RE: [Asterisk-Users] Broadvoice and DTMF
Jay: I hope this input is in some way helpful, if only to confirm your findings... I don't have a Sipura to play with, but I do have a X100P...I used my cell phone to call my BV number, then answered the call from my 7960 and transferred the call out the X100P to my cell providers voicemail access number...once the system answered, I was able to send DTMF to log on to my cell phone voicemail. As with before though, no inbound DTMF working to IVR in asterisk... This worked: Cell Phone -- BV -- SIP -- * -- 7960 --Transfer-- Zap -- Cell phone VM retrieval number Marty -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk Sent: Friday, June 11, 2004 6:19 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF I spoke too soon. However, it's getting even weirder now. Works: Landline --POTS- BV --SIP- * --SIP- Sipura --CABLE- Zap --PCI- * -- VoiceMailMain Doesn't work: Landline --POTS- BV --SIP- * -- VoiceMailMain I don't have any other FXS devices to test this with. ___ 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] FW: question about prepaid app_prepaid
Hi,
As you asked, I have included my diff to what I did for the DIAL command. I
probably didn't stick to some * pre-agreed standard of coding or something,
so if these things offend you then I suggest that you close your eyes. :)
The biggest thing to consider when you are doing a prepaid system is, what
if the person with the same account in/out calls twice? I chose, for now,
just to keep track in a database if an account is in use or not. Only
allowing calls to be placed/answered when the account was not engaged with
another call. It was that fastest way to implement my credit system. This
way is too limited for my liking and wanted to look into more on a way to
check in the scheduler to track credits used in real-time to allow multiple
calls out on the same account. I haven't had time to look into this in much
detail, but I am certain I can hack it into the Asterisk system - if not it
could always be done with an external daemon.
Let me know if anyone has thoughts about this.
Hope this helps people.
Storm.
*** app_dial.c 2004-03-18 16:09:14.0 -0800
--- ../../app_dial.c2004-06-10 17:41:14.044176497 -0700
***
*** 67,72
--- 67,73
'P[(x)]' -- privacy mode, using 'x' as database if provided.\n
'g' -- goes on in context if the destination channel hangs up\n
'A(x)' -- play an announcement to the called party, using x as
file\n
+ 'B(x)' -- Timeout in 'x' seconds after call was bridged.\n /*
CHANGE: Storm Petersen */
In addition to transferring the call, a call may be parked and then
picked\n
up by another user.\n
The optional URL will be sent to the called party if the channel
supports\n
***
*** 360,365
--- 361,367
struct localuser *u;
char info[256], *peers, *timeout, *tech, *number, *rest, *cur;
char privdb[256] = , *s;
+ char szBrdgTO[256] = , *s2;// CHANGE: buffer to store
Bridging Time out. Storm Petersen */
char announcemsg[256] = , *ann;
struct localuser *outgoing=NULL, *tmp;
struct ast_channel *peer;
***
*** 380,385
--- 382,390
struct varshead *headp, *newheadp;
struct ast_var_t *newvar;
int go_on=0;
+ time_t myt;
+ int iBrdgTO=0; /* CHANGE: Time out after call
bridged. Storm Petersen
*/
+
if (!data) {
ast_log(LOG_WARNING, Dial requires an argument
(technology1/number1technology2/number2...|optional timeout)\n);
***
*** 416,422
ast_log(LOG_WARNING, Dial argument takes format
(technology1/number1technology2/number2...|optional timeout)\n);
goto out;
}
!
if (transfer) {
/* XXX ANNOUNCE SUPPORT */
--- 421,449
ast_log(LOG_WARNING, Dial argument takes format
(technology1/number1technology2/number2...|optional timeout)\n);
goto out;
}
!
! /*
! ** CHANGE: Added by Storm Petersen
! ** TIME OUT AFTER CALL WAS BRIDGED.
! */
! if ((s = strstr(data, B())) {
! /* Timeout after Bridging */
! strncpy(szBrdgTO, s + 2, sizeof(szBrdgTO) - 1);
! s2 = szBrdgTO;
! /* Copy the timeout string */
! while(*s2 (*s2 != ')'))
! s2++;
! if (*s2 == ')')
! {
! *s2 = '\0';
! iBrdgTO = atoi(szBrdgTO) + 1;
! }
! else {
! ast_log(LOG_WARNING, Bridge timeout lacking ')'\n);
! iBrdgTO = 0;
! }
! }
!
if (transfer) {
/* XXX ANNOUNCE SUPPORT */
***
*** 703,708
--- 730,746
// Ok, done. stop autoservice
res2 = ast_autoservice_stop(chan);
}
+
+ /*
+ ** CHANGE: Added by Storm Petersen
+ ** Set TimeOut After call was Bridged.
+ */
+ if(iBrdgTO)
+ {
+ time(myt);
+ chan-whentohangup = myt + iBrdgTO;
+ }
+
res = ast_bridge_call(chan, peer, allowredir_in, allowredir_out,
allowdisconnect);
if (res != AST_PBX_NO_HANGUP_PEER)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of usedcanon
Sent: Thursday, June 10, 2004 5:08 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] FW: question about prepaid app_prepaid
I would be interested to share ideas, if you have guidence to offer I would
be greatful
Umar.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Storm D. J.
Petersen
Sent: 11 June 2004 00:48
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] FW: question about prepaid app_prepaid
Hi, I found that the PREPAID system
RE: [Asterisk-Users] ssh key problem
Try logging on to the ftp server on the machine itself. It could be a permissions issue. Umar. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Dean CollinsSent: 12 June 2004 01:54To: [EMAIL PROTECTED]Subject: [Asterisk-Users] ssh key problem Hi Ive need to reinstall my asterisk software (hard drive failure). Im back and running to a make samples state. I have backed up all of my conf files (ok so they were about a week old but much better than starting from scratch), the problem I am having is with WS_FTP Pro. Basically I used to connect to my asterisk server using this software no problems just using root as username and password but I can no longer connect to the new installation. I also use Putty to connect from the same windows machine and no problems with using this. For some reason WS_FTP Pro will not all me to connect with new install, I have deleted WS and reinstalled twice but still no luck. I think it may have something to do with SSH keys. Any thoughts? Cheers, Dean
[Asterisk-Users] Prepending for 9NxxNxxx - adding the area code for 7 digit dialing
How do you prepend. I want to be able to dial 7 digits instead of of 11 for local calls. Can someone post there extensions.conf part that is relavent? ___ 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] 2 NuFone lines- which one to dial out on
I am setting up 2 nufone lines. I want to make them both availiable for dial-out. How do you syntax it in extensions.conf so that it figures out which one is avaliable and dials out on it. Also how do you setup the name part of callerid for the outgoing lines? ___ 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] Broadvoice and DTMF
Makes me think that the problem isn't with Broadvoice at all, but rather with Asterisk's DTMF recognition. I'm running CVS Head from late April. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marty Mastera Sent: Saturday, June 12, 2004 2:11 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Jay: I hope this input is in some way helpful, if only to confirm your findings... I don't have a Sipura to play with, but I do have a X100P...I used my cell phone to call my BV number, then answered the call from my 7960 and transferred the call out the X100P to my cell providers voicemail access number...once the system answered, I was able to send DTMF to log on to my cell phone voicemail. As with before though, no inbound DTMF working to IVR in asterisk... This worked: Cell Phone -- BV -- SIP -- * -- 7960 --Transfer-- Zap -- Cell phone VM retrieval number Marty -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk Sent: Friday, June 11, 2004 6:19 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF I spoke too soon. However, it's getting even weirder now. Works: Landline --POTS- BV --SIP- * --SIP- Sipura --CABLE- Zap --PCI- * -- VoiceMailMain Doesn't work: Landline --POTS- BV --SIP- * -- VoiceMailMain I don't have any other FXS devices to test this with. ___ 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] Prepending for 9NxxNxxx - adding the area code for 7 digit dialing
Quite simple really,
You could do the following assuming your area code is 0207 (london !)
exten = 9NXXNXXX,1, Dial(SIP/0207${EXTEN})
Umar.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jacob Hunter
Sent: 12 June 2004 09:15
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Prepending for 9NxxNxxx - adding the area code
for 7 digit dialing
How do you prepend. I want to be able to dial 7 digits instead of of
11 for local calls.
Can someone post there extensions.conf part that is relavent?
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Re: [Asterisk-Users] Prepending for 9NxxNxxx - adding the area code for 7 digit dialing {Scanned}
If I have understood correctly then this is what you want:
exten = _55.,1,Dial(IAX2/userid:[EMAIL PROTECTED]/441332${EXTEN}||r)
The last line is what you are after I think. All local calls in my area begin
with 5-7 so I pattern match this and then extend it with 44 (UK code) and
1332 (local trunk) before passing to my VOIP/PSTN gateway.
Regards
Chris Wilber
On Saturday 12 June 2004 09:14, Jacob Hunter wrote:
How do you prepend. I want to be able to dial 7 digits instead of of
11 for local calls.
Can someone post there extensions.conf part that is relavent?
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Re: [Asterisk-Users] Prepending for 9NxxNxxx - adding the area code for 7 digit dialing
usedcanon wrote:
Quite simple really,
You could do the following assuming your area code is 0207 (london !)
exten = 9NXXNXXX,1, Dial(SIP/0207${EXTEN})
Umar.
The London code is 020 the 7 or the 8 is part of the local number now.
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Re: [Asterisk-Users] Prepending for 9NxxNxxx - adding the area code for 7 digit dialing
im in US On Jun 12, 2004, at 1:14 AM, Jacob Hunter wrote: How do you prepend. I want to be able to dial 7 digits instead of of 11 for local calls. Can someone post there extensions.conf part that is relavent? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prepending for 9NxxNxxx - adding the area code for 7 digit dialing {Scanned}
Jason,
I changed my mind about how to do this and didn't change what I cut/pasted
into my reply to you. Should have said:
[5-7]X,1,Dial(IAX2/userid:[EMAIL PROTECTED]/441332${EXTEN}||r)
I think the way I originally gave would mean that internal extensions couldn't
begin with 5, 6 or 7 which would affect parking.
Regarding your later post where you mention you are US - is there and
particular significance of this?
Regards
Chris
On Saturday 12 June 2004 10:54, Christopher Wilber wrote:
If I have understood correctly then this is what you want:
exten = _55.,1,Dial(IAX2/userid:[EMAIL PROTECTED]/441332${EXTEN}||r)
The last line is what you are after I think. All local calls in my area
begin with 5-7 so I pattern match this and then extend it with 44 (UK code)
and 1332 (local trunk) before passing to my VOIP/PSTN gateway.
Regards
Chris Wilber
On Saturday 12 June 2004 09:14, Jacob Hunter wrote:
How do you prepend. I want to be able to dial 7 digits instead of of
11 for local calls.
Can someone post there extensions.conf part that is relavent?
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RE: [Asterisk-Users] Prepending for 9NxxNxxx - adding the area code for 7 digit dialing
I do something very similar with my home system. I'm using Voicepulse, and
want to be able to dial 10 digits instead of 11 for local calls. My local
area codes are 321 and 407. Everything else is considered LD.
exten = _407NXX,1,Dial(IAX2/[EMAIL PROTECTED]/1${EXTEN})
exten = _321NXX,1,Dial(IAX2/[EMAIL PROTECTED]/1${EXTEN})
exten = _1NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
Assuming I wanted to dial only 7 digits for calls within 407 I could do this:
exten = _NXX,1,Dial(IAX2/[EMAIL PROTECTED]/1407${EXTEN})
Hope this helps!
Reid
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jacob Hunter
Sent: Saturday, June 12, 2004 6:49 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Prepending for 9NxxNxxx - adding the area code
for 7 digit dialing
im in US
On Jun 12, 2004, at 1:14 AM, Jacob Hunter wrote:
How do you prepend. I want to be able to dial 7 digits instead of of
11 for local calls.
Can someone post there extensions.conf part that is relavent?
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Re: [Asterisk-Users] Prepending for 9NxxNxxx - adding the area code for 7 digit dialing
Does this look right
exten = _9NXX,1,SetCallerID(831-XXX-)
exten = _9NXX,2,Dial(IAX2/[EMAIL PROTECTED]/1831${EXTEN})
exten = _9NXX,3,Congestion
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RE: [Asterisk-Users] Prepending for 9NxxNxxx - adding the area code for 7 digit dialing
Not sure about the SetCallerID bit, but the rest does. Your SetCallerID may
work (not sure). I use SetCallerID(Reid Forrest (407) 555-1212)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jacob Hunter
Sent: Saturday, June 12, 2004 7:48 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Prepending for 9NxxNxxx - adding the area code
for 7 digit dialing
Does this look right
exten = _9NXX,1,SetCallerID(831-XXX-)
exten = _9NXX,2,Dial(IAX2/[EMAIL PROTECTED]/1831${EXTEN})
exten = _9NXX,3,Congestion
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Re: [Asterisk-Users] 2 NuFone lines- which one to dial out on
On Sat, 12 Jun 2004, Jacob Hunter wrote: I am setting up 2 nufone lines. I want to make them both availiable for dial-out. How do you syntax it in extensions.conf so that it figures out which one is avaliable and dials out on it. You don't really have 2 nufone lines - you have two Nufone numbers. You can send as many outbound calls as you like to Nufone - just Dial(IAX2/...) away. Set the callerid on the outbound calls the way you want them and that's what the callee will see. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Prepending for 9NxxNxxx - adding the area code for 7 digit dialing
I know :-), it was just an example.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris Lee
Sent: 12 June 2004 11:06
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Prepending for 9NxxNxxx - adding the area
code for 7 digit dialing
usedcanon wrote:
Quite simple really,
You could do the following assuming your area code is 0207 (london !)
exten = 9NXXNXXX,1, Dial(SIP/0207${EXTEN})
Umar.
The London code is 020 the 7 or the 8 is part of the local number now.
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RE: [Asterisk-Users] ssh key problem
I dont know how to log onto ftp from asterisk itself. I dont know why this would matter as I can log on using Putty. This is what happens on ws_ftp pro Connecting to 192.168.7.20:22 Connected to 192.168.7.20:22 in 0.00 seconds, Waiting for Server Response Server Welcome: SSH-1.99-OpenSSH_3.8p1 Debian 1:3.8p1-3 Client Version: SSH-2.0-WS_FTP-8.03-2003.12.16 DSS Signature Verified Session Keys Created Ciphers Created New Client-Server ciphers in place. New Client-Server ciphers in place. Completed SSH Key Exchange. New Keys in place. Then each user name comes back as failed even though they work with putty From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of usedcanon Sent: Saturday, 12 June 2004 5:54 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] ssh key problem Try logging on to the ftp server on the machine itself. It could be a permissions issue. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Dean Collins Sent: 12 June 2004 01:54 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ssh key problem Hi Ive need to reinstall my asterisk software (hard drive failure). Im back and running to a make samples state. I have backed up all of my conf files (ok so they were about a week old but much better than starting from scratch), the problem I am having is with WS_FTP Pro. Basically I used to connect to my asterisk server using this software no problems just using root as username and password but I can no longer connect to the new installation. I also use Putty to connect from the same windows machine and no problems with using this. For some reason WS_FTP Pro will not all me to connect with new install, I have deleted WS and reinstalled twice but still no luck. I think it may have something to do with SSH keys. Any thoughts? Cheers, Dean
RE: [Asterisk-Users] Prepending for 9NxxNxxx - adding the area code for 7 digit dialing
looks fine to me
Umar
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jacob Hunter
Sent: 12 June 2004 12:48
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Prepending for 9NxxNxxx - adding the area
code for 7 digit dialing
Does this look right
exten = _9NXX,1,SetCallerID(831-XXX-)
exten = _9NXX,2,Dial(IAX2/[EMAIL PROTECTED]/1831${EXTEN})
exten = _9NXX,3,Congestion
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RE: [Asterisk-Users] ssh key problem
Take a look in your syslogs on the asterisk server for clues. Not sure how Debian logs, but on Redhat you need to inspect /var/log/secure and /var/log/messages. It looks like your keys are fine. Id check to make sure ws_ftp is using the correct SSH version (i.e. SSH 2). Compare your putty settings with your ws_ftp settings and see what the differences are. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Saturday, June 12, 2004 7:59 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] ssh key problem I dont know how to log onto ftp from asterisk itself. I dont know why this would matter as I can log on using Putty. This is what happens on ws_ftp pro Connecting to 192.168.7.20:22 Connected to 192.168.7.20:22 in 0.00 seconds, Waiting for Server Response Server Welcome: SSH-1.99-OpenSSH_3.8p1 Debian 1:3.8p1-3 Client Version: SSH-2.0-WS_FTP-8.03-2003.12.16 DSS Signature Verified Session Keys Created Ciphers Created New Client-Server ciphers in place. New Client-Server ciphers in place. Completed SSH Key Exchange. New Keys in place. Then each user name comes back as failed even though they work with putty From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of usedcanon Sent: Saturday, 12 June 2004 5:54 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] ssh key problem Try logging on to the ftp server on the machine itself. It could be a permissions issue. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Dean Collins Sent: 12 June 2004 01:54 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ssh key problem Hi Ive need to reinstall my asterisk software (hard drive failure). Im back and running to a make samples state. I have backed up all of my conf files (ok so they were about a week old but much better than starting from scratch), the problem I am having is with WS_FTP Pro. Basically I used to connect to my asterisk server using this software no problems just using root as username and password but I can no longer connect to the new installation. I also use Putty to connect from the same windows machine and no problems with using this. For some reason WS_FTP Pro will not all me to connect with new install, I have deleted WS and reinstalled twice but still no luck. I think it may have something to do with SSH keys. Any thoughts? Cheers, Dean
Re: [Asterisk-Users] BudgeTone hold?
check the ver of the flash on your phone for firmware you might need to update. ___ 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] Cisco ATA-186 Firmware upgrade
I am currently running 2.16. Is there good reason to get the update to 3.1? Anything significant? Otherwise I am happy how it is, i just don't want to miss out on anything. ___ 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] Problem with E1
Hi I'm from spain, so forgive my English. I'm somewhat new to asterisk, and i'm having trouble in getting my line to work (i'm in a little hurry!). I have an E100P, and it seems everything is configured ok, but when receiving a call, I get the following message D-Channel on span 1 up (four times), then the call ends, asterisk says D-Channel on span 1 down. And that's all. My telco says they reveive no answer on the line. Asterisk gets the call and does nothing. I have done the cable from the modem to the card myself, crossing pins 1 and 2 with 4 and 5 respectively, as my telco has told me, and the card shows a greed led (not the flashing red I had when using a LAN crossover cable). Can the cable be the problem? I'm using asterisk CVS-05/11/04-17:24:48 Zaptel.conf -- span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 loadzone = es defaultzone=es Zapata.conf: -- [channels] language=es context=default switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no jitterbuffers=4 group = 1 channel = 1-15,17-31 I have also tried with span=1,0,0,ccs,hdb3 I have asked for crc, but its not active in my telco. Can anyone help? By the way, how can I print information on zap intensive debugging to a file in case I need to post? Thanks a lot! ___ 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] Broadvoice and DTMF
Does anyone know how Asterisk detects the DTMF tone on a SIP connection. Is it through the audio, or is it through the SIP Info? James Jones Broadvoice Technical Support From: [EMAIL PROTECTED] on behalf of Jay Milk Sent: Sat 6/12/2004 4:52 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Makes me think that the problem isn't with Broadvoice at all, but rather with Asterisk's DTMF recognition. I'm running CVS Head from late April. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marty Mastera Sent: Saturday, June 12, 2004 2:11 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Jay: I hope this input is in some way helpful, if only to confirm your findings... I don't have a Sipura to play with, but I do have a X100P...I used my cell phone to call my BV number, then answered the call from my 7960 and transferred the call out the X100P to my cell providers voicemail access number...once the system answered, I was able to send DTMF to log on to my cell phone voicemail. As with before though, no inbound DTMF working to IVR in asterisk... This worked: Cell Phone -- BV -- SIP -- * -- 7960 --Transfer-- Zap -- Cell phone VM retrieval number Marty -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk Sent: Friday, June 11, 2004 6:19 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF I spoke too soon. However, it's getting even weirder now. Works: Landline --POTS- BV --SIP- * --SIP- Sipura --CABLE- Zap --PCI- * -- VoiceMailMain Doesn't work: Landline --POTS- BV --SIP- * -- VoiceMailMain I don't have any other FXS devices to test this with. ___ 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] MWI on Cisco ATA-186 (SIP)
I am trying to set up the Message Waiting Indicator (stutter tone/light) so that my cisco ata-186 will let my phones know there is a message waiting. However this does not seem to be very well documented. I found this on wiki [EMAIL PROTECTED] ... where does that go? Do I put it in my SIP.conf definition for my cisco ata, or where. In my SIP cisco definition i already have a mailbox=mailboxnumber. What do I need to do to get this to work? And which context am I putting there, is it the same context as the sip device, or is the context from the voicemail.conf. ___ 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] MWI on Cisco ATA-186 (SIP)
I do not have a Cisco ATA-186 but rather a Grandstream Handytone ATA-286... To get MWI going on it I would assume you would do the same. You put it in your sip.conf... Here's my example... [201] type=friend secret=x host=dynamic context=intern canreinvite=no dtmfmode=rfc2833 mailbox=201 of course it not being exactly the same :D Hope this works out for you. Stephen Rosebush [EMAIL PROTECTED] Jacob Hunter wrote: I am trying to set up the Message Waiting Indicator (stutter tone/light) so that my cisco ata-186 will let my phones know there is a message waiting. However this does not seem to be very well documented. I found this on wiki [EMAIL PROTECTED] ... where does that go? Do I put it in my SIP.conf definition for my cisco ata, or where. In my SIP cisco definition i already have a mailbox=mailboxnumber. What do I need to do to get this to work? And which context am I putting there, is it the same context as the sip device, or is the context from the voicemail.conf. ___ 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] MWI on Cisco ATA-186 (SIP)
It should go in sip.conf the context is whatever context you specified in voicemail.conf Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jacob Hunter Sent: 12 June 2004 14:31 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] MWI on Cisco ATA-186 (SIP) I am trying to set up the Message Waiting Indicator (stutter tone/light) so that my cisco ata-186 will let my phones know there is a message waiting. However this does not seem to be very well documented. I found this on wiki [EMAIL PROTECTED] ... where does that go? Do I put it in my SIP.conf definition for my cisco ata, or where. In my SIP cisco definition i already have a mailbox=mailboxnumber. What do I need to do to get this to work? And which context am I putting there, is it the same context as the sip device, or is the context from the voicemail.conf. ___ 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] Cisco ATA-186 Firmware upgrade
There probably are a number of fixes. I have not used the ATA's for some time, however as the saying goes .. If it ain't broke don't fix it. So if it is working for you don't bother. Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jacob Hunter Sent: 12 June 2004 14:12 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco ATA-186 Firmware upgrade I am currently running 2.16. Is there good reason to get the update to 3.1? Anything significant? Otherwise I am happy how it is, i just don't want to miss out on anything. ___ 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] MWI on Cisco ATA-186 (SIP)
the mail=1234 seems to have worked... is it necessary to do the [EMAIL PROTECTED] I dont think i set a contect (as there is only 2 mailboxes) so would it be default.. On Jun 12, 2004, at 6:40 AM, usedcanon wrote: It should go in sip.conf the context is whatever context you specified in voicemail.conf Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jacob Hunter Sent: 12 June 2004 14:31 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] MWI on Cisco ATA-186 (SIP) I am trying to set up the Message Waiting Indicator (stutter tone/light) so that my cisco ata-186 will let my phones know there is a message waiting. However this does not seem to be very well documented. I found this on wiki [EMAIL PROTECTED] ... where does that go? Do I put it in my SIP.conf definition for my cisco ata, or where. In my SIP cisco definition i already have a mailbox=mailboxnumber. What do I need to do to get this to work? And which context am I putting there, is it the same context as the sip device, or is the context from the voicemail.conf. ___ 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] Broadvoice and DTMF
Depends how you have it configured. To the best of my knowledge asterisk supports, inband, info and RFC2833 Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of James Jones Sent: 12 June 2004 14:26 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Does anyone know how Asterisk detects the DTMF tone on a SIP connection. Is it through the audio, or is it through the SIP Info? James Jones Broadvoice Technical Support From: [EMAIL PROTECTED] on behalf of Jay Milk Sent: Sat 6/12/2004 4:52 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Makes me think that the problem isn't with Broadvoice at all, but rather with Asterisk's DTMF recognition. I'm running CVS Head from late April. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marty Mastera Sent: Saturday, June 12, 2004 2:11 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Jay: I hope this input is in some way helpful, if only to confirm your findings... I don't have a Sipura to play with, but I do have a X100P...I used my cell phone to call my BV number, then answered the call from my 7960 and transferred the call out the X100P to my cell providers voicemail access number...once the system answered, I was able to send DTMF to log on to my cell phone voicemail. As with before though, no inbound DTMF working to IVR in asterisk... This worked: Cell Phone -- BV -- SIP -- * -- 7960 --Transfer-- Zap -- Cell phone VM retrieval number Marty -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk Sent: Friday, June 11, 2004 6:19 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF I spoke too soon. However, it's getting even weirder now. Works: Landline --POTS- BV --SIP- * --SIP- Sipura --CABLE- Zap --PCI- * -- VoiceMailMain Doesn't work: Landline --POTS- BV --SIP- * -- VoiceMailMain I don't have any other FXS devices to test this with. ___ 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] problem with app_prepaid!
Im trying to compile asterisk with app_prepaid. Im getting compile error. I have paste my logs here: http://www.pastebin.com/73053. I wold be so pleasure if any one can help me! Best Regards Hekuran ___ 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] MWI on Cisco ATA-186 (SIP)
If you put the name of the context in the voicemail.conf the 'default' I would assume you wouldn't need to. Steve Jacob Hunter wrote: the mail=1234 seems to have worked... is it necessary to do the [EMAIL PROTECTED] I dont think i set a contect (as there is only 2 mailboxes) so would it be default.. On Jun 12, 2004, at 6:40 AM, usedcanon wrote: It should go in sip.conf the context is whatever context you specified in voicemail.conf Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jacob Hunter Sent: 12 June 2004 14:31 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] MWI on Cisco ATA-186 (SIP) I am trying to set up the Message Waiting Indicator (stutter tone/light) so that my cisco ata-186 will let my phones know there is a message waiting. However this does not seem to be very well documented. I found this on wiki [EMAIL PROTECTED] ... where does that go? Do I put it in my SIP.conf definition for my cisco ata, or where. In my SIP cisco definition i already have a mailbox=mailboxnumber. What do I need to do to get this to work? And which context am I putting there, is it the same context as the sip device, or is the context from the voicemail.conf. ___ 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] MWI on Cisco ATA-186 (SIP)
If you have more than one context than yes, otherwise I believe it will work with out it. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jacob Hunter Sent: 12 June 2004 14:45 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] MWI on Cisco ATA-186 (SIP) the mail=1234 seems to have worked... is it necessary to do the [EMAIL PROTECTED] I dont think i set a contect (as there is only 2 mailboxes) so would it be default.. On Jun 12, 2004, at 6:40 AM, usedcanon wrote: It should go in sip.conf the context is whatever context you specified in voicemail.conf Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jacob Hunter Sent: 12 June 2004 14:31 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] MWI on Cisco ATA-186 (SIP) I am trying to set up the Message Waiting Indicator (stutter tone/light) so that my cisco ata-186 will let my phones know there is a message waiting. However this does not seem to be very well documented. I found this on wiki [EMAIL PROTECTED] ... where does that go? Do I put it in my SIP.conf definition for my cisco ata, or where. In my SIP cisco definition i already have a mailbox=mailboxnumber. What do I need to do to get this to work? And which context am I putting there, is it the same context as the sip device, or is the context from the voicemail.conf. ___ 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] Broadvoice and DTMF
has anyone tried info? James Jones Broadvoice Technical Support From: [EMAIL PROTECTED] on behalf of usedcanon Sent: Sat 6/12/2004 9:45 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Depends how you have it configured. To the best of my knowledge asterisk supports, inband, info and RFC2833 Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of James Jones Sent: 12 June 2004 14:26 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Does anyone know how Asterisk detects the DTMF tone on a SIP connection. Is it through the audio, or is it through the SIP Info? James Jones Broadvoice Technical Support From: [EMAIL PROTECTED] on behalf of Jay Milk Sent: Sat 6/12/2004 4:52 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Makes me think that the problem isn't with Broadvoice at all, but rather with Asterisk's DTMF recognition. I'm running CVS Head from late April. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marty Mastera Sent: Saturday, June 12, 2004 2:11 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Jay: I hope this input is in some way helpful, if only to confirm your findings... I don't have a Sipura to play with, but I do have a X100P...I used my cell phone to call my BV number, then answered the call from my 7960 and transferred the call out the X100P to my cell providers voicemail access number...once the system answered, I was able to send DTMF to log on to my cell phone voicemail. As with before though, no inbound DTMF working to IVR in asterisk... This worked: Cell Phone -- BV -- SIP -- * -- 7960 --Transfer-- Zap -- Cell phone VM retrieval number Marty -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk Sent: Friday, June 11, 2004 6:19 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF I spoke too soon. However, it's getting even weirder now. Works: Landline --POTS- BV --SIP- * --SIP- Sipura --CABLE- Zap --PCI- * -- VoiceMailMain Doesn't work: Landline --POTS- BV --SIP- * -- VoiceMailMain I don't have any other FXS devices to test this with. ___ 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 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] Capture user input
Title: Message Hi I just wanted to know if anyone has done the following, or knows how to. When a customer dials into *, we would then ask them to for an account number (which they would type in with the key pad), we would then ask them to select and option (1 to 9). We would then like to capture this information and send it in an email. All hints 'n tips welcome...
[Asterisk-Users] 'background' problem
I have a 'day' and a 'night' mode. In the day mode, I play a 'background' message which is interruptable by the pushing of a DTMF key - ie - all is normal. In night mode - I decided to get smarter... I play two backgrounds with a 'sayunixtime' in between and now DTMF does nothing - the menu times out to my 'lets get the operator then'... If I change the three commands to a single 'playback' - everything works as expected. Is this because 'sayunixtime' breaks things? Should I use something else instead of the first 'playback'? This is with a very recent version of Head CVS. Code: exten = s,7,Playback(posix-welcome-afterhours) ; Welcome to Posix; Systems After hours support, Our business hours are Monday ; to Friday, 8am to 5pm. The time is now exten = s,8,SayUnixTime(||AIMP); A:Day, I:Hours, M:Minutes, P:am/pm exten = s,9,Playback(posix-welcome-afterhours-try) ; Please dial 1 ; for support, ...Blah... or Stay on the line for an operator Suggestions? -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ 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] Sending SABME continuosly. Urgent help needed!
Hi, I'm trying to install an E1 PRI, and I need it working by Monday, but although everything seems ok, I get no response to calls. When I make a pri extense debug on span 1, I repeatedly get the following: Sending Set Asynchronous Balanced Mode Extended [ 00 01 7f ] Unnumbered frame: SAPI: 00 C/R: 0 EA: 0 TEI: 999 EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended] 0 bytes of data. And nothing else. When making a call to that E1, I see the message D-Channel on span 1 up 4 times, and then a Informational frame, with TEI:000 EA:1 and anything else with zero (13 bytes of data). Stopping T_203 timer Starting T_200 timer Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 0/0x0) (Originator) Message type: RESTART (70) Channel ID (len= 5) [Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan:0 Chan Sel Reserved Ext: 1 Coding:0 Number Specified Channel Type: 3 Ext: 1 Spare: o Resetting Inidicated Channel (0) ] Then D-Channel on span 1 downn, and finally, after a while: (...) Warning[11276]: chan_zap.c:5993 zt_pri_error: PRI: Read on 46 failed: Unknown error 500 (...) Notice[11276]: chan_zap.c:6708 pri_dchannel: PRI got event: 8 on span 1 I think I have Asterisk stable version 1.0, CVS updated today Can anyone help me? Please! :S Zaptel.conf -- span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 loadzone = es defaultzone=es Zapata.conf: -- [channels] language=es context=default switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no jitterbuffers=4 group = 1 channel = 1-15,17-31 I have also tried with span=1,0,0,ccs,hdb3 immediate = yes The line has not CRC activated (I have asked) Thanks! ___ 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] Cisco ATA-186 Firmware upgrade
im interested if there are any codec adds or major things like that... On Jun 12, 2004, at 6:44 AM, usedcanon wrote: There probably are a number of fixes. I have not used the ATA's for some time, however as the saying goes .. If it ain't broke don't fix it. So if it is working for you don't bother. Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jacob Hunter Sent: 12 June 2004 14:12 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco ATA-186 Firmware upgrade I am currently running 2.16. Is there good reason to get the update to 3.1? Anything significant? Otherwise I am happy how it is, i just don't want to miss out on anything. ___ 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] Call Relaying
Hello All, I have a small * server in my home office with several IP phones. The system is not fully in service yet as I'm still hunting for a cost effective FXO adapter that I can rely upon for my two primary PSTNs. That said, I'd like to move it into service for another application...which brings up a question. I'd really like to stop making international calls from my cell phone when I'm travelling. Can someone point me to an example of extension logic that accepts an incomming call on a known connection then allows the caller to access local dialtone to make an international call? I have a DID from VoicePulse Connect which I don't really use for much. I could make that the gateway so that all calls comming in on that DID have access to outbound dialing. I could also screen the incomming callerid so that only my or my wife's cell phones get validated. Anyone have something comparable that I might look at as a starting point? Thanks, Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 Who do you think you're foolin'? - Paul Simon ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ 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] Broadvoice and DTMF
Hi, Yes, I have tried all three: inband, rfc2833 and info. No luck with any. Michael Swan Neon Software, Inc. At 10:31 AM 6/12/2004 -0400, you wrote: has anyone tried info? James Jones Broadvoice Technical Support From: [EMAIL PROTECTED] on behalf of usedcanon Sent: Sat 6/12/2004 9:45 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Depends how you have it configured. To the best of my knowledge asterisk supports, inband, info and RFC2833 Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of James Jones Sent: 12 June 2004 14:26 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Does anyone know how Asterisk detects the DTMF tone on a SIP connection. Is it through the audio, or is it through the SIP Info? James Jones Broadvoice Technical Support From: [EMAIL PROTECTED] on behalf of Jay Milk Sent: Sat 6/12/2004 4:52 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Makes me think that the problem isn't with Broadvoice at all, but rather with Asterisk's DTMF recognition. I'm running CVS Head from late April. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marty Mastera Sent: Saturday, June 12, 2004 2:11 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Jay: I hope this input is in some way helpful, if only to confirm your findings... I don't have a Sipura to play with, but I do have a X100P...I used my cell phone to call my BV number, then answered the call from my 7960 and transferred the call out the X100P to my cell providers voicemail access number...once the system answered, I was able to send DTMF to log on to my cell phone voicemail. As with before though, no inbound DTMF working to IVR in asterisk... This worked: Cell Phone -- BV -- SIP -- * -- 7960 --Transfer-- Zap -- Cell phone VM retrieval number Marty -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk Sent: Friday, June 11, 2004 6:19 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF I spoke too soon. However, it's getting even weirder now. Works: Landline --POTS- BV --SIP- * --SIP- Sipura --CABLE- Zap --PCI- * -- VoiceMailMain Doesn't work: Landline --POTS- BV --SIP- * -- VoiceMailMain I don't have any other FXS devices to test this with. ___ 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 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] 2 NuFone lines- which one to dial out on
I have tried it with 4 simultaneous calls and it worked like a charm. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 12, 2004 8:41 AM Subject: Re: [Asterisk-Users] 2 NuFone lines- which one to dial out on On Sat, 12 Jun 2004, Simon Dorfman wrote: Steve, Do you know if the same is true with inbound calls? Let's say I have an 800 number with nufone and I have 10 Snom phones hooked up to *. If 10 people call in 10 seconds, assuming I have * configured correctly, can all 10 Snom phone users pick up and answer the 10 calls? Thanks, Simon in New Orleans I believe that the answer is yes - though I haven't tried it personally. 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 ___ 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] BudgeTone hold?
On Jun 11, 2004, at 8:04 PM, Seth Mattinen wrote: I can't seem to make the Hold button function on the GS BudgeTone-100. I'm trying a procedure like this: 1) On a call 2) Press Hold button 3) Hang up phone You can sorta do this by pressing the speakerphone button prior to placing the receiver on the hookswitch. When you pick up the receiver, just press the hold button again to resume your call. I too found out the hard way. HTH, Ryan Thrash ___ 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] Problems with Alcatel Speedtouch ST280
Title: Message Does anybody has experience with the SIP phone of Alcatel the ST280. I can't make a call with this phone. Everytime I make a call I get the error Jun 12 19:38:38 WARNING[1133718080]: chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 1 (Response) From an other excentsion I can call this sip phone. If I look with sip show peers it;s also good registered. What I'm doing wrong. The telephone is on the same network as the asterisk server is. Edwig Knol Net TopicLaakweg 204 2521 SW Den Haag [EMAIL PROTECTED] www.nettopic.nlOffice: +31(0)70-3240534Fax: +31(0)70-3191512
Re: [Asterisk-Users] BudgeTone hold?
On Jun 12, 2004, at 7:11 AM, [EMAIL PROTECTED] wrote: check the ver of the flash on your phone for firmware you might need to update. I'm using the latest version I could find, 1.0.5.00. There doesn't seem to be any official source for the latest firmware that I could tell, however. -- Seth ninja monkey Mattinen [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] DECT delay once hungup
I've got the following setup: IAXy - Dect Base Station. When you dial from a SIP phone (cisco 7960), the rings with very little delay. However, if you hangup it takes 3-4 rings after hanging up before the dect base station phone stops ringing. The same applies when an incoming call is directed from PSTN FXO - Dect Base. Is there a fix to this I've looked about on voip-info but cant find any information that might be causing it. Kind Regards, Chris Bond ___ 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] DECT delay once hungup
Hi, - Original Message - From: Chris Bond [EMAIL PROTECTED] I've got the following setup: IAXy - Dect Base Station. When you dial from a SIP phone (cisco 7960), the rings with very little delay. However, if you hangup it takes 3-4 rings after hanging up before the dect base station phone stops ringing. The same applies when an incoming call is directed from PSTN FXO - Dect Base. Is there a fix to this I've looked about on voip-info but cant find any information that might be causing it. I have a Siemens Gigaset S100 DECT phone connected to an ATA186 with the same problem. I have the second ATA line connected to another DECT (Philips Onis2 Memo) which does not have this delay. It seems to be a phone problem. BR, Dan ___ 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] 2 NuFone lines- which one to dial out on
FWIW, if anyone is interested the same goes for Voicepulse. I've been using it for multiple inbound and outbound calls for about a month. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Saturday, June 12, 2004 12:44 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 2 NuFone lines- which one to dial out on I have tried it with 4 simultaneous calls and it worked like a charm. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 12, 2004 8:41 AM Subject: Re: [Asterisk-Users] 2 NuFone lines- which one to dial out on On Sat, 12 Jun 2004, Simon Dorfman wrote: Steve, Do you know if the same is true with inbound calls? Let's say I have an 800 number with nufone and I have 10 Snom phones hooked up to *. If 10 people call in 10 seconds, assuming I have * configured correctly, can all 10 Snom phone users pick up and answer the 10 calls? Thanks, Simon in New Orleans I believe that the answer is yes - though I haven't tried it personally. 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 ___ 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] Simplified Voicemail app / keeping peace withcohabitants
I didn't know that; I just hit 33 like with Bell Canada's voice mail to fast forward to the end, and then 7 to erase... Can you hit 77 during message playback to do the same thing? I've gotten pretty used to 339 to resave those messages that will be deleted shortly.. ___ 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] Asterisk on FreeBSD News
Asterisk on FreeBSD News CURRENTLY: Asterisk and libpri in Asterisk-current (CVS head) build and run on FreeBSD 5.2.1. The last major change (support for multiple CPUs) has been incorporated into Asterisk-current. Asterisk should now be thread safe on FreeBSD and testing on dual CPUs has just begun, thanks to Chris Stenton [Thanks Chris!] Zapata and zaptel in Asterisk-current have not yet been ported. The zaptel driver has been enhanced significantly since the 0.9.0 version present in FreeBSD's ports. So, the Asterisk application will not build or run with FreeBSD's zaptel 0.9.0 as it stands. CHANGES THIS WEEK: We widened support for flavors of FreeBSD and OpenBSD. Between Olle Johansson, Rich Neese and I, we're working on: - FreeBSD 4.9, 5.2.1 and -current, as well as - OpenBSD 3.5. Work has begun on merging FreeBSD support into the zaptel driver in Asterisk-current. Maxim Sobolev, the maintainer of FreeBSD's zaptel port, wants to work together on it, and Mark Spencer supports the goal of FreeBSD zaptel support in Asterisk CVS. The most important criteria for achieving this is to avoid *any* diminished performance on Linux. WHAT'S NEXT: Rich Neese is leading the effort for FreeBSD support for *all* bundled modules and features including: - OpenH323 - SCCP/Skinny - Speex and other codecs - Conferencing - Spandsp - Fax app - Festival - Paging and Intercom - Calling card app - Postgres We intend to continue to integrate support for the FreeBSD ports so that installation of necessary 3rd party packages is taken care of automatically. We continue to focus on getting *BSD changes accepted into Asterisk-current CVS current rather than distributing patches or snapshots for testing. This means waiting longer for the code to be available for testing, but achieving stability sooner for FreeBSD in Asterisk-current in the long run, since testing will focus on Asterisk-current, where it belongs. This was once a criteria for the FreeBSD zaptel bounty, so we hope that this is what the community wants. 1-713-218-7616 (enum) Rich Murphey ___ 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] Broadvoice and DTMF
are you doing internal codec translation internally (i.e. g.711u to g.729)? James Jones Broadvoice Technical Support From: [EMAIL PROTECTED] on behalf of Michael Swan Sent: Sat 6/12/2004 12:23 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Hi, Yes, I have tried all three: inband, rfc2833 and info. No luck with any. Michael Swan Neon Software, Inc. At 10:31 AM 6/12/2004 -0400, you wrote: has anyone tried info? James Jones Broadvoice Technical Support From: [EMAIL PROTECTED] on behalf of usedcanon Sent: Sat 6/12/2004 9:45 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Depends how you have it configured. To the best of my knowledge asterisk supports, inband, info and RFC2833 Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of James Jones Sent: 12 June 2004 14:26 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Does anyone know how Asterisk detects the DTMF tone on a SIP connection. Is it through the audio, or is it through the SIP Info? James Jones Broadvoice Technical Support From: [EMAIL PROTECTED] on behalf of Jay Milk Sent: Sat 6/12/2004 4:52 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Makes me think that the problem isn't with Broadvoice at all, but rather with Asterisk's DTMF recognition. I'm running CVS Head from late April. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marty Mastera Sent: Saturday, June 12, 2004 2:11 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Jay: I hope this input is in some way helpful, if only to confirm your findings... I don't have a Sipura to play with, but I do have a X100P...I used my cell phone to call my BV number, then answered the call from my 7960 and transferred the call out the X100P to my cell providers voicemail access number...once the system answered, I was able to send DTMF to log on to my cell phone voicemail. As with before though, no inbound DTMF working to IVR in asterisk... This worked: Cell Phone -- BV -- SIP -- * -- 7960 --Transfer-- Zap -- Cell phone VM retrieval number Marty -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk Sent: Friday, June 11, 2004 6:19 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF I spoke too soon. However, it's getting even weirder now. Works: Landline --POTS- BV --SIP- * --SIP- Sipura --CABLE- Zap --PCI- * -- VoiceMailMain Doesn't work: Landline --POTS- BV --SIP- * -- VoiceMailMain I don't have any other FXS devices to test this with. ___ 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 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] Call Relaying
The DISA application is traditionally for this functionality. But beware, anything that give outside users the ability to make calls can be abused. -Brett Original message Date: Sat, 12 Jun 2004 12:20:28 -0400 From: Michael Graves [EMAIL PROTECTED] Subject: [Asterisk-Users] Call Relaying To: [EMAIL PROTECTED] [EMAIL PROTECTED] Hello All, I have a small * server in my home office with several IP phones. The system is not fully in service yet as I'm still hunting for a cost effective FXO adapter that I can rely upon for my two primary PSTNs. That said, I'd like to move it into service for another application...which brings up a question. I'd really like to stop making international calls from my cell phone when I'm travelling. Can someone point me to an example of extension logic that accepts an incomming call on a known connection then allows the caller to access local dialtone to make an international call? I have a DID from VoicePulse Connect which I don't really use for much. I could make that the gateway so that all calls comming in on that DID have access to outbound dialing. I could also screen the incomming callerid so that only my or my wife's cell phones get validated. Anyone have something comparable that I might look at as a starting point? Thanks, Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 Who do you think you're foolin'? - Paul Simon ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ 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] 2 NuFone lines- which one to dial out on
gafachi.com too.. they only have new york for did, BUT they have 2c a minute anywhere in the us... I have nufone inbound and gafachi out On Jun 12, 2004, at 12:15 PM, Reid A. Forrest wrote: FWIW, if anyone is interested the same goes for Voicepulse. I've been using it for multiple inbound and outbound calls for about a month. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Saturday, June 12, 2004 12:44 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 2 NuFone lines- which one to dial out on I have tried it with 4 simultaneous calls and it worked like a charm. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 12, 2004 8:41 AM Subject: Re: [Asterisk-Users] 2 NuFone lines- which one to dial out on On Sat, 12 Jun 2004, Simon Dorfman wrote: Steve, Do you know if the same is true with inbound calls? Let's say I have an 800 number with nufone and I have 10 Snom phones hooked up to *. If 10 people call in 10 seconds, assuming I have * configured correctly, can all 10 Snom phone users pick up and answer the 10 calls? Thanks, Simon in New Orleans I believe that the answer is yes - though I haven't tried it personally. 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 ___ 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] DECT delay once hungup
If you take * out of the equation and put the DECT phone and a normal handset straight on to a PSTN line, you will probably and the same thing happens - the DECT phone keeps ringing for 3-4 rings after you answer the call on the normal handset. Simon Brown -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Bond Sent: Sunday, 13 June 2004 4:52 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] DECT delay once hungup I've got the following setup: IAXy - Dect Base Station. When you dial from a SIP phone (cisco 7960), the rings with very little delay. However, if you hangup it takes 3-4 rings after hanging up before the dect base station phone stops ringing. The same applies when an incoming call is directed from PSTN FXO - Dect Base. Is there a fix to this I've looked about on voip-info but cant find any information that might be causing it. Kind Regards, Chris Bond ___ 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] Broadvoice and DTMF
At 05:26 PM 6/12/2004 -0400, you wrote: are you doing internal codec translation internally (i.e. g.711u to g.729)? James Jones Broadvoice Technical Support Hi, I don't know the answer to your question. We have specified g711-ulaw as the only negotiated codec when connecting to BV as that's what BV suggested as their preferred codec. Michael Swan Neon Software, 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] ISDN Bristuff + analogue
Hi does anyone know if its possible to run Bristuff together with a tdm card in the same computer. I get an error when trying to start asterisk in chan_zap.c My zaptel.conf looks like this: loadzone=us defaultzone=us fxsks=1 fxoks=2 fxoks=3 span=1,1,3,ccs,ami bchan=4-5 dchan=6 Thanks Clive _ For super low premiums ,click here http://www.dialdirect.co.za/quote ___ 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] Call forwarding
Hi, I've just installed asterisk a few days ago and figured most of it out, but I can't seem to forward calls. In the setup before I used asterisk, when people called my (ISDN) number and I didn't answer the phone in 4 rings, the call would be transferred to my cellphone. I ofcourse made that setting by once dialing, from the ISDN line, *610*X#, being my cellphone number. The advantages of those transfers is that 1) on my cellphone I see the original called ID 2) after the transfer the ISDN line is not busy anymore. I tried to achieve the same in *, but so far no luck. I know I can transfer calls like this (my setup now), in extensions.conf: exten = s,1,Dial(SIP/21,14) exten = s,2,Dial(Modem/g1:06,30) Which first tries to dial SIP extension 21 for 14 seconds (my regular phone), and then dials my cellphone. It works but has 2 disadvantages; on the cellphone I can only see the phone number of the * server calling out, not of the person who is actually trying to reach me, and when transferring calls this way both my ISDN lines on the BRI are used for this connection (1 incoming and 1 outgoing). Now I am not too familiar with modern telco equipment, but is there a way * can tell the telco's switch to transfer this call to another number, just like it would in the *610* setup? Regards, Kenneth van Grinsven ___ 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] Call Relaying
Brett, Thank you for this. It was as I expected. To minimize my security risk I have limited the disa context to a series of preset speed dials to my UK associates, which is primarily the cellular cost that I wanted curbed. Thanks again, Michael On Sat, 12 Jun 2004 16:37:15 -0500, Brett Nemeroff wrote: The DISA application is traditionally for this functionality. But beware, anything that give outside users the ability to make calls can be abused. -Brett Original message Date: Sat, 12 Jun 2004 12:20:28 -0400 From: Michael Graves [EMAIL PROTECTED] Subject: [Asterisk-Users] Call Relaying To: [EMAIL PROTECTED] [EMAIL PROTECTED] Hello All, I have a small * server in my home office with several IP phones. The system is not fully in service yet as I'm still hunting for a cost effective FXO adapter that I can rely upon for my two primary PSTNs. That said, I'd like to move it into service for another application...which brings up a question. I'd really like to stop making international calls from my cell phone when I'm travelling. Can someone point me to an example of extension logic that accepts an incomming call on a known connection then allows the caller to access local dialtone to make an international call? I have a DID from VoicePulse Connect which I don't really use for much. I could make that the gateway so that all calls comming in on that DID have access to outbound dialing. I could also screen the incomming callerid so that only my or my wife's cell phones get validated. Anyone have something comparable that I might look at as a starting point? Thanks, Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 Who do you think you're foolin'? - Paul Simon ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ 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 -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ...All we are is dust in the wind. - Kansas ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ 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 behind Iptables: What's the magic?
I have a similar problem. I use a BT101 (outside) to dial another BT101 (inside). The phone inside can hear me but I cannot hear it. Thanks, Luan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Cuthie Sent: 09 June 2004 16:49 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk behind Iptables: What's the magic? Which way is the audio working? -brian Isamar Maia wrote: I tried some combinations of setup seen in some postings and didn't get success on this yet. I have grandstream phones outside the network trying to call an * server inside my network through NAT/Iptables. The problem that I'm facing is one-way audio. Any suggestion? 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 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.701 / Virus Database: 458 - Release Date: 07/06/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.705 / Virus Database: 461 - Release Date: 12/06/2004 ___ 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] 2 NuFone lines- which one to dial out on
If anyone signed up for gafachi because of this please email me On Jun 12, 2004, at 3:03 PM, Jacob Hunter wrote: gafachi.com too.. they only have new york for did, BUT they have 2c a minute anywhere in the us... I have nufone inbound and gafachi out On Jun 12, 2004, at 12:15 PM, Reid A. Forrest wrote: FWIW, if anyone is interested the same goes for Voicepulse. I've been using it for multiple inbound and outbound calls for about a month. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Saturday, June 12, 2004 12:44 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 2 NuFone lines- which one to dial out on I have tried it with 4 simultaneous calls and it worked like a charm. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 12, 2004 8:41 AM Subject: Re: [Asterisk-Users] 2 NuFone lines- which one to dial out on On Sat, 12 Jun 2004, Simon Dorfman wrote: Steve, Do you know if the same is true with inbound calls? Let's say I have an 800 number with nufone and I have 10 Snom phones hooked up to *. If 10 people call in 10 seconds, assuming I have * configured correctly, can all 10 Snom phone users pick up and answer the 10 calls? Thanks, Simon in New Orleans I believe that the answer is yes - though I haven't tried it personally. 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 ___ 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] BudgeTone hold?
Richard Neese wrote: check the ver of the flash on your phone for firmware you might need to update. Are you running a version of the BT firmware that permits the handset to be placed on hook without terminating a call on Hold? If so, what is the version number? Certainly no release of the BT firmware I've tried supports this. If not, what was your post based upon? Reading a Release Note that indicated that this issue had been addressed or mere hopeful speculation? g. ___ 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] DID/T1
So DIDs are sharing available channels. In particular for ISDNs are DIDs sharing available channels? -- David Kwok CISSP,(ISC)2 61282315751 ext 1002 FWD#/IAXTEL# : 17001813482 ext 1002 ___ 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] cdr_addon_mysql.c
Ed Devine wrote: Following the latest * CVS update, my MySQL was broken. Following the update, Asterisk-addons would compile fine, but when I ran asterisk I got the following error: ERROR[1202489024]: cdr_addon_mysql.c:203 mysql_log: Failed to insert into databas I then tried using the patch (bug id 0001823) from bugs.digium.com, and found that Asterisk-addons would no longer compile, giving me the following errors: [...] I don't know, if it helps, but I also had some problems compiling cdr_addon_mysql.c, recently. To finally solve it that's what I did: 1) mkdir /tmp/A; cd /tmp/A 2) logged in to cvs 3) cvs checkout asterisk asterisk-addons 4) cd asterisk-addons 5) adjust the CFLAG section in the Makefile to look as follows: CFLAGS+=-fPIC CFLAGS+=-I../asterisk CFLAGS+=-I../asterisk/include CFLAGS+=-D_GNU_SOURCE 6) then run make --ca ___ 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] DID/T1
So DIDs are sharing available channels. Not necessarily -- you are confusing two separate concepts: 1: DIDs are a mechanism for having more than one phone number that can be routed to one or more voice channels and for identifying which number was dialled when a call is presented by the CO. Depending on the part of the world you are in and the technology used, this mechanism may be referred to as DDI or MDN (Multiple Directory Numbers). Sharing available channels is the ability to have incoming calls (to the same number or different numbers) routed by the CO to more than one channels depending on channel availability and the rotation algorithm used by the CO. This is usually referred to as rotary, overlines, or call forward busy. In particular for ISDNs are DIDs sharing available channels? Let's deconstruct this question into its component parts. (a) Can ISDN support DID numbers? Technically yes. Whether or not your telco will provide this functionality is a question only your telco can answer. We have seen telcos that would not provide DIDs (or MDNs as they are usually called with respect to BRI circuits) on ISDN BRI circuits but would on PRIs. (b) Can ISDN support overlines. Technically yes. Your telco may, as above, be less cooperative. Again, our experience is that the possibilities with PRIs are greater than BRIs. In short -- you will have to talk to your service provider (or change service providers) to get what you want. Hope this helps. George Pajari www.netVOICE.ca www.ip-centrex.ca ___ 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] Junghanns QuadBRI stable?
Hi, I have an asterisk box with a QuadBRI. I am close to make it work like I want: Incoming calls to SIP are OK, outgoing calls are quiet working thanks to Michael Sandee advices. My real problem is that my * box is freezing very often when I probe qozap (bri-stuff 0.0.2). I have tried two distribs (Debian Kernel 2.6, Mandrake 9.1 Kernel 2.4) and it always work for a few minutes and freezes. At present I am working on a HP DL380. Tomorrow, I'll try to make it work on a debian kernel 2.4 with a different PC (Shuttle SB51G) My question: Is Junghanns QuadBRI stable? If so, on which platform/distrib may I use it? Thanks Gaël ___ 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] ASTTAPI 0.03 hangup not working
Hey Nick, what a nice project. This will help us in database and outlook aided dialing. Great One Problem: If you initiate a call via Outlook, * dials the extension, afterwards dials the tapi-requested number and proceeds bridging the two calls together. Thats fine. When you end the call via outlook hangup-button, *-manager gets a hangup-command (seen in verbose mode), but * doesn't really hangup and the call is still there. Outlook is trying forever to hangup and the dial button is greyed out. When you hangup the phone, * of course cuts all related channels but Outlook does not realize it. A restart of Outlook makes it work again but just till next hangup is done. What I've seen is that Outlook doesn't recognize that the call is successfully bridged and ongoing - it still shows that it's dialing. Regards Rob ___ 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 on FreeBSD News
Asterisk on FreeBSD News thank you! ___ 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] 2 NuFone lines- which one to dial out on
How did you find them? There is nothing in google except that they used to be a budget webhosting company and some sort of media business prior to that. http://web.archive.org/web/*/http://www.gafachi.com - Original Message - From: Jacob Hunter [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 12, 2004 8:56 PM Subject: Re: [Asterisk-Users] 2 NuFone lines- which one to dial out on Just so you all know gafachi.com has a very good referral program.. you get a good amount of minutes (10% of whatever your referrals purchased). So make sure to tell people u refer to put your username as their referral code. They are a great service provider, from what i see... great support in chat and AIM... so id support them and pass on the word. jacob On Jun 12, 2004, at 4:56 PM, Jacob Hunter wrote: If anyone signed up for gafachi because of this please email me On Jun 12, 2004, at 3:03 PM, Jacob Hunter wrote: gafachi.com too.. they only have new york for did, BUT they have 2c a minute anywhere in the us... I have nufone inbound and gafachi out On Jun 12, 2004, at 12:15 PM, Reid A. Forrest wrote: FWIW, if anyone is interested the same goes for Voicepulse. I've been using it for multiple inbound and outbound calls for about a month. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Saturday, June 12, 2004 12:44 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 2 NuFone lines- which one to dial out on I have tried it with 4 simultaneous calls and it worked like a charm. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 12, 2004 8:41 AM Subject: Re: [Asterisk-Users] 2 NuFone lines- which one to dial out on On Sat, 12 Jun 2004, Simon Dorfman wrote: Steve, Do you know if the same is true with inbound calls? Let's say I have an 800 number with nufone and I have 10 Snom phones hooked up to *. If 10 people call in 10 seconds, assuming I have * configured correctly, can all 10 Snom phone users pick up and answer the 10 calls? Thanks, Simon in New Orleans I believe that the answer is yes - though I haven't tried it personally. 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 ___ 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 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] 2 NuFone lines- which one to dial out on
In the Asterisk chat on IRC... its very helpful... luck and chance! theyre great! On Jun 12, 2004, at 7:12 PM, Steve Totaro wrote: How did you find them? There is nothing in google except that they used to be a budget webhosting company and some sort of media business prior to that. http://web.archive.org/web/*/http://www.gafachi.com - Original Message - From: Jacob Hunter [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 12, 2004 8:56 PM Subject: Re: [Asterisk-Users] 2 NuFone lines- which one to dial out on Just so you all know gafachi.com has a very good referral program.. you get a good amount of minutes (10% of whatever your referrals purchased). So make sure to tell people u refer to put your username as their referral code. They are a great service provider, from what i see... great support in chat and AIM... so id support them and pass on the word. jacob On Jun 12, 2004, at 4:56 PM, Jacob Hunter wrote: If anyone signed up for gafachi because of this please email me On Jun 12, 2004, at 3:03 PM, Jacob Hunter wrote: gafachi.com too.. they only have new york for did, BUT they have 2c a minute anywhere in the us... I have nufone inbound and gafachi out On Jun 12, 2004, at 12:15 PM, Reid A. Forrest wrote: FWIW, if anyone is interested the same goes for Voicepulse. I've been using it for multiple inbound and outbound calls for about a month. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Saturday, June 12, 2004 12:44 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 2 NuFone lines- which one to dial out on I have tried it with 4 simultaneous calls and it worked like a charm. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 12, 2004 8:41 AM Subject: Re: [Asterisk-Users] 2 NuFone lines- which one to dial out on On Sat, 12 Jun 2004, Simon Dorfman wrote: Steve, Do you know if the same is true with inbound calls? Let's say I have an 800 number with nufone and I have 10 Snom phones hooked up to *. If 10 people call in 10 seconds, assuming I have * configured correctly, can all 10 Snom phone users pick up and answer the 10 calls? Thanks, Simon in New Orleans I believe that the answer is yes - though I haven't tried it personally. 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 ___ 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 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] (no subject)
I have a list of all my local (free) on my POTS prefixes. Is there a way to integrate that so * decides if it is going to use iax or POTS? There is about 60 prefixes.. 1831-XXX extension.conf clipping help -- Gafachi.com - referal code hunter81 instant iax termination - 2 cents a minute Also they have a great referal program, tell them jacob, hunter81 sent you
[Asterisk-Users] Local calls to x100p all else to iax term
Sorry I screwed up. I wanted to write back with the proper subject. I have a list of all my local prefixes(free) on my POTS. Is there a way to integrate that so * decides if it is going to use iax or POTS? There is about 60 prefixes.. 1831-XXX extension.conf clipping help -- Gafachi.com - referal code hunter81 instant iax termination - 2 cents a minute Also they have a great referal program, tell them jacob, hunter81 sent you
