson's one on the Mediatrix. The cards are about AU$2200 RRP I think. There is an openswitch 6 which is about half price and openline4 which is a bit different. Cheers, Woody -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, February 02, 2004 11:06 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Pictures of new multiport FXO/FXS from digum -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, 31 January 2004 8:56 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Pictures of new multiport FXO/FXS from digum Hello, I have heard that digium (mark) is going to introduce a new multiport FXO/FXS card next month. Does anyone have any details or some pictures of the card. The latest I know about the card is that it will have 16 ports and I wonder how these 16 ports will be connected to a PCI interface using RJ11? Any guess from the intelligent ones out there? Is there an external connector (from PCI card) going into a little box with bunch of RJ11? The voicetronix openline12 (12 fxo/fxs) does it vi RJ45 ports -i.e. 4 lines per RJ45 which connects to some sort of breakout box to split into 4 RJ11s. Using standard connectors means you can do things like run 1 standard cable from rhe server to the wall or another office rather than 4. Cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM400 FXO???
Thud! Will there be no FXO daughter boards for the TDM400? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gene Kochanowsky Sent: Thursday, January 29, 2004 7:55 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] TDM400 FXO??? Any word/news on the progress availability of the FXO daughter boards? Gene
[Asterisk-Users] IPKall-FWD-Asterisk
Hi Folks, I recently setup an asterisk system in order to provide a telephone phone system for my web hosting business at a very low expense. My problem is that DTMF tones are not being recognized when calling the IPKall phone number. Calling my server via FWD and IAXTel works out fine however. Has anybody experienced this with the IPKall service? are they not passing the DTMF tones or am I doing something wrong? I've tried switching dtmfmode to all the options, but still nothing. Thanks for your help! - Joshua Colp.
Re: [Asterisk-Users] Pictures of new multiport FXO/FXS from digum
Does the voicetronix card work with Asterisk? Yes, and no. It works in that you are able to use it to make and receive calls - but to say it works well would be an overstatement. We are currently using the OpenLine4 card and are having problems dialing (card dials too early ; doesn't support 'w' to delay dialing ; DTMF isn't recognised correctly from certain phones - namely Cisco 7960) , and also have quality problems when using more than one line simultaneously. For single port usage though, it's fine. If you don't yet have a card then I would suggest for the meantime looking elsewhere. There may be nothing wrong with the Voicetronix hardware as such, but clearly it's still got some compatibility issues with Asterisk. Terence ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] x100p card conflicts with DSL modem
I don'thave a DSL filter, Butall my telephones do have filter and work very well. I'll try to use a filter. Thank you all. Best, Michael = Perhaps you forgot to put a filter between the line and your X100P? -Tilghman Eric Wieling [EMAIL PROTECTED] wrote: Do you have a DSL filter on your X100P? Just like any other telephonedevice it needs a DSL filter to keep it from messing up your DSLservice. On Tue, 2004-02-03 at 12:14, Michael Zheng wrote: Hi, all When I use x100p card, my DSL modem can not connect with ISP. Is my card bad or all x100p conflict with DSL modem? Best, Michael __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Go to http://www.digium.com/index.php?menu=documentation and look atthe "Unofficial Links" section. This section has links to a widevariety of 3rd party Asterisk related pages. My page is the"Asterisk Resource Pages".BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it!
RE: [Asterisk-Users] TDM400 FXO???
Will there be no FXO daughter boards for the TDM400? There will be. Units are again in production after having an issue that had us stuck for about 2 months. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voip phones
I am in Japan and I was just going around in some shops in the web... Isamar I found a site somewhere that referenced the livedoor sipphone to: LivedoorSIP phone terminal development original page Http: //www.grandstream.com/y-product.htm The manual it is Which means Livedoor sip phone is infact the grandstream phone. Not even a clone, it's the same thing. :) How did you come across that? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Isamar Maia Sent: Wednesday, February 04, 2004 4:59 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] voip phones not to mention, fortune cookies are included! :) Hey Chinaman... I was wondering if the following SIP phone is just a Grandstream's OEM or just a japanese copy... http://sipphone.livedoor.com/ What do you think? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IPKall-FWD-Asterisk
Joshua, I've been looking into doing the same for my biz as well. I haven't heard of IPKall and perhaps they aren't setup for what you want to do. If this a vital part of your business I'd consider using a commercial IAX provider to give # a toll free or local # for users to call in. If you want more information on how I do it you can reach me at [EMAIL PROTECTED] -- William Suffill On Tue, 2004-02-03 at 20:47, Joshua Colp wrote: Hi Folks, I recently setup an asterisk system in order to provide a telephone phone system for my web hosting business at a very low expense. My problem is that DTMF tones are not being recognized when calling the IPKall phone number. Calling my server via FWD and IAXTel works out fine however. Has anybody experienced this with the IPKall service? are they not passing the DTMF tones or am I doing something wrong? I've tried switching dtmfmode to all the options, but still nothing. Thanks for your help! - Joshua Colp. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voip phones
The photo of the phone says Grandstream Budgtoe 100 when you click to see the larger image of the phone the text on the buttons becomes clear. They look to be selling aservice and the phone to go with it but I'd have to as my wife. She's fluent in Japaneese. I'm not even close. Hey Chinaman... I was wondering if the following SIP phone is just a Grandstream's OEM or just a japanese copy... http://sipphone.livedoor.com/ What do you think? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] diax softphone
I have my asterisk box on the public network. I have a winders box on the public network, running diax. I have a winders box, same setup, behind my Linux iptables firewall, on a private network. Both boxes cann register iax2 to asterisk, and dial, but as soon as asterisk tries to do the native bridge, the call fails. I've poked at this way too much today... what ports do I need to open for this to work? Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 (888) 924-3728 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IPKall-FWD-Asterisk
Greetings, It appears you are correct, as a test I just set it up so when an incoming call came in it dialed Tellme and their system didn't pick up on the DTMF tones either. I guess I will have to wait for my other phone number to be setup. - Joshua Colp. Joshua,I've been looking into doing the same for my biz as well. I haven'theard of IPKall and perhaps they aren't setup for what you want to do.If this a vital part of your business I'd consider using a commercialIAX provider to give # a toll free or local # for users to call in. Ifyou want more information on how I do it you can reach me at[EMAIL PROTECTED] -- William Suffill
[Asterisk-Users] Cisco AC Power Cubes for Sale
Wehave (2) cartons of (56) AC Power Cubes for the Cisco 7905, 7910, 7940 and 7960 IP Phones. These are brand new, and include the power cord. They come with a 1 year warranty. Cost is $17/ea, minimum order of 10 pcs. Cory Andrews***b2 Technologies454 Sonwill DriveBuffalo, NY 14225***voice - 716.630.1555fax - 716.630.1548email - [EMAIL PROTECTED]
[Asterisk-Users] voicemail issue
I've looked online through both google and bugs.digium.com and cannot seem to find this problem anywhere, so i'll ask its unpatched source code for both and everything else works fine. Does anyone else have the issue of asterisk dying with no messages when trying to transfer a voicemail from one mailbox to another? I though it might be an issue with the CVS-12/07/03-20:38:37 so I updated my backup server on Tues 01/15/04 to see if that would resolve the issue. It still dies with only this debuging code. DEBUG[-1084641152]: File asterisk.c, Line 497 (quit_handler): Asterisk ending (0). My backup asterisk server is a dd copy of the primary asterisk server, so everything is configured the same I just change the hdd I'm running in the system. This is occurs when I try to transfer a voicemail, from any user to any user, I've even tried with only one context. Any thoughts on what I can try? Anyone else seen this, or has anyone else even tried to use this feature? Rohde ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Call pre-announcement (was: Asterisk-Users digest, Vol 1 #2711 - 15 msgs)
At 6:26 PM -0500 2/3/04, tad wrote: you can do this with MeetMe, but you don't have to. you can also use Parking, which makes things a little simpler. in either case, the strategy is going to be something like: 1. Record the soundfile 2. Park the inbound caller 3. Use a .call file or the manager interface to initiate an outbound call to the mobile phone 4. play soundfile and prompt the mobile phone user to accept/reject the call 5a. if the call is accepted, transfer the parked call to the mobile user. 5b. if the call is not accepted, unpark the call to some extension and take whatever action you like. be warned, * will probably have no idea whether your outbound call is answered by a human, an answering machine, or hits a busy signal. be further warned, i'm fairly new to * too. however, i have successfully implemented the above. hopefully one of the more astute readers will jump in if it seems like i'm leading you too far astray. ;) hope this helps, tad Tad - Can you be more specific here? I'd like to see this find it's way into the archives on the mailing list first, and then if it makes sense, the Wiki. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail issue
At 8:31 PM -0700 2/3/04, Rohde wrote: I've looked online through both google and bugs.digium.com and cannot seem to find this problem anywhere, so i'll ask its unpatched source code for both and everything else works fine. Does anyone else have the issue of asterisk dying with no messages when trying to transfer a voicemail from one mailbox to another? I though it might be an issue with the CVS-12/07/03-20:38:37 so I updated my backup server on Tues 01/15/04 to see if that would resolve the issue. It still dies with only this debuging code. DEBUG[-1084641152]: File asterisk.c, Line 497 (quit_handler): Asterisk ending (0). My backup asterisk server is a dd copy of the primary asterisk server, so everything is configured the same I just change the hdd I'm running in the system. This is occurs when I try to transfer a voicemail, from any user to any user, I've even tried with only one context. Any thoughts on what I can try? Anyone else seen this, or has anyone else even tried to use this feature? Rohde Rohde - If you have a consistent and reproducable crash of the code, please look in http://bugs.digium.com/ for anyone who perhaps has similar symptoms. (Hint: use the search feature to look for the same type of error keywords) If you find nobody with the same bug, please post the exact circumstances you created to cause the crash in a new bug ticket. If you generate a corefile, please do a bt on the core. Do a quick search of the mailing list for bt corefile or similiar keywords if you're not familiar with that process. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM400 FXO???
Thanks for the reply Mark. When do you expect to ship and are you taking orders? Gene -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Spencer Sent: Tuesday, February 03, 2004 9:40 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] TDM400 FXO??? Will there be no FXO daughter boards for the TDM400? There will be. Units are again in production after having an issue that had us stuck for about 2 months. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Detecting answer supervison from an AGI app
I've got a dumb Western Electric payphone and some homebuilt hardware to control the coin relay which is accessible to Asterisk through the AGI interface. I'd like to be able to set the state of the coin relay to collect at the end of a call if a called party answers. [Hey, I admit this project is being persued just for the fun of it ] Looking through the documentation, there is a way to get channel state, but it only tells you that the channel is bridged and does not tell you the called party has answered. Is there another interface (unix sockets maybe?) which I can use to query the answer supervison state for a given channel? Steve. You can use the manager interface. See the Wiki or various how-to guides scattered across the four corners of Google. You'll only be able to detect answer supervision with PRI or VoIP calls outbound - FXO circuits have no answer supervision. I'm interested in your payphone application; I was given a payphone for the holidays, and I'd like to semi-gut it and put an ATA type device in there and an 802.11 adapter. Feed it 110VAC and point a yagi at it from a few miles away. For that matter, ten or fifteen of them. This is one of those spare-time projects that I've been intentionally ignoring, because I'll quickly become distracted with trying to build a company that does wireless payphones via IP. I have the business plan mostly written in my head already. sigh I assume mine generates coin tones; it would be pretty snappy if * could detect those tones and do something with them when detected in-band in the audio stream. I don't know how the payphones would normally squelch these tones from being replayed into the phones; that would be an interesting thing to dig around inside the phone to find out about (in my free time!) because I know that playback of tones became impossible in the mid 80's due to squelching locally on the phones. I like the novelty of a VoIP phone that collects coins - I'm suspicious of this whole pre-paid thing. ;-) Yet Another Application That Needs Audio Processing After Dial (YAATNAPAD)... JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail issue
How do you start asterisk? using safe_asterisk? or what cli options do you give it? bkw On Tue, 3 Feb 2004, John Todd wrote: At 8:31 PM -0700 2/3/04, Rohde wrote: I've looked online through both google and bugs.digium.com and cannot seem to find this problem anywhere, so i'll ask its unpatched source code for both and everything else works fine. Does anyone else have the issue of asterisk dying with no messages when trying to transfer a voicemail from one mailbox to another? I though it might be an issue with the CVS-12/07/03-20:38:37 so I updated my backup server on Tues 01/15/04 to see if that would resolve the issue. It still dies with only this debuging code. DEBUG[-1084641152]: File asterisk.c, Line 497 (quit_handler): Asterisk ending (0). My backup asterisk server is a dd copy of the primary asterisk server, so everything is configured the same I just change the hdd I'm running in the system. This is occurs when I try to transfer a voicemail, from any user to any user, I've even tried with only one context. Any thoughts on what I can try? Anyone else seen this, or has anyone else even tried to use this feature? Rohde Rohde - If you have a consistent and reproducable crash of the code, please look in http://bugs.digium.com/ for anyone who perhaps has similar symptoms. (Hint: use the search feature to look for the same type of error keywords) If you find nobody with the same bug, please post the exact circumstances you created to cause the crash in a new bug ticket. If you generate a corefile, please do a bt on the core. Do a quick search of the mailing list for bt corefile or similiar keywords if you're not familiar with that process. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VOIP Deployment Concerns
Before I got into my question for the day I'd like to applaud all the helpful folks and time spent behind the asterisk project to get it where it is today. Great work and between this list, the doc list and the irc channel it's been a pleasure to deal with people willing to help others when and if they can. I will be moving in a few months and I'm concerned as to what kind of bandwidth I would need to work effectively. The reason I posed the question here is simple most of my work is remote SSH to various BSD/Linux machines but a majority of my business calls from the office and clients will be routed through Asterisk. Currently I use a SIP phone to my local * then IAX2 w/ 2 providers mentioned on this list. Unfortunately quality begins to suffer based on the amount of upstream used by the other users here. Comcast only has a 256 upstream on it's basic package. I've read the IAX2 Trunking Comparisions from John Todd. Where I'll be moving will more than likely be installing broadband from Adelphia Cable 256up 3mbit down unless I have a valid reason to require the 512up/4mbit down prem. package at approximately 80 a month. I don't have any personal experience with them since I do live in NJ and will be relocating to FL. Any advise would greatly appreciated. 1 last thing without starting a flame war who on this list sells the Grandstream BudgetTone's. Yes I know there are probably better options but I need to keep costs down for internal and personal deployments where some other options would be overkill. Sincerely, William Suffill ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Minor Registration Problem With Polycom Soundpoint IP 500
We recently took a few Polycom Soundpoint IP 500 to test out in Asterisk environment. So far it has been good. Call Hold, Transfer, DMTF etc. However, I do notice every now and then the Polycom fails to register with Asterisk. Asterisk console outputs the following: Feb 3 13:02:32 WARNING[278546]: chan_sip.c:2365 __transmit_response: Unable to determine sequence number from ''Feb 3 13:02:34 NOTICE[278546]: chan_sip.c:5125 handle_request: Failed to authenticate user "DavidLiu" sip:[EMAIL PROTECTED];tag=9F67E426-59D92ED7Feb 3 13:02:36 NOTICE[278546]: chan_sip.c:5125 handle_request: Failed to authenticate user "DavidLiu" sip:[EMAIL PROTECTED];tag=BFDEF35B-1CBC4F2C in sip.conf: canreinvite=yes host=dynamiccanreinvite=yesdtmfmode=rfc2833context=sipport=5060 Usually say after the phone failed to register with Asterisk, I can attempt to place a call. It will fail of course. But then I can try calling again and usually the call will go through and it will successfully re-register itself without needing a restart. What can this be? Surely Polycom is re-registering every 3600 before Asterisk times it out. But Asterisk is just refusing it. By the way, anyone knowwhetherAsterisk is geared towardsRFC3261 or RFC2543? I know Asterisk is not a fully SIP Proxy but lets say if a SIP PSTNGW or a SIP phone is designed under the spec 2543 as suppose to 3261, will it work better or the same with Asterisk? David
[Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk
So, I've managed to consistently lock up my Cisco 7960 (SIP 6.1) to the point where it needs to be unplugged, due to software errors. This is a first. My suspicions are that this bug in Asterisk is causing the lockups: http://bugs.digium.com/bug_view_page.php?bug_id=889 It seems unusual to me that a low volume of bogus SIP messages should lock up the 7960, but that seems to be the case. It seems this only happens on my 7960 that I have completely full of extensions (all six line buttons are lit, two of them are auto-answer.) I think this is one bug tickling another bug; bad messages from * are killing the 7960. I'd like anyone else with experiences with this type of failure with Asterisk to give me a shout; I'm going to report this to Cisco somehow, but don't have enough evidence. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax, trunking, etc.
The majority of sip to pstn gateway providers (vonage, voicepulse, and others) appear to be setup for a one line only type of set up. Their web sites seem to be heavily geared for these one line setups. Anyone willing to comment on what type of pricing plans these providers offer when using iax2 trunking or other methods with asterisk to send multiple (and possibly simultaneous) calls through their gateways ? Thanks, Chris Netlabz, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone used a Grandstream ATA286 with Asterisk
an associate of mine sent me an email of the slick sheet on this one. I understand that mentioning this vendor has resulted in some flamethrowing on the list, and I do not want to cause trouble - just looking for some info. Thanks! Sam Z ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax, trunking, etc.
I use Saww.net and Nufone for IAX2 to PSTN at a per min basis. So if i pushed 5 calls i'd be charge per min for each call. Granted both the companies above cater to * quite heavily. On Wed, 2004-02-04 at 01:40, Chris Clifton wrote: The majority of sip to pstn gateway providers (vonage, voicepulse, and others) appear to be setup for a one line only type of set up. Their web sites seem to be heavily geared for these one line setups. Anyone willing to comment on what type of pricing plans these providers offer when using iax2 trunking or other methods with asterisk to send multiple (and possibly simultaneous) calls through their gateways ? Thanks, Chris Netlabz, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
