Re: [Asterisk-Users] Dialing 800 numbers with VOIP
On Tue, 10 Feb 2004, Chris Clifton wrote: I'll second this. For the past 4 days, Vonage can't figure out how to process our visa check card. In the meantime, Nufone has us setup with an account, ready to roll. - Chris Clifton Interesting. I've been trying to get Jeremy to set up a second 800 DID for us since Feb 4th and I'm not having any luck. Our Paypal payment has been processed, and I've spoken to Jeremy on IRC.. perhaps today will be my lucky day. :) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] Spanish indications configurationº
Hi all We've been using * for a while here in Spain, but some people has told us that they have problems when they type an extension calling to us. I've been trying to find out what's going on, and it's an issue that only happens with some ISDN and analog calls, not from mobile calls as long as i have observe. My concern is about the indications.conf Spanish telco lines configuration, Is in the * list any Spanish user that can share this configuration with me and see if it's ok?? i would really appreciate it. Diego
Re: [Asterisk-Users] x101p beeps/sceeching
On Thursday 12 February 2004 18:11, Jeff Gustafson wrote: I'm experiencing periodic beeps or screeching when I'm on a call via the x101p card to/from PSTN. Echo cancellation seems to be working fine. The beeps seem to happen with echo cancellation on or off. Is there a setting I can tweak for this? The problem does not occur if I'm making pure SIP calls. If you're not using an ADSI-capable phone, turn ADSI off in zapata.conf. -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] Digium connectivity issue?
Are others seeing hugh delays and/or lack of connectivity to Digium? 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
[Asterisk-Users] GS BT-100 echo
I picked up a GS 100 phone based on the overall good response I've heard of these phones. One thing I'm fighting with, which I can't find any info on, is a *real* bad local echo on the GS. The remote end doesn't hear it, and all the docs I see about echocancel deal with hardwired phones/ports (fxs/fso). Phone software is: Software Version: Program--1.0.4.45Bootloader--1.0.0.13HTML--1.0.0.20 if that matters. sip.conf for the phone is: [gs1] type=friend username=gs1 secret= host=dynamic canreinvite=no nat=yes qualify=1000 disallow=all allow=alaw allow=ulaw 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] chan_local and variables
We need to implement the following:
Call comes in, ring ZAP/1 (6 rings)
For the last two rings, also ring ZAP/2
I have the following (which works as expected):
[incoming]
exten = s,1,DIAL(Local/[EMAIL PROTECTED] Local/[EMAIL PROTECTED],18)
[test1]
exten = 123,1,Dial(ZAP/1)
exten = 124,1,Wait(12)
exten = 124,2,Dial(ZAP/2)
I can't figure out how to back this into a macro. I would like to use the
setup below, but it seems impossible to pass variables down into the local
channel. Can anyone confirm this, or suggest some alternative? (I've
tried the /n on the chan_local, with no success)
[macro-standard-extension-coverage]
exten = s,1,SetVar(PrimaryChannel=${ARG1})
exten = s,2,SetVar(DelayedChannel=${ARG2})
exten = s,3,Dial(Local/[EMAIL PROTECTED] Local/[EMAIL PROTECTED],18)
[delayed]
exten = 1,1,Dial(${PrimaryChannel})
exten = 2,1,Wait(12)
exten = 2,2,Dial(${DelayedChannel})
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Re: [Asterisk-Users] Solved! x101p beeps/sceeching
It turned out that a Gb/E card was too close to the modem causing activity on the card to bleed over to the modem. ...Jeff On Thu, 2004-02-12 at 16:11, Jeff Gustafson wrote: I'm experiencing periodic beeps or screeching when I'm on a call via the x101p card to/from PSTN. Echo cancellation seems to be working fine. The beeps seem to happen with echo cancellation on or off. Is there a setting I can tweak for this? The problem does not occur if I'm making pure SIP calls. ...Jeff ___ 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] Direct mailbox transfer
Thanks guys I will try that in the morning...
Sean
-Original Message-
From: John Fraizer [mailto:[EMAIL PROTECTED]
Sent: Thursday, February 12, 2004 5:33 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Direct mailbox transfer
Sean Garland wrote:
How would one implement a direct mailbox transfer using the macros?
What I want to do is have the person who answers the call to be able
to transfer the call directly into a persons unavailable mailbox.
Thanks
Sean Garland, MCP+I, A+
Siskiyou Technology Consultants
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Add the following context and make sure it's available to the person who
will be transferring people:
[direct-vm]
exten = _*9.,1,Voicemail2(u${EXTEN:2})
exten = _*9.,2,Hangup()
To transfer someone straight to VM, they simple blind transfer them to
*9[voicemail extension]
Works like a charm.
John
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Re: [Asterisk-Users] More external call control
On Thu, 12 Feb 2004, [EMAIL PROTECTED] waxed: My questions are as follows, (but before I begin; I know there is queueing and some ACD functionality in *, but I need to do this externally. I want the queueing decisions to be external because my central queue engine handles things like email, chat, etc as well as calls): You might then want to consider just putting people in MusicOnHold extensions, Parking extensions, etc. Putting them in a * queue -- when they are already in your own external queue -- would be a flawed redundancy, owing to differences in queue logic. In other words, can I send some message to * that will tell it to route a call in queue to a specific extension by a unique ID (because there may be los of calls queued). This is possible. But not really through AGI. You would need to use the manager interface, which is more for external control. Not to say that you couldn't create manager commands that would in turn put you in the dialplan to run a specific AGI... While the call is in queue, can I send commands to have different announcements played? There's no moh stuff for the manager interface, but you could look at the 'Redirect' manager command. Maybe consider moving someone from one queue to another, where there happens to be different moh. If a call hangs up while in queue, is that a step in extensions.conf so I can call my script with that info? Extension 'h' is reserved for hangups, but it isn't limited to just queues. --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info Pronunciation Guide: Maj == May Fingerprint: 43D6 799C F6CF F920 6623 DC85 C8A3 CFFE F0DE C146 ___ 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] 800 numbers / Skinny - IAX2
Hi Folks, I'm trying to route IAX2 calls to 800 numbers from a Skinny channel and the log says: -- IAX2[69.73.19.178:4569]/4 stopped sounds -- Asked to indicate 'Stop tone' condition on channel Skinny/[EMAIL PROTECTED] And no audio happens. It's working for Zap and SIP channels, though. Tried to google about that but didn't find anything. Any thoughts? Thanks, 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] Hide outgoing CallerId on Zap interface
Hi there I know I have asked a somehow similar question earlier but since then I've tried some different things which isn't working. I want to completely hide my outgoing CallerId when dialing out on my Zap interface. I've tried a lot of different settings in sip.conf and hoped that zap would hide the CallerId if sip was told to do so, but that wasn't the case. Then I've tried to set hidecallerid=yes in zapata.conf (and restarted *) but this only results in my main number CallerId being displayed. Is it somehow possible to completely hide the CallerId, like when someone from a secret number is calling and the display on my mobile says Secret number ? And if that is possible, is it then possible to do it on a per-user basis configured via sip.conf? regards, Mickey Binder ___ 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] festival voices
(Parameter.set 'Audio_Method 'linux16audio) ;(Parameter.set 'Audio_Method 'esdaudio) ;(Parameter.set 'Audio_Method 'mplayeraudio) ;(Parameter.set 'Audio_Method 'sunaudio) ; American female I'm using the cepstral frank with festival ;) (set! voice_default 'voice_frank) in /root/.festivalrc bkw On Thu, 12 Feb 2004, Tony Buser wrote: Chris Albertson wrote: try adding a set of parens like this: festivalcommand=((voice_don_diphone)(tts_textasterisk %s'file)(quit))\n Unfortunately that results in the following error at the asterisk console: Feb 12 19:45:27 WARNING[409626]: app_festival.c:437 festival_exec: Festival returned ER And the following error in the festival_server: SIOD ERROR: unbound variable : don_diphone SIOD ERROR: unbound variable : \n Have you seen festivox? It's a tool for building voices The key to making festival sound natural is to get the timming and entonation right. The astrisk app uses festivels demo test to speech application which is just that a quick dirty demo. Have you seen the markup language on the CMU site? http://www-2.cs.cmu.edu/~awb/festival_demos/sable.html Sable can do MUCH better then the simple tts application. Thanks, I'll take a look at that. So to use Sable I'd have to use festival from like an AGI script and not inside the Festival() function in extension.conf? I blindly tried pasting sable markup in there and the best I could do was get it to read back the markup and all. :) ___ 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] Anybody going to the Spring VON converence [OT]
I'm going. Would be great to have an Asterisk gathering. /O ___ 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] channel bank - Adit 600
hi! I would like to check the apllicability of Adit 600 and Adtran 750 in converting FXO + FXS to E1/T1 channel to be sent to a * voip box. we r currently using pleidaes channel bank and it has the problem FXO lines hanging forever.(Don't disconnect). Bundle of FXOs are obtained from inhouse PBX. What I'm looking for is a simple Channel Bank doing, FXO + FXS(bundle) - E1/T1 with Answer/Disconnect supervision Voice Activity Detection. impedance matching, minimized echo, etc... Can I achieve the above simple task with Adit600 ? Eventhough it's been recommended by * h/w guide it looks someting different to a channel bank. Perhaps someone could pop an Idea on this. denzel. ___ 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] Voicemail Password Digit Timeout
I was wondering if there was any way to change the digit timeout or some setting of that sort on the voicemail password entry. Currently when our users enter their passwords they have to enter them very rapidly, otherwise asterisk will log the number twice. So if someone entered a voicemail password of 1234 slowly and deliberately on our system the asterisk receives it as the following number, 11223344 and thus returns the passcode invalid message. System: Asterisk CVS-02/10/04-13:27:57 built by [EMAIL PROTECTED] on a i686 running Linux 3 X100P cards 5 Snom200 phones Sincerely, Ryan R. Fligg Secured Digital Storage, Inc. 104 SW 4th St. Des Moines, IA 50309 Phone: (515)-244-6290 Cell: (720)-841-5802 Website: www.dstorage.com E-Mail: [EMAIL PROTECTED] attachment: winmail.dat
[Asterisk-Users] Codecs compile error on yellowdog
greetings I'm running yellow dog 3.1 compiling Asterisk 0.7.1 during the make process it seems to die at the GSM build. (summerized) As build goes' through must remake `src/add.o'. entering dirctory `/usr/local/asterisk-0.7.1/codecs/gsm' gcc -march= -fomit-frame-pointer -c -DneedFunctionprototypes=1 -funroll-loops -fPIC -DSASR -DNDEBUG -DWAV49 -I./inc src/add.c puting child (gsm/lib/libgsm.a) leave child put child (arc/add.o) leave child cc1: invalid option `arch=' Got SIGCHLD; 1 unreaped children reaping loosing child make[1]; *** [src/add.o] error 1 removing child from chain then it does the same for libgsm.a make: *** [gsm/lib/libgsm.a] Error 2 I'm thinking that the libgsm is not installed in my distribution can anyone guide me to on what I need to install to get passed this error? TIA --jeff --- jeff donovan basd network operations (610) 807 5571 x4 AIM xtdonovan fwd# 248217 ___ 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: Rhino channel bank and aastra PT390 phones
Hi, Anyone setup a Rhino channel bank ?... any issues. I got it working with normal pots phones but I cant get it to work with Aastra PT390 phones. The phones get dialtone but the asterisk does see any DTMF digits dialed from the phone. Any ideas would be helpfull. Thanks John Bittner Simlab.net ___ 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] System freeze
On Thursday 12 February 2004 16:34, you wrote: On Thu, 2004-02-12 at 12:28, B. J. Bomar wrote: I too have seen a couple of system freezes for no apparent reason. I am * on a RH9 box with kernel 2.4.20-28.9. Without wanting to sound like a RH basher that I normally am, could this be a RH issue since I haven't noticed(maybe foggy memory) any other distros mentioned as freezing? RH is known for having their custom patches to the kernel that may or may not make it to the official tree. Rarely, but it does happen, one patch could be wrong or broken. Steven, You know me well enough to know that I am not defending RedHat :) We have had random system lockups on our Mandrake 9.1 / TE410P system over the past couple of weeks - since this Dual PIII sever was put on line. We have not had anything jump out at us as a probable cause. Tilghman asked if we should put our prior server back in service and I told him no because we had already sold the T400P out of it :/ Memo to Digium, some of us are pining for another run of T400Ps Howard White ___ 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: Codecs compile error on yellowdog
I must be doing something wrong i have installed gsm.rpm manually and tried to recompile, but i still get the same error. make[1]: Entering directory `/usr/local/asterisk-0.7.1/codecs' make -C gsm lib/libgsm.a make[2]: Entering directory `/usr/local/asterisk-0.7.1/codecs/gsm' gcc -O6 -march=ppc -fomit-frame-pointer -c -DNeedFunctionPrototypes=1 -funroll-loops -fPIC -DSASR -DNDEBUG-DWAV49 -I./inc src/add.c cc1: invalid option `arch=ppc' make[2]: *** [src/add.o] Error 1 make[2]: Leaving directory `/usr/local/asterisk-0.7.1/codecs/gsm' make[1]: *** [gsm/lib/libgsm.a] Error 2 make[1]: Leaving directory `/usr/local/asterisk-0.7.1/codecs' make: *** [subdirs] Error 1 I also can't find and libgsm RPM that will build on PPC --- sheesh --- jeff donovan basd network operations (610) 807 5571 x4 AIM xtdonovan fwd# 248217 ___ 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] HELP!!!! Having problems Starting Asterisk
I have been trying to start asterisk all night after a reboot I keep getting this error scrolling up the screen ouch: error while writing audio data broken pipe when I go to another console there are 4 instances of mpg123 running and when I do TOP they are taking 100% CPU between them I have re installed mgp123 but it still doesn't help any Ideas? Thanks in advance Robb ___ 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] Direct mailbox transfer
My extensions start with 7XXX
[ext-direct-to-vm]
exten = _67XXX,1,Wait(1)
exten =
_67XXX,2,Playback(/var/lib/asterisk/sounds/voicemail/YOUR_CONTEXT_HERE/${EXT
EN:1}/greet)
exten = _67XXX,3,Voicemail2(${EXTEN:[EMAIL PROTECTED])
exten = _67XXX,4,Playback(vm-goodbye)
exten = _67XXX,5,Hangup
Transfer to 6 plus the extension and you get the name of the mailbox owner
and then it goes direct to his/her mailbox
John
- Original Message -
From: Sean Garland [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, February 12, 2004 7:15 PM
Subject: [Asterisk-Users] Direct mailbox transfer
How would one implement a direct mailbox transfer using the macros?
What I want to do is have the person who answers the call to be able to
transfer the call directly into a persons unavailable mailbox. Thanks
Sean Garland, MCP+I, A+
Siskiyou Technology Consultants
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[Asterisk-Users] Adtran 750 - what do I need
Could anyone tell me what I need to include in the purchase of an Adtran 750 to work with a T100P? Obviously, I'd need a combination or FXO and FXS boards to fit my application, but, are there any other boards that are required? Does every Adtran include the proper port to connect to the T100P, or is that needed as well? Assuming it is not included, are the FXO, FXS and T1 cards all that is needed? Does anyone suggest a particular vendor for this piece of equipment? ___ 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
