Re: [asterisk-users] Audio dropping
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roger Burton West Sent: 28 May, 2011 23:50 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Audio dropping On Fri, May 27, 2011 at 10:31:57AM +0200, Mark Scholten wrote: What could the reason be audio in 1 direction is dropping? (Normally from the Asterisk server to the mentioned SIP clients.) No clear information is in the logs (it is like the call ended normally) and not all calls are having problem (most not, but it happens to often for us to start using VoIP more at the moment). While the most usual problem is packet filtering / NAT, this generally manifests as no audio at all in one direction, not a drop in mid-call. But it's possible that one of the intermediate transit providers is doing something clever. (Disabling ping, as you mention in your later email, is often a good indicator of a company with insufficient Clue.) Are you in a position to tunnel the traffic over a VPN or similarly flat and unfilterable network link? (This might be a good idea anyway.) Hello Roger, I'm not in a position to start doing that. The ping is disabled on the firewall that is also doing the NAT (not something the provider does). However the provider is blocking other things (incoming email for example, for outgoing it is something I can understand). The strange thing is it is dropping mid-call, that is the reason I don't think it is only a NAT problem. Another network we use offered us a server so we can test if it is related to the other network we use. Regards, Mark -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audio dropping
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik Sent: 27 May, 2011 10:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Audio dropping On Fri, 2011-05-27 at 10:31 +0200, Mark Scholten wrote: Hello, We see some strange behavior with phone calls, we use Asterisk 1.8.3.3. SIP clients (all behind NAT at different locations, so not a single NAT solution is used): - x-lite - linksys pap2t - polycom kirk (multiple type numbers) - polycom (multiple type numbers, hardware phones) Our Asterisk servers stays in between (some calls are recorded). Asterisk is running on a physical server (no virtual server software) with old hardware (Xeon 3.2 GHz with hypertrading and 4GB RAM, mainly used for buffers). We use a MySQL backend (CDR records are stored in it and SIP users are stored in a MySQL database). We use a SIP provider with a trunk for outgoing and incoming calls, this is also an Asterisk server if I'm correct. We currently do around 1000 calls a week and max. do 10 calls at the same time. The Asterisk server is not behind a NAT. What could the reason be audio in 1 direction is dropping? (Normally from the Asterisk server to the mentioned SIP clients.) No clear information is in the logs (it is like the call ended normally) and not all calls are having problem (most not, but it happens to often for us to start using VoIP more at the moment). To test if it was the firewall we disabled the firewall on the Asterisk server and moved the Asterisk server before the other firewalls we have. What could the problem be? And even more important what could solve it (and/or explain it)? Kind regards, Mark Scholten Hi Are the broadband connections to the target SIP extensions dedicated for VoIP or does any other traffic run over them? We tend to find that 80% of call quality issues are caused by the broadband connection. A good diagnostic tool to keep an eye on the broadband connections involved is smokeping http://oss.oetiker.ch/smokeping/ We find it an absolute godsend. === Hello, Ping is disabled on the location with most call problems (filtered). A few clients with dedicated audio broadband connections have the same problem compared to shared broadband connections. This makes it very difficult to find the problem. A supplier we use offered to test it with a server from them, if that solves it it was probably a network issue. Regards, Mark -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Audio dropping
Hello, We see some strange behavior with phone calls, we use Asterisk 1.8.3.3. SIP clients (all behind NAT at different locations, so not a single NAT solution is used): - x-lite - linksys pap2t - polycom kirk (multiple type numbers) - polycom (multiple type numbers, hardware phones) Our Asterisk servers stays in between (some calls are recorded). Asterisk is running on a physical server (no virtual server software) with old hardware (Xeon 3.2 GHz with hypertrading and 4GB RAM, mainly used for buffers). We use a MySQL backend (CDR records are stored in it and SIP users are stored in a MySQL database). We use a SIP provider with a trunk for outgoing and incoming calls, this is also an Asterisk server if I'm correct. We currently do around 1000 calls a week and max. do 10 calls at the same time. The Asterisk server is not behind a NAT. What could the reason be audio in 1 direction is dropping? (Normally from the Asterisk server to the mentioned SIP clients.) No clear information is in the logs (it is like the call ended normally) and not all calls are having problem (most not, but it happens to often for us to start using VoIP more at the moment). To test if it was the firewall we disabled the firewall on the Asterisk server and moved the Asterisk server before the other firewalls we have. What could the problem be? And even more important what could solve it (and/or explain it)? Kind regards, Mark Scholten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A few questions regarding Asterisk 1.8.0
Good luck as with any new version there may be some bugs so if you bump up against ones report them so they can be fixed. Also don't just drop it into production with out testing it on a box for a bit. 1.8 has a lot of changes. Most appear to be for the better. The only important difference I could find while testing was that with 1.6.x you could use nat=route. This doesn't work anymore with 1.8.0 (and I didn't find it in the ChangeLog 1.8.0). Changing nat=route to nat=yes seems to work. (I mention it here so others can find it in the future). Regards, Mark -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Door Contacts via Asterisk?
Hello, We did something like that in the past (but for 1 company, but it shouldn't be really different). The easiest solution for us was to use a door opener that could work with almost any normall phone connection and use a Linksys pap2t or something similar. With kind regards, Mark Scholten From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith Sent: Monday, November 15, 2010 7:35 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Door Contacts via Asterisk? Hi all, I have had (what I consider) an odd request. The installation I'm working on now is an office on a multi-floor building. They 're looking for some kind of solution with the phone system to provide door control. We are a non-profit so of course I'm looking for something VERY inexpensive. I'm sure /someone/ has done something like this. I'd appreciate any ideas. Cassius Smith -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] A few questions regarding Asterisk 1.8.0
Hello, I have a few questions regarding Asterisk 1.8.0. If you can answer a question, please do so. Is Asterisk 1.8.0 stable enough for production environments? Is it possible (and if yes what is the best option) to use CDR MySQL with Asterisk 1.8.0? With 1.6.x we use the add-on package for that, however we could do something with scripts to do it (but I don't like the idea). If it is stable and there is a good option for CDR with MySQL we will start using it very soon. Regards, Mark -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX Options
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Alejandro Imass
Sent: Monday, August 02, 2010 9:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] FAX Options
Hi,
Is FAXing with Asterisk a practical option ? Or is it better just to
use a plain fax connected to an FXS and just switch with Asterisk. I
specifically wanted to know if there was any experience using just the
fax scanner to send faxes and receive them via asterisk and the to
e-mail. My idea was to take my old fax connect it to an FXS port and
send faxes with the fax machine (using the fax mainly as a scanner),
but receive them through our existing FXO jack that is connected to
the PSTN. the scheme would be something like:
PSTN -- FXO -
|
|Asterisk
|
FAX -- FXS -
I'm using Asterisk 1.4.26.2 on FreeBSD 8.0
Here we have the following setup, could you say if that is acceptable for
you?
Outgoing fax:
Fax - Linksys pap2t (sip, no t38, for settings see
http://www.provu.co.uk/pdf/sipura/ip_faxing_sipura_linksys.pdf) - asterisk
- sip trunk provider (this could also be some sip - pstn solution I guess)
Incoming fax:
Sip trunk provider - asterisk - email
For the incoming fax I use a separate context, below I've listed an example:
exten = 1000,1,Answer
exten =
1000,2,Set(FAXFILE=/var/spool/asterisk/fax/${CALLERID(num)}_${EPOCH}_client.
tif)
exten = 1000,3,Set(CLID=${CALLERID(num)})
exten = 1000,4,Set(EMAIL=email address)
exten = 1000,5,Set(TRADENAME=tradename (used in the email))
exten = 1000,6,Wait(3)
exten = 1000,7,ReceiveFax(${FAXFILE})
exten = 1000,8,Hangup
exten = h,1,System(/usr/bin/php /etc/scripts/fax2mail.php ${FAXFILE}
${CLID} ${EMAIL} ${TRADENAME})
fax2mail.php (tiff2pdf and phpmailer are required):
?php
$faxfile = $_SERVER[argv][1];
$callerid = $_SERVER[argv][2];
$email = $_SERVER[argv][3];
shell_exec(/usr/bin/tiff2pdf
-o/var/spool/asterisk/fax/.$callerid..pdf .$faxfile);
$bijlage = /var/spool/asterisk/fax/.$callerid..pdf;
switch ( $_SERVER[argv][4] ) {
case trade: $tradename = 'our trademark'; $from =
'f...@domain.tld';$fromname = $tradename.' - Fax system'; break;
default: $tradename = 'our trademark'; $from =
'f...@domain.tld';$fromname = $tradename.' - Fax system'; break;
}
require(/etc/scripts/class.phpmailer.php);
$mail = new PHPMailer();
// $mail-IsMail(); // telling the class to use Mail functie
van PHP
$mail-IsSMTP(); // telling the class to use SMTP
$mail-Host = ; // SMTP server
$mail-SMTPAuth = true; // turn on SMTP authentication
$mail-Username = ; // SMTP username
$mail-Password = ; // SMTP password
$mail-From = $from;
$mail-AddAddress($email);
$mail-FromName = $fromname;
$mail-AddAttachment($bijlage);
$mail-Subject = Received fax from .$callerid;
$mail-AddReplyTo = $email;
$mail-IsHTML(false);
$mail-Body = email body;
$mail-Send();
//shell_exec(/bin/rm /var/spool/asterisk/fax/.$callerid..pdf
.$faxfile);
?
I agree that it isn't a beautiful solution, however it works. Sending a fax
directly with asterisk is probably also possible (I didn't test it).
Asterisk version: 1.6.2.6 (yes I know that I should update)
Regards, Mark
TIA,
Alejandro Imass
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Re: [asterisk-users] # -key not to be 'transfer'
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Sunday, August 01, 2010 3:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] # -key not to be 'transfer' Hello list, whenever I press the #-key I hear a voice saying 'transfer'. How can I use the #-key without this voice-message or without having it the function of unattended transfer ?! The T or t option is not set in my Dial()-command so I don't know where this transfer is coming from in the first place. Kind regards, Jonas. Look in features.conf for the # and replace it for example with ##. Regards, Mark -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Confirm answering a call
Hello, I am working on getting the following to work and I couldn't find it in the documentation I did read. Where should I look or does someone have an example how I can do it? Current situation: Incoming call - 3 SIP phones + 2 mobile phones ring - if mobile phone goes to voicemail the call is answered by that voicemail (if a phone is in use for another call the call directly goes to that voicemail) Situation I want: Incoming call - 3 SIP + 2 mobile phones ring - if the call is answered by a mobile phone the person picking up the call needs to press 1 (or another key on the phone) to answer the phone, if that key is not pressed all phones keep ringing as being it an unanswered call If it is also required to press the key on the SIP phones than that is acceptable. Is it possible? Where should I look? I know some systems use it. Regards, Mark -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bridging old system (ESI IVX E) with new Asterisk server
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of John Novack Sent: Tuesday, May 04, 2010 12:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bridging old system (ESI IVX E) with new Asterisk server Eddie Mikell wrote: All: My company has an existing ESI IVX E-class system with 45 phones. I can add one more card, to expand it another 6 phones, but it's $8000, and then the system will have to be replaced. That is worse than highway robbery. I feel sure with some careful searching you can do much better for the card and the 6 phones. Coupling the two systems together will not be well received by the users. Just one opinion If you really want to move to Asterisk, then do it, but it will take more than a weekend of careful consideration and testing! We did something like that by a client not too long ago. First we did place the new phones next to the existing phones (so user could use it for outbound calls + intern calls and get familiar with it). After 2 weeks we did start the move for the incoming number and now they are using it for all calls (except the mobile phones). To explain everything to the persons that need to use it we did create a short manual and we give a few persons a special number to call if they need help. Currently we are planning to do the same thing at another client. We did decide to make options available on the asterisk server and not on the phone (as they also use PAP2T boxes at some locations and options have to work the same way everywhere). Mark -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of James Lamanna Sent: Saturday, May 01, 2010 9:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286 It seems that the PAP2T does support TFTP and an XML-based config for deployments... I've used both the Grandstream 286 and the Linksys PAP2T. I have been able to get some limited faxing to work using T30 with a PAP2T. Configuration and provisioning of the Linksys is very easy through either the web GUI or XML configuration files, which can be transferred through TFTP or HTTP. I can only hope that Cisco will update the firmware of the PAP2T to support T38 one day... Related to this: what is a good ATA with a reasonable price that works correct for sending and receiving a fax? The faxes we have (on multiple locations) require a analog connection. We have sending a fax working with the PAP2T and incoming faxes are send to an email address (as a PDF). Regards, Mark -- James -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
