Re: [asterisk-users] remove queue call
Yup I just copy and paste to it but it shown not a known channel. On Thu, Aug 28, 2008 at 6:47 PM, Steven Howes [EMAIL PROTECTED] wrote: Did you tab complete it to make sure it was right? On 28 Aug 2008, at 11:39, Rilawich Ango wrote: I got the message below after I issue the soft hangup. sip01*CLI soft hangup Local/[EMAIL PROTECTED],2 Local/[EMAIL PROTECTED],2 is not a known channel Any other way to kill the call without affecting other queues and calls? On Thu, Aug 28, 2008 at 4:09 PM, Steven Howes [EMAIL PROTECTED] wrote: Try CLI soft hangup Local. On 28 Aug 2008, at 09:01, Rilawich Ango wrote: Hi , Actually, there are 3 queues in the server. Only one queue (2700) has problem. I want to reset or remove the caller only in 2700 without affecting other queues or calls. Does it work for my case? On Thu, Aug 28, 2008 at 11:49 AM, Andy Kuo [EMAIL PROTECTED] wrote: Hi, Try CLI soft hangup Local. Andy On 8/27/08, Rilawich Ango [EMAIL PROTECTED] wrote: Hi all, I have the following queue and members. I found that there is a call stuck in the queue so other call can't enter the queue. I want to know whether we can remove the call (by CLI) to free the queue. ango 2700 has 1 calls (max unlimited) in 'rrmemory' strategy (35s holdtime), W:0, C:134, A:48, SL:88.8% within 120s Members: Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken no calls yet Callers: 1. Local/[EMAIL PROTECTED],2 (wait: 113:19, prio: 0) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api- digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Queue's
Hello Philipp, Yes, I have autofill set in queues.conf. I suspect that this behaviour is because the Polycom phones I use have 2 lines. Has anyone used this function with polycom phones before? Also, my agents are Dynamic, perhaps this works better with Static agents? Here's my queues.conf (with commented lines deleted for easier reading): [general] persistentmembers = yes autofill = yes monitor-type = MixMonitor [sales] strategy = rrmemory wrapuptime=15 Date: Thu, 28 Aug 2008 13:56:49 +0200 From: Philipp Kempgen [EMAIL PROTECTED] Subject: Re: [asterisk-users] Asterisk Queue's To: Asterisk Users asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 Tobias Ahlander schrieb: I have a sample queue with two dynamic agents. When the first caller calls in to the system, the first agents phone starts to ring. Then another caller calls in to the queue, but the other phone doesn't start to ring until the first agents pick up his queued call. I want the second call to start ringing on the second agents phone right away, since he's available. Here's the output from the queue from the CLI: [...] Has anyone seen this problem before or have a solution on it? Is it possible somehow to tell Asterisk to only send one queue'd call to the Agent at the time? Did you set autofill=yes in queues.conf? Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reliable wireless SIP phones
On Thu, 28 Aug 2008, Michael Graves wrote: I've had essentially no problems with my snom m3s. Someone from snom has been in touch to confirm that they are now putting more effort into the firmware for this phone. There are a few new features that I'd like to see that are already in their plans. One feature I hope they add is a way to use a different outgoung accout on a handset - right now, it's a very fiddly process to change outgoing accounts involving entering the PIN! If only it were like the Siemens where you dial, push green and hold, then select... And I am finding them a bit plasticy compared to the Siemens units. (Which is bizarre as their desk phones feel much more robust!) Proprietary batteries too. (Siemens use AAA's) Ah well - if that's what the customer wants then that's what they'll get... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Console softphone
Hello! Tanks a lot for the first hints. I chose console dial and originate for a start, but I'm open to test the browser phone alternative as well. Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] remove queue call
Hmm I would rather tab complete than copy and paste. It might do some sort of escaping. Is there no time of day when all the queues are empty? S On 29 Aug 2008, at 07:14, Rilawich Ango wrote: Yup I just copy and paste to it but it shown not a known channel. On Thu, Aug 28, 2008 at 6:47 PM, Steven Howes [EMAIL PROTECTED] wrote: Did you tab complete it to make sure it was right? On 28 Aug 2008, at 11:39, Rilawich Ango wrote: I got the message below after I issue the soft hangup. sip01*CLI soft hangup Local/[EMAIL PROTECTED],2 Local/[EMAIL PROTECTED],2 is not a known channel Any other way to kill the call without affecting other queues and calls? On Thu, Aug 28, 2008 at 4:09 PM, Steven Howes [EMAIL PROTECTED] wrote: Try CLI soft hangup Local. On 28 Aug 2008, at 09:01, Rilawich Ango wrote: Hi , Actually, there are 3 queues in the server. Only one queue (2700) has problem. I want to reset or remove the caller only in 2700 without affecting other queues or calls. Does it work for my case? On Thu, Aug 28, 2008 at 11:49 AM, Andy Kuo [EMAIL PROTECTED] wrote: Hi, Try CLI soft hangup Local. Andy On 8/27/08, Rilawich Ango [EMAIL PROTECTED] wrote: Hi all, I have the following queue and members. I found that there is a call stuck in the queue so other call can't enter the queue. I want to know whether we can remove the call (by CLI) to free the queue. ango 2700 has 1 calls (max unlimited) in 'rrmemory' strategy (35s holdtime), W:0, C:134, A:48, SL:88.8% within 120s Members: Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken no calls yet Callers: 1. Local/[EMAIL PROTECTED],2 (wait: 113:19, prio: 0) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api- digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api- digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip conversations overlapping!!!!
Every one PSTN line connected to the FXS port of sipura.. Though these 4 lines comes in one cable if that has to do with anything! Not clear for me, develop some more you topology. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip conversations overlapping!!!!
i appologize for not making myself clear.. i have my asterisk box, connexted to 4 sipura3102.. these sipuras has 4 PSTN lines connected to them through one cable, which has 8 lines inside of it (2 connected to an RJ11 and plugged into its respecitve fxs port in the sipura) on the other side, i have 20 softphones.. these softphones has asterisk as their gateway.. where they could call eachother! or call/recieve calls through any of the sipuras... my prob is as such: when i call from softphone#1 to sipura #1, sound is pretty good and everything is working perfectly.. though if asterisk recieves a call from another sipura.. lets say its sipura #2, then! i could hear the attendnat answering the incoming phone in my current conversation, and i could hear some1 picking up and answerinfg the call..! if i ask them to hang up! my line breaks as well.. Date: Fri, 29 Aug 2008 10:40:57 +0200 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] sip conversations overlapping Every one PSTN line connected to the FXS port of sipura.. Though these 4 lines comes in one cable if that has to do with anything! Not clear for me, develop some more you topology. _ News, entertainment and everything you care about at Live.com. Get it now! http://www.live.com/getstarted.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk cdr_mysql inexact values
I have a simple cdr configured with the default tables, here is a row of a
good cdr report
calldate | clid | src |
dst | dcontext | channel | ect . ect
2008-08-29 10:16:49 | C. BOUTON 40 | 40 | XXX | phonesystems |
SIP/40-08776938 | ect . ect
I have replaced the number by XX, but it is there. But sometimes
i get this:
calldate | clid | src |
dst | dcontext | channel | ect . ect
2008-08-29 10:17:06 | C. SAGNIER 60 | 60 | s |
phonesystems | SIP/111-08799690 | ect . ect
You see that s in dst ? I know from where it is coming but i have no idea
how to remove it. I am using one macro for dial out, it is easy for me to
manage multiple outgoing peers and max channels for them. I am using
spriority inside that macro, so somehow cdr SOMETIMES report
s as dst. If you can help me to arange my macro to remove that s from cdr or
by any advice i would be gratefull. My macro:
[macro-phonesystems]
exten = s,1,NoOp(We are calling=${ARG1})
exten = s,2,GotoIf($[${GROUP_COUNT(ph0)}=1]?100:3)
exten = s,3,Set(GROUP()=ph0)
exten = s,4,Dial(Sip/${ARG1:[EMAIL PROTECTED],40,TwW)
exten = s,5,NoOP(PH0)
exten = s,100,GotoIf($[${GROUP_COUNT(ph1)}=1]?200:101)
exten = s,101,Set(GROUP()=ph1)
exten = s,102,Dial(Sip/${ARG1:[EMAIL PROTECTED],40,Tw)
exten = s,103,NoOp(PH1)
exten = s,200,GotoIf($[${GROUP_COUNT(ph2)}=2]?300:201)
exten = s,201,Set(GROUP()=ph2)
exten = s,202,Dial(Sip/${ARG1:[EMAIL PROTECTED],40,Tw)
exten = s,203,NoOp(PH2)
exten = s,300,GotoIf($[${GROUP_COUNT(ph3)}=2]?400:301)
exten = s,301,Set(GROUP()=ph3)
exten = s,302,Dial(Sip/${ARG1:[EMAIL PROTECTED],40,Tw)
exten = s,303,NoOp(PH3)
exten = s,400,GotoIf($[${GROUP_COUNT(ph4)}=2]?400:500)
exten = s,401,Set(GROUP()=ph4)
exten = s,402,Dial(Sip/${ARG1:[EMAIL PROTECTED],40,Tw)
exten = s,403,NoOp(PH4)
exten = s,500,Playback(all-circuits-busy-now)
And my portion of extensions.conf from where we are jumping to that macro
exten =
_00[123459]!,1,Monitor(gsm,${CALLERID(num)}APP-${EXTEN}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
exten = _00[123459]!,2,GotoIf($[${DB(internet/disponible)}=1]?3:7)
exten = _00[123459]!,3,GotoIf($[${DB(moyende/telecom)}=0]?4:6)
exten = _00[123459]!,4,Macro(phonesystems,${EXTEN})
exten = _00[123459]!,5,Hangup()
;this hangup is for marcro returning
exten = _00[123459]!,6,GotoIf($[${DB(moyende/telecom)}=1]?7:8)
;case 8 should never happen, just in case.
exten = _00[123459]!,7,Dial(mISDN/g:intern-out/${EXTEN:1})
exten = _00[123459]!,8,Dial(mISDN/g:intern-out/${EXTEN:1})
Thank you.
___
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Re: [asterisk-users] sip conversations overlapping!!!!
Remove pstn lines from sipura and call sipura to sipura ... any problems ? Still with pstn lines removed call sipura1 sipura2 and after sipura 3sipura1 do you still hear any voices? if not it's you cable to pstn. Give us feedback ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with DTMF on IVRs
I tried DTMFmode=auto and it did not help. Any further ideas? -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reliable wireless SIP phones
On Fri, 29 Aug 2008 07:40:18 +0100 (BST), Gordon Henderson wrote: One feature I hope they add is a way to use a different outgoung accout on a handset - right now, it's a very fiddly process to change outgoing accounts involving entering the PIN! I have a little video of the process of changing the active voip account for outbound dialing. I just posted it to my blog this morning. http://blog.mgraves.org. If only it were like the Siemens where you dial, push green and hold, then select... And I am finding them a bit plasticy compared to the Siemens units. (Which is bizarre as their desk phones feel much more robust!) Proprietary batteries too. (Siemens use AAA's) Ah well - if that's what the customer wants then that's what they'll get... Yeah, but the m3 doesn't have any of the keyboard lag that the Siemens phones suffer. When entering a number of the S685IP the phone seems very slow to acknowledge each keypress. My only gipe is that lack of an uploadable contact list, which I'm told will be in the firmware update planned in September. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Tips and Tricks: Dynamic Subroutines inAGI
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Darren Sessions
Sent: Thursday, August 28, 2008 10:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk Tips and Tricks: Dynamic
Subroutines inAGI
...
The hurdle in doing something like this was how to
dynamically execute
a subroutine from the results of the database query which
were dumped
into a variable. The method I used with the subroutine reference
doesn’t allow for arguments to be passed (if anyone finds / knows a
way to do this, let me know), so I use global variables.
This is a simple example of dynamic subroutine execution
(without the
database query):
use strict;
use warnings;
our $called_number;
our $calling_number;
sub run_me {
$AGI-verbose(”Called Number = “.$called_number, 1);
$AGI-verbose(”Calling Number = “.$calling_number, 1);
}
sub set_variables {
$called_number = “8005551212″;
$calling_number = “300222″;
}
sub dynamic_execute {
my ($sub) = @_;
if (!$sub) {
$AGI-verbose(”No subroutine name passed!!”, 1);
return(-1);
}
my $exec = \{$sub};
return($exec-());
}
set_variables();
dynamic_execute(”run_me”);
If you don't mind disabling strict refs (no strict 'refs';), you could easily
do this.
This would allow you to use something like: $sub($argument1, $argument2);
The only other way I can think of (though I have not tried it) would be to
populate a hash with subroutine refs and use the string as the index into it.
Something like this:
#!/usr/bin/perl
use strict;
use warnings;
sub print_ref { print @_; };
my %sub_hash = (print_ref, \print_ref);
sub print_stuff {
my $sub = shift;
my $string = shift;
$sub($string);
}
print_stuff($sub_hash{print_ref}, This is printed.\n);
The first idea uses the symbol table directly, and the second one essentially
is building your own symbol table.
Hope that helps,
- Brad
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Re: [asterisk-users] Reliable wireless SIP phones
I've had essentially no problems with my snom m3s. Someone from snom has been in touch to confirm that they are now putting more effort into the firmware for this phone. There are a few new features that I'd like to see that are already in their plans. When I saw held this phone at Astricon 2007, I was impressed with the price/value ratio, but I was frankly a bit turned off by how 'creaky' it felt. If I held it between my ear and shoulder I'd hear the dreaded chintzy plastic creaking sound. I think the one I saw may have been a pre-production prototype. Does your Snom M3 have a cheap plastic creaky build? Does it feel 'chintzy' to you? -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reliable wireless SIP phones
On Fri, 29 Aug 2008, Michael Graves wrote: On Fri, 29 Aug 2008 07:40:18 +0100 (BST), Gordon Henderson wrote: One feature I hope they add is a way to use a different outgoung accout on a handset - right now, it's a very fiddly process to change outgoing accounts involving entering the PIN! I have a little video of the process of changing the active voip account for outbound dialing. I just posted it to my blog this morning. http://blog.mgraves.org. Yup. That's the way to do it. 9 button presses before you get to select. So.. Try getting a mom pop accountancy company to go through having to push 9 key-preses before they get to select the account they want to use for outgoing calls (business or home on their case) THEN remembering to switch it back again. (Their response was Fuck that, can't you make it easier?) Bzzt. So Sorry Mr. Snom. Does not compute is my answer to that. (so since they still liked the Snoms otherwise, my solution is to get them to dial a star at the end of a number to select their 'home' account, otherwise it goes out on their work account and the dialplan fixes it up) Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reliable wireless SIP phones
On Fri, 29 Aug 2008, Karl Fife wrote: I've had essentially no problems with my snom m3s. Someone from snom has been in touch to confirm that they are now putting more effort into the firmware for this phone. There are a few new features that I'd like to see that are already in their plans. When I saw held this phone at Astricon 2007, I was impressed with the price/value ratio, but I was frankly a bit turned off by how 'creaky' it felt. If I held it between my ear and shoulder I'd hear the dreaded chintzy plastic creaking sound. I think the one I saw may have been a pre-production prototype. Does your Snom M3 have a cheap plastic creaky build? Does it feel 'chintzy' to you? Yes. Yes. It's strictly in the Grandstream class of phones as far as build quality seems to go. (without the good price that Grandstreams are!!!) Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Console softphone
On Thursday 28 August 2008 19:59:55 Lee, John (Sydney) wrote: Better still - is it possible to SSH (or some sort of connection method) from a remote PC to the Asterisk server and make a call using CLI? Sure, you can use the CLI 'console dial' command. Do you mean that I will be able to hear the call from my PC if I do 'console dial' on the remote Asterisk server provided that I install browser softphone on the server? No, the 'console dial' command works strictly with the soundcard on the server itself. If you want remote origination use, use the CLI command 'originate' for that. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Tips and Tricks: Dynamic Subroutines inAGI
Impressive work Bradley! I tested it and it worked great, even with my
mandatory 'use strict'.
Thanks,
- Darren
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug 29, 2008, at 5:47 AM, Watkins, Bradley wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Darren Sessions
Sent: Thursday, August 28, 2008 10:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk Tips and Tricks: Dynamic
Subroutines inAGI
...
The hurdle in doing something like this was how to
dynamically execute
a subroutine from the results of the database query which
were dumped
into a variable. The method I used with the subroutine reference
doesn’t allow for arguments to be passed (if anyone finds /
knows a
way to do this, let me know), so I use global variables.
This is a simple example of dynamic subroutine execution
(without the
database query):
use strict;
use warnings;
our $called_number;
our $calling_number;
sub run_me {
$AGI-verbose(”Called Number = “.$called_number, 1);
$AGI-verbose(”Calling Number = “.$calling_number,
1);
}
sub set_variables {
$called_number = “8005551212″;
$calling_number = “300222″;
}
sub dynamic_execute {
my ($sub) = @_;
if (!$sub) {
$AGI-verbose(”No subroutine name passed!!”, 1);
return(-1);
}
my $exec = \{$sub};
return($exec-());
}
set_variables();
dynamic_execute(”run_me”);
If you don't mind disabling strict refs (no strict 'refs';), you
could easily do this.
This would allow you to use something like: $sub($argument1,
$argument2);
The only other way I can think of (though I have not tried it) would
be to populate a hash with subroutine refs and use the string as the
index into it.
Something like this:
#!/usr/bin/perl
use strict;
use warnings;
sub print_ref { print @_; };
my %sub_hash = (print_ref, \print_ref);
sub print_stuff {
my $sub = shift;
my $string = shift;
$sub($string);
}
print_stuff($sub_hash{print_ref}, This is printed.\n);
The first idea uses the symbol table directly, and the second one
essentially is building your own symbol table.
Hope that helps,
- Brad
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Re: [asterisk-users] Reliable wireless SIP phones
On Aug 29, 2008, at 8:18 AM, Karl Fife wrote: I've had essentially no problems with my snom m3s. Someone from snom has been in touch to confirm that they are now putting more effort into the firmware for this phone. There are a few new features that I'd like to see that are already in their plans. When I saw held this phone at Astricon 2007, I was impressed with the price/value ratio, but I was frankly a bit turned off by how 'creaky' it felt. If I held it between my ear and shoulder I'd hear the dreaded chintzy plastic creaking sound. I think the one I saw may have been a pre-production prototype. Does your Snom M3 have a cheap plastic creaky build? Does it feel 'chintzy' to you? -Karl I haven't justified getting a DECT sip device yet. I did plug in a few DECT devices to my SIP ATA's which work nice (and cheap). The WIP330 came in and so far I'm really liking it. I really like being able to grab a wifi connection and reg the extension. smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Queue's
Tobias Ahlander wrote: Hello Philipp, Yes, I have autofill set in queues.conf. I suspect that this behaviour is because the Polycom phones I use have 2 lines. Has anyone used this function with polycom phones before? Also, my agents are Dynamic, perhaps this works better with Static agents? Here's my queues.conf (with commented lines deleted for easier reading): [general] persistentmembers = yes autofill = yes monitor-type = MixMonitor [sales] strategy = rrmemory wrapuptime=15 Depending on which Asterisk version you are using, there was a bug in the queue application for some 1.4 releases where the autofill option would only be set properly if it were placed inside a queue. In other words, you may want to try putting autofill=yes inside the [sales] queue in your configuration. Also, if you're using a version of Asterisk 1.2, autofill is not a valid option and you'll be stuck with the behavior you're seeing. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reliable wireless SIP phones
Someone did report to me that they had difficulty getting MWI working on the snoms. I've not been able to test this with Asterisk yet. Since i rely on MWI from my desk phone (Polycom IP650) MWI on the cordless is not setup, and I don't let calls to the cordless go to VM right now. Michael, You other may already know this, but you can make the MWI on your Snom cordless sync the MWI status of the mailbox belonging to your Polycom Desk phone, not it's own mailbox. In other words, don't hesitate to use the Snom MWI, just because you don't need another mailbox The cordless will indicate the MWI status of your desk phone's mailbox. Here's how I did it. 1. Put the mailbox I wanted to subscribe to in the sip.conf context belonging to the device. ALA: [205] ; ; Cordless ; type=friend username=205 secret=[sh] host=dynamic context=lc-route [EMAIL PROTECTED];subscribing to mailbox 202 NOT x205 2. Configure the Snom to mailbox 202 even though it's x205 Do this under Identity(n), Login, Mailbox Viola! -Karl On Fri, 29 Aug 2008 06:34:02 -0500, Michael Graves [EMAIL PROTECTED] said: On Fri, 29 Aug 2008 07:40:18 +0100 (BST), Gordon Henderson wrote: One feature I hope they add is a way to use a different outgoung accout on a handset - right now, it's a very fiddly process to change outgoing accounts involving entering the PIN! I have a little video of the process of changing the active voip account for outbound dialing. I just posted it to my blog this morning. http://blog.mgraves.org. If only it were like the Siemens where you dial, push green and hold, then select... And I am finding them a bit plasticy compared to the Siemens units. (Which is bizarre as their desk phones feel much more robust!) Proprietary batteries too. (Siemens use AAA's) Ah well - if that's what the customer wants then that's what they'll get... Yeah, but the m3 doesn't have any of the keyboard lag that the Siemens phones suffer. When entering a number of the S685IP the phone seems very slow to acknowledge each keypress. My only gipe is that lack of an uploadable contact list, which I'm told will be in the firmware update planned in September. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip conversations overlapping!!!!
RoLaNd RoLaNd wrote: i appologize for not making myself clear.. i have my asterisk box, connexted to 4 sipura3102.. these sipuras has 4 PSTN lines connected to them through one cable, which has 8 lines inside of it (2 connected to an RJ11 and plugged into its respecitve fxs port in the sipura) Lines plugged into the fxS ports? I hope you have them in the LINE ports (fxO) Are there any telephones plugged directly into the Sipuras? Into the PHONE ports (FXS) my prob is as such: when i call from softphone#1 to sipura #1, sound is pretty good and everything is working perfectly.. though if asterisk recieves a call from another sipura.. lets say its sipura #2, then! i could hear the attendnat answering the incoming phone in my current conversation, and i could hear some1 picking up and answerinfg the call..! if i ask them to hang up! my line breaks as well.. I would double check the wiring of the 8 line cable. 4 POTS lines = 4 pairs of tip ring. Are there some of the tipring pairs mixed up? eg tip from line 1 mixed with ring from line 2, etc. This is the most likely scenario since I can't imagine Asterisk bridging the calls without being asked to. Otherwise, are we still missing something in the topology here? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reliable wireless SIP phones
So.. Try getting a mom pop accountancy company to go through having to push 9 key-preses before they get to select the account they want to use for outgoing calls (business or home on their case) THEN remembering to switch it back again. (Their response was Fuck that, can't you make it easier?) LOL to your customer response :-) Sometimes I wonder: Do the people who design these things actually use telephones? Hopefully a hardware makers will use Astricon 2008 to announce some new wireless endpoints. There are some really nice SIP-DECT offerings out there now, but I think they're quite spendy, even for coorporate budgets. You'd really have to have a very specific vertical wireless application to get an decent ROI on that kind of cash outlay. -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Wi-SIP vs. SIP-DECT
Anybody care to muse on Wi-SIP vs. SIP-DECT? My limited research indicates that none of the WiSip phones will ever be able to match the performance of DECT phones. Maybe I'm wrong but a Wi-SIP phone seems like a DIESEL sports car. There is nothing wrong with the technology, but it seems like a shoe-horned fit into the requirements of a wireless endpoint. DECT uses a wireless radio layer that was engineered from the ground-up with the design priorities of a wireless endpoit. I notice that the standby times of Wi-SIP vs. SIP-DECT are a great illustration of this point. I guess there's no low-power way to participate in a WiFi network, hense standby battery life that sucks in Wi-SIP. I've never actually demoed a Wi-SIP phone on premesis, but if the range of my WiFi LAPTOP vs. my DECT 6.0 headset is any indication, (DECT more than double the range) I'd guess it to be quite hard to make a case for Wi-SIP unless you're doing some straight-up network application integration right onto the phone. Can anyone speak to this? -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] track 1.6 progress
I have an upcoming need for SIP over TCP which is part of the 1.6 feature set. What is the best way to find out a rough idea as to when 1.6 will go out of beta into production? Is there a user group (svn perhaps) for tracking 1.6 stability and progress? thank you, jim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Faxing through Zap cards
I think I have this straight, but I wanted to bounce it off anyone who might be more knowledgeable. We are installing an Asterisk server at a location that only has PRI. Inbound fax comes in on the PRI with its own DID. Currently, the PBX handling it just has a PRI port and sends calls for that DID to an FXS port that the fax machine is connected to. My plan was to use a Digium TE card for the PRI and a TDM card with an FXS port to connect the fax to. I know I'll need to use the fax detection feature to disable echo cancellation. I can't use the Zaptel dacs features because I won't know ahead of time which channel the fax calls will come in on (though I could potentially use it for outbound faxes). So the call path would look like this: PSTN --- PRI --- TExxx --- Asterisk --- TDMxxx FXS port --- fax machine Does this sound reasonable? Any gotchas I should watch out for? -James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing through Zap cards
James Sneeringer wrote: We are installing an Asterisk server at a location that only has PRI. Inbound fax comes in on the PRI with its own DID. Currently, the PBX handling it just has a PRI port and sends calls for that DID to an FXS port that the fax machine is connected to. This will work fine My plan was to use a Digium TE card for the PRI and a TDM card with an FXS port to connect the fax to. I know I'll need to use the fax detection feature to disable echo cancellation. No, Asterisk will detect the tones on it's own and disable EC. PSTN --- PRI --- TExxx --- Asterisk --- TDMxxx FXS port --- fax machine Does this sound reasonable? Any gotchas I should watch out for? Yes. You may also want to look into HylaFAX+, it's a wonderful piece of software that will allow you to have more control over your faxes. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connecting two asterisks via IAX
Hi, I need to connect 2 asterisks in 2 different countries (A and B) for one company so it's possible to make connections between the 2 offices. For connectivity reasons (NAT traversal) i want to connect the 2 asterisk with IAX so that when a user on office A connects via SIP to user on office B the call is going trought IAX channel. Can anyone give me an ideia how to accomplish this? the schema: (via SIP)(via IAX)(via SIP) Office A - Asterisk A --- Asterisk B - Office B Thanks in advance, Nuno ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing through Zap cards
On Fri, 29 Aug 2008, Doug Lytle wrote: James Sneeringer wrote: We are installing an Asterisk server at a location that only has PRI. Inbound fax comes in on the PRI with its own DID. Currently, the PBX handling it just has a PRI port and sends calls for that DID to an FXS port that the fax machine is connected to. This will work fine Just make sure you get the IRQs separated and you might want to think about a TDM410 rather than a TDM400 card. It did not work for me when I tried it with a TDM400 card - the PRI card would lose interrupts and eventually reset, dropping all calls. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wi-SIP vs. SIP-DECT
On Friday 29 August 2008 09:58:56 Karl Fife wrote: Anybody care to muse on Wi-SIP vs. SIP-DECT? My limited research indicates that none of the WiSip phones will ever be able to match the performance of DECT phones. Maybe I'm wrong but a Wi-SIP phone seems like a DIESEL sports car. There is nothing wrong with the technology, but it seems like a shoe-horned fit into the requirements of a wireless endpoint. DECT uses a wireless radio layer that was engineered from the ground-up with the design priorities of a wireless endpoit. I notice that the standby times of Wi-SIP vs. SIP-DECT are a great illustration of this point. I guess there's no low-power way to participate in a WiFi network, hense standby battery life that sucks in Wi-SIP. I've never actually demoed a Wi-SIP phone on premesis, but if the range of my WiFi LAPTOP vs. my DECT 6.0 headset is any indication, (DECT more than double the range) I'd guess it to be quite hard to make a case for Wi-SIP unless you're doing some straight-up network application integration right onto the phone. Can anyone speak to this? I think the primary reason for going Wi-SIP is the buildout factor. Yes, while range is limited in WiSIP, the fact that the phone is entirely self-contained means that you can build out additional WAPs, transitioning between them as the phone moves around an area. Additionally, an existing wireless infrastructure can be taken advantage of, instead of building a separate network for the phones. While DECT repeaters exist that serve this same purpose, the tools to ensure that all areas within a service area are served are still a little lacking. Basically, you're left with deploying stations, running around with a phone to every nook and cranny, hoping the battery life stands up, and deploying repeaters in a haphazard fashion to address the issues. Compare that to the professional tools you can find to fully deploy Wifi hotspots on the first try, and you'll find a much less painful deployment cycle. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wi-SIP vs. SIP-DECT
Karl Fife wrote: Anybody care to muse on Wi-SIP vs. SIP-DECT? My limited research indicates that none of the WiSip phones will ever be able to match the performance of DECT phones. Maybe I'm wrong but a Wi-SIP phone seems like a DIESEL sports car. Just for fun! http://www.cumminsracing.com/ Rod -- There is nothing wrong with the technology, but it seems like a shoe-horned fit into the requirements of a wireless endpoint. DECT uses a wireless radio layer that was engineered from the ground-up with the design priorities of a wireless endpoit. I notice that the standby times of Wi-SIP vs. SIP-DECT are a great illustration of this point. I guess there's no low-power way to participate in a WiFi network, hense standby battery life that sucks in Wi-SIP. I've never actually demoed a Wi-SIP phone on premesis, but if the range of my WiFi LAPTOP vs. my DECT 6.0 headset is any indication, (DECT more than double the range) I'd guess it to be quite hard to make a case for Wi-SIP unless you're doing some straight-up network application integration right onto the phone. Can anyone speak to this? -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_mobile
I am trying to test the chan_mobile functionality. i have not been able to compile the SVN version. I get an error about needing ncurses when doing make menuselect. I have verified that I have libncurses5-dev installed. I have complied version 1.6.0-beta9 w/ asterisk-addons-1.6.0-beta4 with success, but encountered an issue when trying to pair the device. It will not accept the PIN. I am using linux distro: Debian unstable 2.6.26-1-686 #1 SMP any recommendations? Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wi-SIP vs. SIP-DECT
On Fri, 29 Aug 2008, Karl Fife wrote: Anybody care to muse on Wi-SIP vs. SIP-DECT? My limited research indicates that none of the WiSip phones will ever be able to match the performance of DECT phones. Maybe I'm wrong but a Wi-SIP phone seems like a DIESEL sports car. There is nothing wrong with the technology, but it seems like a shoe-horned fit into the requirements of a wireless endpoint. DECT uses a wireless radio layer that was engineered from the ground-up with the design priorities of a wireless endpoit. I notice that the standby times of Wi-SIP vs. SIP-DECT are a great illustration of this point. I guess there's no low-power way to participate in a WiFi network, hense standby battery life that sucks in Wi-SIP. I've never actually demoed a Wi-SIP phone on premesis, but if the range of my WiFi LAPTOP vs. my DECT 6.0 headset is any indication, (DECT more than double the range) I'd guess it to be quite hard to make a case for Wi-SIP unless you're doing some straight-up network application integration right onto the phone. Can anyone speak to this? I've used both - with good results. However, I've also done a *lot* of network building using Wi-Fi and it's not that good for telephony. Firstly it's half duplex (so's DECT, GSM, etc. but that's OK, as it's designed that way), and what I've found, certinly in the consumer and some of the access point aimed at businesses is that the radio turn-around time sometimes becomes significant. In tests, I found that most units would degrade horribly when the packet size was 140 bytes or so. VoIP packets are 160 bytes, so we're mostly OK there. The packet size and frequency (50 packets a second, both ways) is the biggest killer for access points. They're really optimised for streaming data one way, so big 1500 byte packet down, tiny ACK packet back. Intersperse that with VoIP and you get issues. Even with fancy access point that have traffic management, sending just one big 1500 byte packet can have an impact on a stream on 160 byte packets that need to be sent at a specific rate. So you'll get away with it on your home network (or small office) if you're the only one using Wi-Fi. I make calls with my Nokia E90 + SIP Wi-Fi and for the most part, it's fine. But get someone else on the same access point and have them do some file-bashing to a local server and you'll get issues. If you can go to the expense of running a totally separate Wi-Fi network just for VoIP then you'll probably be fine, but my old AP barely copes with 2 concurrent calls. 3 calls and you start to get loss. Then you've got the issues of channel separation. There are only 3 true clear channels in the spectrum that don't overlap. I can see 5 other access points from my office now (and the town CCTV system runs in the 2.4GHz band too )-: and I'm in a rural location, so what hope is there in a busy office complex... So Wi-Fi works, but only just... DECT with repeaters seems so much better... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing through Zap cards
On Fri, Aug 29, 2008 at 10:31 AM, Doug Lytle [EMAIL PROTECTED] wrote: Yes. You may also want to look into HylaFAX+, it's a wonderful piece of software that will allow you to have more control over your faxes. Thanks for the input, Doug. Moving away from standalone fax machines to a fax server, such as HylaFAX, would certainly be nice as well. It's more of a logistical problem for us, in that some of our offices just don't have document scanners, and they frequently have to fax paper documents. -James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing through Zap cards
I would suggest that you go with a dual port T1 card and a channel bank. you will have much greater fleibality as well as more stable connection. On 8/29/08, Gordon Henderson [EMAIL PROTECTED] wrote: On Fri, 29 Aug 2008, Doug Lytle wrote: James Sneeringer wrote: We are installing an Asterisk server at a location that only has PRI. Inbound fax comes in on the PRI with its own DID. Currently, the PBX handling it just has a PRI port and sends calls for that DID to an FXS port that the fax machine is connected to. This will work fine Just make sure you get the IRQs separated and you might want to think about a TDM410 rather than a TDM400 card. It did not work for me when I tried it with a TDM400 card - the PRI card would lose interrupts and eventually reset, dropping all calls. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing through Zap cards
On Fri, Aug 29, 2008 at 10:47 AM, Gordon Henderson [EMAIL PROTECTED] wrote: Just make sure you get the IRQs separated and you might want to think about a TDM410 rather than a TDM400 card. Thanks, I'll keep that in mind. We're buying new hardware for this, so we can spec it out accordingly. -James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reliable wireless SIP phones
At 05:48 AM 8/29/2008, you wrote: (so since they still liked the Snoms otherwise, my solution is to get them to dial a star at the end of a number to select their 'home' account, otherwise it goes out on their work account and the dialplan fixes it up) We have 5 outgoing numbers we want to use selectively and we just dial 1,2,3,4 or 6 first which picks the proper rules. If you forget it asks which line you want to use and 1 is the line we want guests to use so it all works fine if you don't know what's going on. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Issue when dialing multiple extensions using ------Please Help
Hi, I have a simple dialplan. [test] exten = _X.,1,Dial(SIP/1000SIP/1002) I have registered user whose context is test. Now I am dialing any number, so it will enter into test context. It will dial 1000 1002 both. Both keeps ringing. Now the problem is, when any of them answer, another one keeps ringing for 10 to 15 sec. Please help me , what's wrong here. Thanks, Krunal Patel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] music on hold is not working
Hi, I have made class for MOH uploaded a mp3 file to the folder. Now I am using this class for music on hold during dialing. Now when call has been established, I put the other end on hold. So from that end I should listen uploaded file. But I am not getting audio. Please help me. Thanks, Krunal Patel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing through Zap cards
James Sneeringer wrote: Thanks for the input, Doug. Moving away from standalone fax machines to a fax server, such as HylaFAX, would certainly be nice as well. It's more of a logistical problem for us, in that some of our offices We don't either, we use HylaFAX+ to augment the fax machines.For example in bound: PRI = Asterisk = HylaFAX+ (PDF/Email/Fax machine/Document Archive) Example Out bound: (Fax Machine/PC) = HylaFAX+ (Email/PRI) =PSTN=Fax machine Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing through Zap cards
C F wrote: I would suggest that you go with a dual port T1 card and a channel bank. you will have much greater fleibality as well as more stable connection. This is what we do. Along with an ADIT 600 (eBay special) Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reliable wireless SIP phones
On Fri, Aug 29, 2008 at 09:56:36AM -0500, Karl Fife wrote: So.. Try getting a mom pop accountancy company to go through having to push 9 key-preses before they get to select the account they want to use for outgoing calls (business or home on their case) THEN remembering to switch it back again. (Their response was Fuck that, can't you make it easier?) LOL to your customer response :-) Sometimes I wonder: Do the people who design these things actually use telephones? And it makes me ask: is there any free software hardware phone? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting two asterisks via IAX
Easy.
Just create a peer in each office that connects to the other, basic example
on server 1.
iax.conf
[office2]
type=friend
host=office2
disallow=all
allow=ulaw
context=internal_office_dialing
username=office1
secret=mypassword
trunk=yes
Create a peer on the office2 server to point back at office1 in the same
way.
extensions.conf on office1.
exten = _2XX,1,Dial(IAX2/office2/${EXTEN})
This is a simple setup where office1 has the 1XX extension block and office2
has the 2XX block.
Tom
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nuno Marques
Sent: Friday, August 29, 2008 11:36 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Connecting two asterisks via IAX
Hi,
I need to connect 2 asterisks in 2 different countries (A and B) for one
company so it's possible to make connections between the 2 offices.
For connectivity reasons (NAT traversal) i want to connect the 2 asterisk
with IAX so that when a user on office A connects via SIP to user on office
B the call is going trought IAX channel.
Can anyone give me an ideia how to accomplish this?
the schema:
(via SIP)(via IAX)
(via SIP)
Office A - Asterisk A --- Asterisk B
- Office B
Thanks in advance,
Nuno
___
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Re: [asterisk-users] Reliable wireless SIP phones
Am Freitag, den 29.08.2008, 09:16 -0700 schrieb Ira: At 05:48 AM 8/29/2008, you wrote: (so since they still liked the Snoms otherwise, my solution is to get them to dial a star at the end of a number to select their 'home' account, otherwise it goes out on their work account and the dialplan fixes it up) We have 5 outgoing numbers we want to use selectively and we just dial 1,2,3,4 or 6 first which picks the proper rules. If you forget it asks which line you want to use and 1 is the line we want guests to use so it all works fine if you don't know what's going on. Hi, on German landlines the user can pick a carrier on a per-call basis (at least if it is a T-Com line). The national dialplan assigns 010NX and 0100NX prefixes for that purpose. I believe similar access codes are in operation in US (10-10-XXX?), and possibly your system blocks those for similar reasons as we do (they are useless on VoIP lines here). So those codes can be used for selecting a non-default line, where I use 01099 to send out a call with my mobile phone Caller ID, 01090 for anonymous no-callerid calling and 010[123][123] to select between three providers with up to three outgoing Caller ID numbers each (and a few others in the 010[4-8]X range). Not pre-pending such a number will make a reasonable default choice here depending on the phone used. If the percentage of non-default line calls is fair below 20% such a long prefix is still better than dialing a digit before each number (with the additional risk of forgetting that, or dialing that digit when calling from other places :-) This assures that guests have no trouble using the telephone, as they just dial the phone number as they would from their own telephone at home. If I want them to not have my caller ID on the callee display (no call back, privacy, whatever) I simply tell them to dial the 01090 first - they will assume I want them to use a certain carrier, and not need any additional instructions. Call-By-Call, being the German name for it, is widely known. Another aspect of carrier selection codes opposed to switching lines on the phone is that it is independant of the phone hardware at hand, be it an analogue + adapter, ISDN + adapter, DECT, SIP hardphone, software... Of course it is not necessary to allow all prefixes from all phones, or have the same meaning of a prefix on all phones (01099 here would be an example that sends a different Caller ID from different phones, depending of the mobile phone number of the person usually using that phone). Best regards, Anselm smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfers on AgentLogin()
So, no answers or is this thread going to remain unanswered too? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: August 28, 2008 6:15 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Transfers on AgentLogin() Oh, by the way, the agent who will be doing the assisted transfer will be using eyebeam. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: August 28, 2008 5:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Transfers on AgentLogin() Hi, I have the same question as: http://lists.digium.com/pipermail/asterisk-dev/2003-November/002320.html ..which like all important things was never answered. How do I do an assisted transfer on AgentLogin()? I don't use zaptel, it's just pure SIP/VoIP. Help please. Thanks, Mark. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call monitor/barge/train
Hi, I'm planning on migrating someone who uses a very mature system. They would be logging in either as AgentLogin() or AQM. The main requirement however, is: The supervisor will have a control panel, where he will see how many of his agents are on call. If they are, he can right-click on the agent and get the options Call Monitor (where the super just listens in on the call, or new reps can listenin), Call Train (where the super and agent can talk to each other for training, but the customer doesn't hear them, or older reps can train newer reps), Call Barge (where everyone can hear everyone else, super agent and caller) How or what can facilitate this? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing through Zap cards
On Fri, Aug 29, 2008 at 11:06 AM, C F [EMAIL PROTECTED] wrote: I would suggest that you go with a dual port T1 card and a channel bank. you will have much greater fleibality as well as more stable connection. Thanks for the tip. If faxing in this setup does prove to be unreliable, this is a good backup. -James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call monitor/barge/train
Mark Hamilton wrote: Hi, I’m planning on migrating someone who uses a very mature system. They would be logging in either as AgentLogin() or AQM. The main requirement however, is: The supervisor will have a control panel, where he will see how many of his agents are on call. If they are, he can “right-click” on the agent and get the options Call Monitor (where the super just listens in on the call, or new reps can listenin), Call Train (where the super and agent can talk to each other for training, but the customer doesn’t hear them, or older reps can train newer reps), Call Barge (where everyone can hear everyone else, super agent and caller) How or what can facilitate this? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users looking at the command Chanspy should give you a lot of relevant information. http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy -Robin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting two asterisks via IAX
Thanks Tom
I'm going to try that.
Regards
Nuno
2008/8/29 Tom Moore [EMAIL PROTECTED]
Easy.
Just create a peer in each office that connects to the other, basic example
on server 1.
iax.conf
[office2]
type=friend
host=office2
disallow=all
allow=ulaw
context=internal_office_dialing
username=office1
secret=mypassword
trunk=yes
Create a peer on the office2 server to point back at office1 in the same
way.
extensions.conf on office1.
exten = _2XX,1,Dial(IAX2/office2/${EXTEN})
This is a simple setup where office1 has the 1XX extension block and
office2 has the 2XX block.
Tom
--
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Nuno Marques
*Sent:* Friday, August 29, 2008 11:36 AM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] Connecting two asterisks via IAX
Hi,
I need to connect 2 asterisks in 2 different countries (A and B) for one
company so it's possible to make connections between the 2 offices.
For connectivity reasons (NAT traversal) i want to connect the 2
asterisk with IAX so that when a user on office A connects via SIP to user
on office B the call is going trought IAX channel.
Can anyone give me an ideia how to accomplish this?
the schema:
(via SIP)(via
IAX)(via SIP)
Office A - Asterisk A --- Asterisk B
- Office B
Thanks in advance,
Nuno
___
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___
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Re: [asterisk-users] GSM recordings
On Thu, Aug 28, 2008 at 01:56:43PM -0500, Javier Prieto Gomez wrote: I think that you can use quicktime.. Correct; Quicktime as a plugin can play GSM. One small note, though: if you're unlucky enough to be using FireFox on Win2k still, you're screwed. The newest QT that will run on Win2k is 7.1.6, and that's blocked by newer versions of FireFox due to a bug. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wi-SIP vs. SIP-DECT
On 29/08/2008 Karl Fife wrote: Anybody care to muse on Wi-SIP vs. SIP-DECT? DECT is an optimised wireless voice system, Wi-Fi isn't. Works for me :) But seriously, I have DECT phones with battery life of a week, I don't think any Wi-Fi device will come within a mile of that (witness the poor life of Nokia Wi-Fi equipped phone batteries). Maybe I'm wrong but a Wi-SIP phone seems like a DIESEL sports car. http://www.americanlemans.com/News/Article.aspx?ID=1872 Phil ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call monitor/barge/train
Robin, Thank you. I'm aware of ChanSpy. And that's a small part, a very small part, to my big picture question. However, I'm still looking for output. Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robin Rodriguez Sent: August 29, 2008 2:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call monitor/barge/train Mark Hamilton wrote: Hi, I'm planning on migrating someone who uses a very mature system. They would be logging in either as AgentLogin() or AQM. The main requirement however, is: The supervisor will have a control panel, where he will see how many of his agents are on call. If they are, he can right-click on the agent and get the options Call Monitor (where the super just listens in on the call, or new reps can listenin), Call Train (where the super and agent can talk to each other for training, but the customer doesn't hear them, or older reps can train newer reps), Call Barge (where everyone can hear everyone else, super agent and caller) How or what can facilitate this? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users looking at the command Chanspy should give you a lot of relevant information. http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy -Robin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR userfield recording name
Hello! Ive setup agents.conf to records all incoming calls and, to get the filename created from CDR userfield Ive uncomment the line createlink=yes. Well, showing into CDR I saw that some filenames appear like: agent-3528-1215461018-19526.gsmagent-3528-1215461018-19526.gsm;agent-4395-12 15461265-19738.gsm and not how would: agent-4263-1215461548-20009.gsm Someone knows how I can solve this issue or why appear in this way? Thanks!! Gustavo A. González Dto. de Infraestructura Despegar.com, Inc. [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing through Zap cards
No it's not a backup, it's the preferred way for lots of reasons. The ones that come to mind: 1. Same PCI slot will be able handle both, which means if properly configured the RTP will not travel the MB hence no IRQ to worry about. 2. One Adit 600 gives you for around $400.00 on eBay 24 FXS ports. 3. You can have POTS on that Adit 600 as well, which will allow you to use some lines for backup etc. On Fri, Aug 29, 2008 at 2:32 PM, James Sneeringer [EMAIL PROTECTED] wrote: On Fri, Aug 29, 2008 at 11:06 AM, C F [EMAIL PROTECTED] wrote: I would suggest that you go with a dual port T1 card and a channel bank. you will have much greater fleibality as well as more stable connection. Thanks for the tip. If faxing in this setup does prove to be unreliable, this is a good backup. -James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Audio data between concurrent SIP and PSTN
Hi. Are the audio streams returned by the user been shared between SIP and PSTN connections? I'm developing a speech recognition engine for Asterisk and I'm facing a problem where Asterisk is crashing when concurrent SIP and PSTN connections occur. I will read the code that implement that to understand it, but if anyone knows about that problem and can explain about it, thank you. -- ___ Allann J. O. Silva ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing through Zap cards
On Fri, Aug 29, 2008 at 4:02 PM, C F [EMAIL PROTECTED] wrote: No it's not a backup, it's the preferred way for lots of reasons. The ones that come to mind: 1. Same PCI slot will be able handle both, which means if properly configured the RTP will not travel the MB hence no IRQ to worry about. 2. One Adit 600 gives you for around $400.00 on eBay 24 FXS ports. 3. You can have POTS on that Adit 600 as well, which will allow you to use some lines for backup etc. I get what you're saying, but... 1. We already know the slots that the cards will go in have their own IRQs. 2. Twenty-four FXS ports is overkill for two faxes. :) 3. Believe it or not, but the office this is going into has no POTS lines, just PRI and T1. So in the end, I think we have a solution that will work, and if it doesn't, I will take your (and CF's) advice with the channel bank. Thanks. -James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audio data between concurrent SIP and PSTN
If I understand what you are asking correctly, no, all media streams - and furthermore, their directional legs - are distinct. IAX2 muxes media streams together into one logical connection. SIP-driven RTP does not. On Fri, August 29, 2008 5:46 pm, Allann Jones wrote: Hi. Are the audio streams returned by the user been shared between SIP and PSTN connections? I'm developing a speech recognition engine for Asterisk and I'm facing a problem where Asterisk is crashing when concurrent SIP and PSTN connections occur. I will read the code that implement that to understand it, but if anyone knows about that problem and can explain about it, thank you. -- ___ Allann J. O. Silva ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wi-SIP vs. SIP-DECT
On Fri, 29 Aug 2008 09:58:56 -0500, Karl Fife wrote: Anybody care to muse on Wi-SIP vs. SIP-DECT? My limited research indicates that none of the WiSip phones will ever be able to match the performance of DECT phones. Maybe I'm wrong but a Wi-SIP phone seems like a DIESEL sports car. There is nothing wrong with the technology, but it seems like a shoe-horned fit into the requirements of a wireless endpoint. DECT uses a wireless radio layer that was engineered from the ground-up with the design priorities of a wireless endpoit. I notice that the standby times of Wi-SIP vs. SIP-DECT are a great illustration of this point. I guess there's no low-power way to participate in a WiFi network, hense standby battery life that sucks in Wi-SIP. I've never actually demoed a Wi-SIP phone on premesis, but if the range of my WiFi LAPTOP vs. my DECT 6.0 headset is any indication, (DECT more than double the range) I'd guess it to be quite hard to make a case for Wi-SIP unless you're doing some straight-up network application integration right onto the phone. Can anyone speak to this? I've used both fairly extensively in a home office setting. DECT is the clear winner. That said, the current crop of wifi APs and SIP handsets can do a good job, but it's gonna be more work and maybe a little more expensive that you think. You need newer APs with WMM. Unless there's a truly compelling reason to go with converged voice+data over wifi I'd recommend DECT in most cases. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reliable wireless SIP phones
On Fri, 29 Aug 2008 07:18:17 -0500, Karl Fife wrote: I've had essentially no problems with my snom m3s. Someone from snom has been in touch to confirm that they are now putting more effort into the firmware for this phone. There are a few new features that I'd like to see that are already in their plans. When I saw held this phone at Astricon 2007, I was impressed with the price/value ratio, but I was frankly a bit turned off by how 'creaky' it felt. If I held it between my ear and shoulder I'd hear the dreaded chintzy plastic creaking sound. I think the one I saw may have been a pre-production prototype. Does your Snom M3 have a cheap plastic creaky build? Does it feel 'chintzy' to you? -Karl I've had mine in service for 8 months and it has not suffered any physical problems. No, it doesn't have the look and feel of a good quality cell phone. That's true. But it hasn't been a problem either. If you're using it in a warehouse you may think differently. The Polycom SpectraLink 8002 wifi handset definitely feels more industrial strength. That's less than ideal for a lot of people. The buttons may last a billion contacts, but your really gotta push to make a positive keypress. It's like the Panasonic Toughbook of cordless phones. Might be less than ideal for an office user. I'm not unhappy with the m3. I may buy a repeater since the S685IP is going bye bye soon. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reliable wireless SIP phones
On Fri, 29 Aug 2008 20:11:54 +0300, Tzafrir Cohen wrote: On Fri, Aug 29, 2008 at 09:56:36AM -0500, Karl Fife wrote: So.. Try getting a mom pop accountancy company to go through having to push 9 key-preses before they get to select the account they want to use for outgoing calls (business or home on their case) THEN remembering to switch it back again. (Their response was Fuck that, can't you make it easier?) LOL to your customer response :-) Sometimes I wonder: Do the people who design these things actually use telephones? And it makes me ask: is there any free software hardware phone? The Siemens DECT line are open source. Not broadly available in the US though. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wi-SIP vs. SIP-DECT
Lets not forget that the DECT specification does allow for data transmission. THere is no reason that in the future you would not be able to have integrated services over DECT. Michael Graves wrote: On Fri, 29 Aug 2008 09:58:56 -0500, Karl Fife wrote: Anybody care to muse on Wi-SIP vs. SIP-DECT? My limited research indicates that none of the WiSip phones will ever be able to match the performance of DECT phones. Maybe I'm wrong but a Wi-SIP phone seems like a DIESEL sports car. There is nothing wrong with the technology, but it seems like a shoe-horned fit into the requirements of a wireless endpoint. DECT uses a wireless radio layer that was engineered from the ground-up with the design priorities of a wireless endpoit. I notice that the standby times of Wi-SIP vs. SIP-DECT are a great illustration of this point. I guess there's no low-power way to participate in a WiFi network, hense standby battery life that sucks in Wi-SIP. I've never actually demoed a Wi-SIP phone on premesis, but if the range of my WiFi LAPTOP vs. my DECT 6.0 headset is any indication, (DECT more than double the range) I'd guess it to be quite hard to make a case for Wi-SIP unless you're doing some straight-up network application integration right onto the phone. Can anyone speak to this? I've used both fairly extensively in a home office setting. DECT is the clear winner. That said, the current crop of wifi APs and SIP handsets can do a good job, but it's gonna be more work and maybe a little more expensive that you think. You need newer APs with WMM. Unless there's a truly compelling reason to go with converged voice+data over wifi I'd recommend DECT in most cases. Michael -- Michael Graves mgravesatmstvp.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zap channel DTMF regeneration
I've got a problem that I hope someone here can shed some light on. It seems that in any calls going over Zap channels (either with a FXO card or PRI card), inbound audio is constantly monitored for DTMF tones, and then these tones are regenerated back in the audio stream either within * if the endpoint is set for inband dtmfmode, or sent out-of-band for the endpoint to regenerate. I have two applications where this has disasterous results: 1. Connecting to another system that does a lot of DTMF signaling 2. Trying to use an iaxmodem and hylafax to receive faxes. The DTMF detection code easily falses and at least mutes the audio. I realize there are certain things (like the IVR) that require the DTMF detection; but on calls from the zap channel to an endpoint, I can't have it muting the audio, and potentially regenerating the tones. Perhaps this could be configurable per extension entry? I think I recall seeing something that looked for the fax initial tone, and would at least disable any echo cancellation - does that also disable DTMF regeneration? I appreciate any help or pointers! Joe Click to go wireless with your computer, ultra fast speed. http://thirdpartyoffers.netzero.net/TGL2231/fc/Ioyw6ijmWbww9f2hhX4f2TcPxLtgcLJ4DlAvmab8VmG43fIiATJTRp/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk CDR Problem
Hi, let me know that you have configured properly in res_pgsql.conf in asterisk with proper, and it is connected properly to database with database details. Thanks, Max Alex Voip Developer On Fri, Aug 29, 2008 at 10:26 AM, Hiren Mistry [EMAIL PROTECTED] wrote: Hi , I have check zapte.conf in and after make some correction that problem solve. But now I am facing other problem. We are using here Postgres Database and the data from CLI it can't insert in Postgres Database. I have also here mention below cdr_pgsql.conf, modules.conf and cdr.conf cdr.conf -- Below [general] [csv] usegmtime=yes ;log date/time in GMT loguniqueid=yes ;log uniqueid loguserfield=yes ;log user field -- cdr_pgsql.conf -- Below [global] hostname=localhost ;hostname=122.160.10.81 port=5432 dbname=asterisk password=postgres user=postgres table=cdr -- modules.conf -- Below [modules] autoload=yes ;preload = res_odbc.so ;preload = res_config_odbc.so noload = pbx_gtkconsole.so ;load = pbx_gtkconsole.so noload = pbx_kdeconsole.so load = res_musiconhold.so noload = chan_alsa.so -- OUTPUT on CLI --- Below localhost*CLI cdr status CDR logging: enabled CDR mode: simple CDR registered backend: csv CDR registered backend: cdr_manager CDR registered backend: cdr-custom localhost*CLI also when I load manually cdr_pgsql.so on CLI then it show error which is also I describe below localhost*CLI module load cdr_pgsql.so [Aug 29 10:22:21] WARNING[8984]: loader.c:362 load_dynamic_module: Error loading module 'cdr_pgsql.so': libpq.so.5: cannot open shared object file: No such file or directory [Aug 29 10:22:21] WARNING[8984]: loader.c:614 load_resource: Module 'cdr_pgsql.so' could not be loaded. So, Please guide me for load Postgres module in asterisk for CDR Database. Subject: Re: [asterisk-users] Asterisk CLI Show Error :- (**Unknown**) instead of (Zap/22-1, ) From: Max Alex [EMAIL PROTECTED] [EMAIL PROTECTED] Date: Thu, 28 Aug 2008 11:48:16 +0530To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com asterisk-users@lists.digium.com Hi Hiren, Have you properly configured the zap channels in asterisk, which device have you configured in asterisk with zaptel? let me know the dial plan for ivr. Thanks, Max Alex Voip Developer On Thu, Aug 28, 2008 at 11:40 AM, Hiren Mistry [EMAIL PROTECTED] wrote: Hi, Everybody, I am planning to make a new IVR on Asterisk I have Installed zaptel , libpri, asterisk, asterisk-addon on CentOS 5 I also start service of zaptel and asterisk it start successfully. But when goto asterisk CLI prompt and check this IVR then all call string with (**Unknown**) instead of (Zap/22-1, ) and I have also 3 other Asterisk base IVR which is also on CentOS. [Asterisk CLI Executing [EMAIL PROTECTED]:1] Answer(**Unknown**, ) in new stack ] Please Help me for Configuring this IVR. -- With Regards, Hiren Mistry -- With Regards, Hiren Mistry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reliable wireless SIP phones
On Fri, Aug 29, 2008 at 06:08:40PM -0500, Michael Graves wrote: On Fri, 29 Aug 2008 20:11:54 +0300, Tzafrir Cohen wrote: On Fri, Aug 29, 2008 at 09:56:36AM -0500, Karl Fife wrote: So.. Try getting a mom pop accountancy company to go through having to push 9 key-preses before they get to select the account they want to use for outgoing calls (business or home on their case) THEN remembering to switch it back again. (Their response was Fuck that, can't you make it easier?) LOL to your customer response :-) Sometimes I wonder: Do the people who design these things actually use telephones? And it makes me ask: is there any free software hardware phone? The Siemens DECT line are open source. Not broadly available in the US though. What does this mean? Could you please provide a link for more information? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
