Re: [asterisk-users] Outgoing Calls Only -- Firewall Rules
Asterisk 1.4.29 or so. access-list _dmz_acl extended permit udp 10.129.42.0 255.255.255.0 any range 1 2 access-list _dmz_acl extended permit udp 10.129.42.0 255.255.255.0 any eq 5060 But yes, all your feedback worked. I didn't need to port-forward any incoming ports, only 5060/1-2 for outgoing UDP. The only issue I'm now having is: --- SIP read from 66.227.100.20:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 209.34.93.68:5060;branch=z9hG4bK3eb38bde;rport=51566 Warning: 392 66.227.100.20:5060 Noisy feedback tells: pid=9611 req_src_ip=209.34.93.68 req_src_port=51566 in_uri=sip:sip.jnctn.netout_uri=sip: sip.jnctn.net via_cnt==1 209.34.93.68 is my IP, 209.34.93.68 is Junction Networks (for this example). I also get it from my backbone providers as well so it's likely something to do with that 51566 req_src_port thing. Any idea what this is an how to configure it to a restricted range of IP addresses? Nicholas Blasgen Partner / Network Operations Refractive Dialer LLC (724) 252-7436 On Sun, Jan 3, 2010 at 8:29 PM, Max McGraw max.mcg...@gmail.com wrote: Nicholas, you haven't specified which version, which does make a lot of difference. 1.6.x can easily traverse NAT. If you are only making outbound calls, you shouldn't need to forward 5060. Unless you have a special NAT that is blocking outbound connections, the SIP.conf settings below should work whether your provider uses SIP registrations or not. My codec related settings may not be applicable to your installation : ; - [general] dtmfmode=rfc2833 relaxdtmf=yess bandwidth=high disallow=all allow=ulaw ; ; NAT stuff ; localnet=192.168.x.0/255.255.255.0 externip=a.b.c.d:5060 nat=yes ; ; Media stuff ; canreinvite=no ; ; [your-voip-provider-para] ; context=default type=friend ; ; your provider's outbound gateway ; host=w.x.y.z ; dtmfmode=rfc2833 relaxdtmf=yess disallow=all allow=ulaw ; ; - On Sun, Jan 3, 2010, Nicholas Blasgenwrote: I'm trying to move my Asterisk deployments under a Virtual IP address and now remember why I dislike this. My primary Asterisk system is now behind a firewall in private address space. My question is what ports are needed to be opened just for the purpose of placing outgoing calls. I would have assumed none, but I can't even get replies on registration from any of my 3 VoIP providers. I tried defining the External IP and some other stuff, but I assume it's fully an issue with the firewall. Do I really need 5060 port forwarded just to register with remote hosts? Nicholas Blasgen Partner / Network Operations Refractive Dialer LLC (724) 252-7436 __ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Outgoing Calls Only -- Firewall Rules
I'm trying to move my Asterisk deployments under a Virtual IP address and now remember why I dislike this. My primary Asterisk system is now behind a firewall in private address space. My question is what ports are needed to be opened just for the purpose of placing outgoing calls. I would have assumed none, but I can't even get replies on registration from any of my 3 VoIP providers. I tried defining the External IP and some other stuff, but I assume it's fully an issue with the firewall. Do I really need 5060 port forwarded just to register with remote hosts? Nicholas Blasgen Partner / Network Operations Refractive Dialer LLC (724) 252-7436 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk???
I just want to also remind people that Skype for SIP is also to be released shortly. When I last talked to Skype they said it would be out in late July. So I assume if you wait another few more weeks the entire issue will be moot. No $60/channel fee, just the free SIP platform for people using the business version of Skype. Nicholas Blasgen Partner / Network Operations Refractive Dialer LLC (724) 252-7436 On Tue, Aug 18, 2009 at 10:35 AM, Pascal Bruno tipas...@gmail.com wrote: Lol but he has a good point and makes a lot of sense. Never thought about that strategy... On Tue, Aug 18, 2009 at 12:16 PM, Thomas Kenyon dig...@sanguinarius.co.uk wrote: Michael Graves wrote: Pricing is a very legitimate way to minimise support effort. It winnows down the market size to a point where the company offering the goods can sustain the projected per user support issues. You can always drop the price later on when you have a better handle on the per user support issue. Michael You make it sound like you're saying it's expensive because it doesn't work :-) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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 2009 - October 13 - 15 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] IAX2 ActiveX Control
I'm sure I saw a MS C++ library that had additional support to be wrapped up as an ActiveX client. But I can't seem to find anything now. SIP ActiveX clients are around. Or maybe this is it: http://www.secondsignal.com/secondsignal/sshome.nsf/html/2ndSignal-IAXClientWrapper2005 Nicholas Blasgen Partner / Network Operations Refractive Dialer LLC (724) 252-7436 On Tue, Aug 18, 2009 at 8:25 AM, Meftah Tayeb tayeb.mef...@gmail.comwrote: hello, please any IAX2 ActiveX control that wrap libiax2 or libiaxclient? i want to develope my softphone in delphi thanks __ Information from ESET NOD32 Antivirus, version of virus signature database 4345 (20090818) __ The message was checked by ESET NOD32 Antivirus. http://www.eset.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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 Trunk groups
I've improved this since this revision, but now a days I don't use limited
systems. But my code has been used in places that need 100 concurrent
outgoing lines.
[macro-which-line]
exten = s,1,set(TRIES=0)
exten = s,n(nextone),set(TRIES=$[${TRIES} + 1]) ; increment TRIES by 1
exten = s,n,set(DIALSTRING=${TRY${TRIES}}) ; assign TRYn to DIALSTRING
exten = s,n,gotoif($[${DIALSTRING} = ]?donehere) ; see if we've run out
of things to try
exten = s,n,ChanIsAvail(${DIALSTRING}) ; it will be up or down, no need for
this to be exclusive
exten = s,n,gotoif($[${AVAILSTATUS} = 0]?:nextone)
exten = s,n,gotoif($[${GROUP_COUNT(${DIALSTRING})} = 2]?nextone) ; have we
used up the allowed calls on this channel
exten = s,n,set(GROUP()=${DIALSTRING}) ; Lock the line! Yay...
exten = s,n,Dial(${DIALSTRING}/1${ARG1}) ; dial the phone
exten = s,n,GotoIf($[${DIALSTATUS} = BUSY]?donehere) ; Don't keep
dialing
exten = s,n,NoOp(Moving to the next one...);
exten = s,n,goto(nextone) ; TEMP
exten = s,n(donehere),MacroExit() ; we only get here if everything failed
Then in GLOBALS you just set things like:
TRY0=SIP/trunk1
TRY1=SIP/trunk2
TRY3=SIP/other1
The above code is limited to 2 lines per channel. The code I used
originally (not sure where I found it anymore, might have been this mailing
list or might have been Voip-Info) support defining how many channels you
wanted to use for each provider (ie, provider1 has 2 lines free, but
provider2 has 5 lines). The original code didn't hold up though since if
multiple lines were being dialed at the exact same instance they would both
return the same availability before dialing the line. So in this one, I try
to lock the line early and if I get some other kind of error I move on to
the next group because I might have failed due to another race condition.
Anyways, tons of problems when you're limited on channels. Mine is the best
and one of a very few I've ever seen. SuperDial, I feel, is a silly idea.
It's exactly the same as a regular Dial string. No clue why you'd use it
over Dial. And the reason Dial doesn't work is because if the Dial'ed line
hangs up it returns back to the orginal Dial Plan. Doesn't help at all.
You hang up on the person, the person goes to the next line in the dial
plan, and you get called again. You hang up, they call you back again.
Soulds like a good way to use up air time.
Nicholas Blasgen
Partner / Network Operations
Refractive Dialer LLC
415.692-5277 (w)
408.497.9796 (c)
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Re: [asterisk-users] Auto-congesting call due to slow response
I'd look at the packet delay. Log some of the packets, see how long until you get a response from the remote host. If the delay is really long, then that's the issue (which by the response I assume that's exactly what's happening). Lower the load on the system and see if the delay improves. Or you can increase the timeout if you really wanted. channels/chan_iax2.c With debugging on, it seems that this data is available. But it's the same timeout as a Peer would be. And trust me, that timeout is huge. So that makes me think of another issue. I know with my VoIP provider, they told me not to trust the PEER POKE responses because I kept seeing my provider connect, disconnect, connect, disconnect. They had me turn off the qualification. (This is all SIP so I'm not sure how it translates to IAX). Might not want to waste the packets to send data to a server that is always available. If you don't get any help, you can try opening it as a bug on Digium's Bug Tracker but I assume the issue isn't a bug but just an overloaded system with a slow response time. Nicholas Blasgen Partner / Network Operations Refractive Dialer LLC 415.692-5277 (w) 408.497.9796 (c) Please update your contact records with my new work number. On Wed, May 27, 2009 at 10:52 AM, Alexander Topolanek at...@ocv.org wrote: Hello, I'm running several asterisks in a carrier environment. The asterisks do mainly gateway business between E1 cards and IAX with some routing logic. On one key server I see issues of Auto-congesting call due to slow response coming every number of calls. The IAX peer is in the same subnet, the servers are not really loaded. Versions in use are 1.2.2 and 1.4.23-rc3, with rsa key authentication in use any ideas? kind regards -- Alexander Topolanek ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting CDR values on failed calls
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate ActionId and Account can be set. Nicholas Blasgen Partner / Network Operations Refractive Dialer LLC 415.692-5277 (w) 408.497.9796 (c) Please update your contact records with my new work number. On Wed, May 27, 2009 at 10:24 AM, John Regal jre...@gmail.com wrote: Hi All, I am relatively new to Asterisk… I have CDR enabled and successfully writing to MS SQL server. In my cdr table I am setting the userfield value with a line in my dialplan. If a call is placed to an invalid number (e.g. 12125551212), I see a cdr record created, however, my userfield value never gets set since the call never made it into the context of my dialplan. I am using AMI with the Originate command to invoke the call. How can I set this value *before* the call is actually passed to my voip provider (of whom quickly responds with “Got SIP response 500 ‘Service Unavailable’ back from *myVoipIPaddress*”) ? Thanks in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Manager API Action Originate
Matt, Oh, I thought it was Asterisk 1.4.23 like I wrote in my first email, but turns out to be Asterisk SVN-branch-1.4-r191778. But yes, I am talking about originateresponse. I'm going to do some more debugging today to see if I can get the more information about the issue. When I either Originate from the CLI or from AMI, I don't get anything on the console for either the errors or the initial connection. I've had a lot of issues trying to debug Originate as a result. And no CDR logs are being recorded. On Tue, May 12, 2009 at 5:36 AM, Matt Riddell li...@venturevoip.com wrote: On 12/05/2009 3:44 p.m., Nicholas Blasgen wrote: Has anyone else had issues with Originate returning the wrong error code? According to the docs, the following errors are supposed to be returned: 0 = no such extension or number 1 = no answer 4 = answered 8 = congested or not available Are you referring to the originateresponse event? Which version of Asterisk? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Manager API Action Originate
Matt Others, So to continue the issue, here's what I've learned. Tested on Asterisk: 1.4.24.1 SVN 193870 SVN 191778 So I think that covers most everything. What I've learned is that any Timeout sends back a response code of ZERO instead of what I would have expected, ONE. Anyone offer any other suggestions to try? My way to test this was to make a simple script to perform an AMI Originate call with a 4 second timeout. I then have a standard tool to display all AMI Events. On every system I tried I would get Response of Failure and Error Code of ZERO. On Tue, May 12, 2009 at 10:13 AM, Nicholas Blasgen nicho...@refractivedialer.com wrote: Matt, Oh, I thought it was Asterisk 1.4.23 like I wrote in my first email, but turns out to be Asterisk SVN-branch-1.4-r191778. But yes, I am talking about originateresponse. I'm going to do some more debugging today to see if I can get the more information about the issue. When I either Originate from the CLI or from AMI, I don't get anything on the console for either the errors or the initial connection. I've had a lot of issues trying to debug Originate as a result. And no CDR logs are being recorded. On Tue, May 12, 2009 at 5:36 AM, Matt Riddell li...@venturevoip.comwrote: On 12/05/2009 3:44 p.m., Nicholas Blasgen wrote: Has anyone else had issues with Originate returning the wrong error code? According to the docs, the following errors are supposed to be returned: 0 = no such extension or number 1 = no answer 4 = answered 8 = congested or not available Are you referring to the originateresponse event? Which version of Asterisk? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Manager API Action Originate
Has anyone else had issues with Originate returning the wrong error code? According to the docs, the following errors are supposed to be returned: 0 = no such extension or number 1 = no answer 4 = answered 8 = congested or not available Now in Asterisk 1.4.23 I get some error code 5's but since they're so few I tend not to worry. But what is concerning is the number of Error 0's I get. Error 0 is No Such Extension (ie, Failure I assume) but my Provider's CDR log shows No Answer. (I would show you my CDR but it seems Originate doesn't log in the CDR like every other call for some reason). Any ideas to correct this issue? Or is there a better updated version of that list that would fix my understanding of what the error codes were? Nicholas Blasgen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ChanSpy or other variant
Thank you Mark. I did try it out myself and figured out that it did work as I wanted. Thanks for the quick reply though. Nicholas Blasgen Partner / Network Operations Refractive Dialer LLC 408.395.2110 (w) 408.497.9796 (c) On Mon, Feb 2, 2009 at 12:06 PM, Mark Michelson mmichel...@digium.comwrote: Nicholas Blasgen wrote: I'm trying to figure out how to listen in to a channel that I specify. I have the impression I've seen this done via Flash web controls, but I'm trying to write something myself and I can't figure out what command would be used. ChanSpy looks great, but I don't see how to specify the channel. I have a channel identifier like SIP/provider-08748db0 which is what I would send to applications like Hangup(chan) or Redirect(chan) but it doesn't look like ChanSpy was written to accept that format. I haven't tried passing SIP/provider-08748db0 to ChanSpy, but from the documentation it seems that it shouldn't work. So the question is, how can I listen into a channel if I know either the channel or the unqiue id? And in the meantime I will play around with ChanSpy more. Chanspy should do exactly what you want. If you ran exten = blah,n,ChanSpy(SIP/provider) Then you would be able to listen to all active calls involving any channel whose name begins with 'SIP/provider'. If it turns out that there is a channel called 'SIP/provider-12345abc', then that channel may be spied on with the above ChanSpy call in the dialplan. The thing to remember is that the chanprefix argument as it is described in ChanSpy's documentation is literally any text that may appear at the start of a channel name. Chanspy(SIP) would allow you to spy on any SIP channel, whereas ChanSpy(S) would allow spying on both SIP and Skinny channels. There is no minimum or maximum limit to what this string may be. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ChanSpy or other variant
I'm trying to figure out how to listen in to a channel that I specify. I have the impression I've seen this done via Flash web controls, but I'm trying to write something myself and I can't figure out what command would be used. ChanSpy looks great, but I don't see how to specify the channel. I have a channel identifier like SIP/provider-08748db0 which is what I would send to applications like Hangup(chan) or Redirect(chan) but it doesn't look like ChanSpy was written to accept that format. I haven't tried passing SIP/provider-08748db0 to ChanSpy, but from the documentation it seems that it shouldn't work. So the question is, how can I listen into a channel if I know either the channel or the unqiue id? And in the meantime I will play around with ChanSpy more. Nicholas Blasgen Partner / Network Operations Refractive Dialer LLC 408.395.2110 (w) 408.497.9796 (c) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transfer via AMI
I have a call between two people. I know their channel identifier. I want to trasfer a call away from one person and pass it to another person. To start, let's talk about a blind transfer. My system places both outgoing calls to people and bridges them together (cheaper, works via AGI). Action: Redirect Channel: prospect ExtraChannel: 0 Exten: SIP/transfer_to Context: default Priority: 1 So that works just fine. I'm having an issue however that when the person who was orginally talking decides to hang up his call, Asterisk disconnects the other line as well, as if the ownership of that line is still controled by the orginal process. I'd love to solve that problem. Maybe putting the SIP/transfer_to into the ExtraChannel and then transfering them to a conference room. Suggestions welcome. Could also be that AGI maintains control of any channels it creates and when the main calling line dies, it kills all the others even if they've been transfered away. Okay, in the end, I'd like this to be assisted transfer. Place the party on hold, call another party, and then bridge the two together. Whenever a channel is taken away from the current person, the call status is returned and my AGI script can continue. So I think it should be fine. Has anyone done anything like this? Any pointers would be great. PS: (update since I wrote this original message a while back), via the web, you click a link. That creates a CALL file which calls your number. Once connected, it passes it to an extension that spawns an AGI program. That AGI program looks in the database for the number you wanted to call and places that phone call. You than chat with that person and decide that you're done with that call and want to go onto your next phone call. I use the Asterisk Manager Interface (AMI) to perform a Redirect on the person you're talking to. Doing this causes the AGI script to continue. -- Nicholas Blasgen [EMAIL PROTECTED] 408.497.9796 (c) ___ -- 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 and Fedora 9
You wont need things like PHP, MySQL, etc but you do need some of the other things otherwise you'll get errors. And while I run these as automated batches, I suggest you take my commands and do them one line at a time. Keep an eye out for errors. yum -y install kernel kernel-devel ntp yum -y install subversion gcc gcc-c++ libtermcap-devel bison yum -y update ntpdate time.apple.com cd /usr/src svn checkout http://svn.digium.com/svn/zaptel/branches/1.4 zaptel svn checkout http://svn.digium.com/svn/asterisk/branches/1.4 asterisk svn checkout http://svn.digium.com/svn/asterisk-addons/branches/1.4 asterisk-addons cd zaptel; ./configure; make; make install; make config; cd .. cd asterisk; ./configure; make; make install; make samples; cd .. cd asterisk-addons; ./configure --with-mysqlclient=/usr; make; make install; make samples; cd .. On Fri, Sep 12, 2008 at 2:02 PM, Jonn R Taylor [EMAIL PROTECTED]wrote: http://www.taylortelephone.com/asterisk/ There are install scripts for Centos 5 Asterisk 1.4. They should work just fine on FC9. If you have a problem just email me. Jonn -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Pascal Bruno *Sent:* Friday, September 12, 2008 9:14 AM *To:* [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Asterisk and Fedora 9 Ok very good, how about for the asterisk addonds and sounds? Can you provide me the commands to get, build and install for the 1.4.21 version? Thanks a lot guys. On Fri, Sep 12, 2008 at 6:07 AM, MFH [EMAIL PROTECTED] wrote: The best way I can think of is: wget http://ftp.digium.com/pub/asterisk/asterisk-1.4.21.2.tar.gz tar -zxvf asterisk-1.4.21.2.tar.gz cd asterisk-1.4.21.2 ./configure make menuselect (You don't have to select anything) make make install make samples Pascal Bruno wrote: I am about to install Asterisk on a Fedora 9 box, but i see with yum, they only have Asterisk 1.6 beta in the package repos which I didn't really want to install until they have a stable release. Does anybody know or have a good and easy way to install Asterisk 1.4 on fedora 9? Thank you. ___ -- 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 -- Nicholas Blasgen [EMAIL PROTECTED] 408.497.9796 (c) ___ -- 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] ASM / AMI Assisted Live Transfer
I have a call between two people. I know their channel identifier. I want to trasfer a call away from one person and pass it to another person. To start, let's talk about a blind transfer. My system places both outgoing calls to people and bridges them together (cheaper, works via AGI). Action: Redirect Channel: prospect ExtraChannel: 0 Exten: SIP/transfer_to Context: default Priority: 1 So that works just fine. I'm having an issue however that when the person who was orginally talking decides to hang up his call, Asterisk disconnects the other line as well, as if the ownership of that line is still controled by the orginal process. I'd love to solve that problem. Maybe putting the SIP/transfer_to into the ExtraChannel and then transfering them to a conference room. Suggestions welcome. Could also be that AGI maintains control of any channels it creates and when the main calling line dies, it kills all the others even if they've been transfered away. Okay, in the end, I'd like this to be assisted transfer. Place the party on hold, call another party, and then bridge the two together. Whenever a channel is taken away from the current person, the call status is returned and my AGI script can continue. So I think it should be fine. Has anyone done anything like this? Any pointers would be great. -- Nicholas Blasgen [EMAIL PROTECTED] 408.497.9796 (c) ___ -- 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] Is there a way to encrypt passwords stored in the realtime database?
I've never used it, but check out the md5 one-way encryption of passwords: http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+md5secret http://books.google.com/books?id=vAT8Mfvp8GsCpg=PA225lpg=PA225dq=asterisk+md5+secretsource=webots=1mUADiyRkPsig=FJSBgcWMY3K0zoilVvgNvibJE4Ahl=ensa=Xoi=book_resultresnum=6ct=result On Wed, Aug 20, 2008 at 10:00 AM, Eric Chamberlain [EMAIL PROTECTED] wrote: We are exploring using Asterisk for a project and we are looking for a way to encrypt/decrypt the peer passwords stored in the realtime database (postrges). Ideally, we want to use a public key to encrypt the passwords before they go into the database and have Asterisk use a private key to decrypt the password as part of the call out process. Has anyone developed something like this? -- Eric Chamberlain Founder RF.com http://RF.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 -- Nicholas Blasgen [EMAIL PROTECTED] 408.497.9796 (c) ___ -- 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] RTP Packets Going To Wrong IP Address
I have a user behind a firewall who's had no issues in the past connecting though his firewall. He's registered just fine. But when he places a call, a large number of them have no audio on either side of the connection. No one can hear him, he can't hear anyone as well. After a lot of poking around (and changing many settings) I noticed that Asterisk is communicating the RTP packets to an internal IP address. My server has no internal IP address, only an external address, so it's not like we're trying to route this anywhere else. As can be seen below, I've already identified the host as being behind a firewall and therefor to not trust packets from it. Anyone have a suggestion? Name/username HostDyn Nat ACL Port Status Realtime jfabriquer/jfabriquer 75.36.34.98 D N 55266OK (145 ms) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Asterisk SVN-branch-1.4-r118365 -- Nicholas Blasgen [EMAIL PROTECTED] 408.497.9796 (c) ___ -- 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] WaitForSilence Problems
Actually, I thought about it for a while. What I want is something that
will allow me to restart the message if another sound is detected.
Something like this:
exten = answermachine,1,Answer()
exten = answermachine,n,WaitForSilence(1000,2)
exten = answermachine,n,Background(message)
exten = answermachine,n,GotoIf($[${BACKGROUND}=DETECTED]?replay:exit)
exten = answermachine,n(replay),Playback(message)
exten = answermachine,n(exit),Hangup()
But Background() is looking for a DTMF tone and doesn't even work the way I
described up there. Is there a function that looks for any significant
sound (ie, a BP) that will return and not continue the audio?
On Thu, Jul 17, 2008 at 1:43 PM, Julian Lyndon-Smith [EMAIL PROTECTED]
wrote:
This is what we use, with (seemingly) good success:
exten = answermachine,1,Answer
exten = answermachine,n,Wait(5)
exten = answermachine,n,WaitForSilence(1000,2)
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[asterisk-users] WaitForSilence Problems
I'm trying to write an application for using after an agent has decided the
person on the other end is an answering machine and would like to drop in a
message automaticly. When I'm testing this using my own voice as an
aswering machine, WaitForSilence works correctly and returns only after a
decent delay. But when I hit my real cell phone's voice mail and transfer
the call to the auto-message system, WaitForSilence returns after the given
delay without waiting for it to finish. I'm thinking this has something to
do with the audio level and it not being enough for WaitForSilence to
register. But that really makes no sense.
-- Executing [EMAIL PROTECTED]:1] Wait(SIP/vitelity-09e4f7c8, 1) in
new stack
-- AGI Script Executing Application: (PLAYBACK) Options: (beep)
-- SIP/vitelity-09e4b8c8 Playing 'beep' (language 'en')
-- Executing [EMAIL PROTECTED]:2]
WaitForSilence(SIP/vitelity-09e4f7c8, 2500) in new stack
-- Waiting 1 time(s) for 2500 ms silence with 0 timeout
-- AGI Script Executing Application: (PLAYBACK) Options: (beep)
-- SIP/vitelity-09e4b8c8 Playing 'beep' (language 'en')
-- Exiting with 2500ms silence = 2500ms required
-- Executing [EMAIL PROTECTED]:3] Playback(SIP/vitelity-09e4f7c8,
recordings/360445792) in new stack
-- SIP/vitelity-09e4f7c8 Playing 'recordings/360445792' (language
'en')
-- AGI Script Executing Application: (PLAYBACK) Options: (beep)
-- SIP/vitelity-09e4b8c8 Playing 'beep' (language 'en')
-- Executing [EMAIL PROTECTED]:4] Hangup(SIP/vitelity-09e4f7c8, ) in
new stack
[dropin]
exten = _X.,1,Answer()
exten = _X.,n,Wait(1)
exten = _X.,n,WaitForSilence(2500)
exten = _X.,n,Playback(recordings/${EXTEN})
exten = _X.,n,Hangup()
I added the Wait(1) and Answer() just as an added thing, but they shouldn't
be needed. Anyone have a suggestion?
--
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[EMAIL PROTECTED]
408.497.9796 (c)
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[asterisk-users] AGI Process Count (HOWTO?)
Is there any way to see the number of AGI processes that Asterisk is handling? Either console, Asterisk Manager, or from within the AGI? I used to just count the number of running copies of my AGI process (ps aux | grep agi) but once in a blue moon one of my AGI processes will become a zombie or for some other reason not stop when Asterisk disconnects from it. I'd like to know, from Asterisk's point of view, the number of external applications it's communicating with. -- /Nick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Proxy Issues
For anyone who cares to know. I finally got it working correctly. Turned out I needed fromuser set. Then it was just playing around until it started working. register = [EMAIL PROTECTED]: 0057510:[EMAIL PROTECTED] [voipexten] authuser=0057510 username=0057510 fromuser=0057510 secret=0057510 fromdomain=directnationalloan.com outboundproxy=las-obproxy.voipzone.us host=directnationalloan.com insecure=port,invite qualify=yes type=peer On 1/17/08, Nicholas Blasgen [EMAIL PROTECTED] wrote: I've set up plenty of Asterisk boxes but never one that had to deal with a proxy server to be able to use a line. Using X-Lite I have no issue with settings as follows: Display Name: Any Name User name: 0057510 Password: 0057510 Authorization user name: blank Domain: directnationalloan.com Checked Register with domain and Send outbound via: Proxy Address: las-obproxy.voipzone.us X-Lite has no issues with registration or placing calls. Now the fun part, Asterisk I've been able to get to register. register = [EMAIL PROTECTED]: 0057510:[EMAIL PROTECTED] It's the placing of calls that I'm getting an error. I've tried so many different configurations that it's somewhat pointless to show you my settings. The one I've been playing around with most recently is: [voipexten] auth=0057510:[EMAIL PROTECTED] username=0057510 secret=0057510 fromdomain= directnationalloan.com type=peer qualify=yes insecure=port,invite outboundproxy=las-obproxy.voipzone.us But of corse that doesn't work. Maybe someone here has an idea. -- /Nick -- /Nick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Proxy Issues
I've set up plenty of Asterisk boxes but never one that had to deal with a proxy server to be able to use a line. Using X-Lite I have no issue with settings as follows: Display Name: Any Name User name: 0057510 Password: 0057510 Authorization user name: blank Domain: directnationalloan.com Checked Register with domain and Send outbound via: Proxy Address: las-obproxy.voipzone.us X-Lite has no issues with registration or placing calls. Now the fun part, Asterisk I've been able to get to register. register = [EMAIL PROTECTED]: 0057510:[EMAIL PROTECTED] It's the placing of calls that I'm getting an error. I've tried so many different configurations that it's somewhat pointless to show you my settings. The one I've been playing around with most recently is: [voipexten] auth=0057510:[EMAIL PROTECTED] username=0057510 secret=0057510 fromdomain=directnationalloan.com type=peer qualify=yes insecure=port,invite outboundproxy=las-obproxy.voipzone.us But of corse that doesn't work. Maybe someone here has an idea. -- /Nick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SVN Server Issue?
I'm no longer on the DEV mailing list, but: # svn checkout http://svn.digium.com/svn/asterisk/branches/1.4 asterisk svn: URL 'http://svn.digium.com/svn/asterisk/branches/1.4' doesn't exist http://svn.digium.com/svn/asterisk/branches/ -- /Nick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to setup redundant SIP peers
Some email asked for some examples. He's an example system that
will use ViaTalk lines (which allow 2 concurrent calls on a channel,
so I use GroupCount to check for a value of 2). It isn't round-robin
and actually I'd pay someone good money to make a revised Dial()
function that would do round robbin on a defined set of SIP trunks. I
solve the Round Robbin issue right now by periodicly changing
where SIP/trunk0 ... SIP/trunkN point to and then reloading the
configurations. Periodic for me is midnight each night. So every night the
order in priority order is reset.
*Again, just in case someone from Asterisk-Dev or Asterisk-Bus is reading
this: I will donate/pay to have a Round Robbin outbound trunk balancing
scheme developed. Should be able to use any Asterisk supported trunk type
(SIP, IAX2, etc). No need to care about maximum concurrent connections
since if it fails then we're out of lines anyways.*
[globals]
TRY1=SIP/trunk0
TRY2=SIP/trunk1
TRY3=SIP/trunk2
TRY4=SIP/trunk3
TRY5=SIP/trunk4
TRY6=SIP/trunk5
TRY7=SIP/trunk6
TRY8=SIP/trunk7
TRY9=SIP/trunk8
TRY10=SIP/trunk9
TRY11=SIP/trunk10
[macro-which-line]
exten = s,1,set(TRIES=0)
exten = s,n(nextone),set(TRIES=$[${TRIES} + 1]) ; increment TRIES by 1
exten = s,n,set(DIALSTRING=${TRY${TRIES}}) ; assign TRYn to DIALSTRING
exten = s,n,gotoif($[${DIALSTRING} = ]?donehere) ; see if we've run out
of things to try
exten = s,n,ChanIsAvail(${DIALSTRING}) ; it will be up or down, no need for
this to be exclusive
exten = s,n,gotoif($[${AVAILSTATUS} = 0]?:nextone)
exten = s,n,gotoif($[${GROUP_COUNT(${DIALSTRING})} = 2]?nextone) ; have we
used up the allowed calls on this channel
exten = s,n,set(GROUP()=${DIALSTRING}) ; Lock the line! Yay...
exten = s,n,Dial(${DIALSTRING}/1${ARG1}) ; dial the phone
exten = s,n,GotoIf($[${DIALSTATUS} = BUSY]?donehere) ; Don't keep
dialing
exten = s,n,NoOp(Moving to the next one...);
exten = s,n,goto(nextone) ; TEMP
exten = s,n(donehere),MacroExit() ; we only get here if everything failed
=
Okay, that's one example. Your simple two line thing might be better done
another way. Let's say we try this:
1) Place a call two two phone lines at once, but have a single line delayed
by 3 seconds.
Something to this effect:
Dial(SIP/trunk0Local/delayed_trunk1)
Where Delayed_Trunk1 is a macro which calls SLEEP(3) and then Dial(Trunk1).
I could go into more detail, but I'm going to assume you can figure out how
to do this.
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Re: [asterisk-users] Realtime sip.conf
I don't understand the USERS vs PEER vs FRIENDS. I just use Peer for everything. Has to do with can I only contact you or can you contact me too? ... Peer does it all. RealTime does have an issue. If you don't turn on caching, then it holds no state information. So if you think you're going to encouter firewall issues and need NAT=yes, then realtime will run in a static mode where you'll need to reload each time you change anything (like a password). I think the proper command is something like SIP PRUNE. Finally, putting something like sip.conf into realtime wasn't a move I wanted to make. I simply generate a SIP.conf file myself via my own program and run a SIP RELOAD (or simply reboot) each time I make a big change. Changes don't happen often so no biggie, where as I did want to make live changes to other SIP users without reloading (like a person using our web interface to change their own password). On 12/29/07, hugolivude [EMAIL PROTECTED] wrote: Hi - I'm looking into realtime and I'm having a bit of a problem with the SIP part. My review of the posts seems to indicate that I should use realtime static for the [general] part of my sip.conf including the registration commands: register=did:secret@domain/did context and use realtime realtime (funny name!) for peers and friends: [myprovider] type=peer auth=md5 username=... fromuser=... fromdomain=... secret=... host=... port=5060 nat=yes canreinvite=yes qualify=no disallow=all allow=ulaw dtmfmode=rfc2833 insecure=port,invite context=incoming-sip Is this correct? What's throwing me off is this statment found @ http://www.voip-info.org/wiki/index.php?page=Asterisk%20RealTime%20Static: NOTE: You can only store a static config OR a RealTime config. You cannot, for example, store sip.conf and use sipfriends via RealTime. If I am correct, it would suggest that I'll have to do a reload when I add a DiD, but a reload won't be necessary if a new SIP client is added. Do I have it right? Also, what's the difference between a peer and a user? I used to think that a user was an agent authorized to call in to my * box, a peer was an agent I could reach and a freind was both. What's throwing me off now is the statement found @ http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peerview_comment_id=14966 : With newer versions of Asterisk the concept of SIP 'users' will be phased out. I can't understand this especially in the context of extconfig.conf that uses both a sipuser and sippeer entry. Could someone clarify for me? Thanks, H ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PHP AGI script
I've got a very nice PHP AGI script but I want to be able to do some database cleanup when the user hangs up the phone. I wish everyone would hang up when they were suposed to, but some people don't. So what does Asterisk send to an AGI file when the line has been disconnected? If I remember reading somewhere correctly, I don't need to use DeadAGI. Instead I'm able to use normal AGI but I just need to catch a SIGTERM or something like that and process it. Does anyone here have any PHP examples of this, maybe? -- /Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Trunk Problems
It gets hard to read my logs when every time someone makes a phone call it displays long pages of Dropping voice frame. Anyone encounter this before? Asterisk is bridging two SIP lines together, so the technology should be the same. Maybe I'll try allowing only ULAW. ** Asterisk Standard debug (level 3) *** -- Called trunk2/12095387895 [Nov 26 13:49:37] WARNING[7744]: channel.c:3021 set_format: Unable to find a codec translation path from unknown to unknown [Nov 26 13:49:37] WARNING[7744]: channel.c:3402 ast_channel_make_compatible: Unable to set read format on channel SIP/trunk2-0990c538 to 524288 -- SIP/trunk1-098dc208 is making progress passing it to Local/[EMAIL PROTECTED],2 -- Local/[EMAIL PROTECTED],1 is making progress passing it to SIP/trunk0-098cf098 [Nov 26 13:49:38] NOTICE[7062]: channel.c:2493 __ast_read: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],1 of format ulaw since our native format has changed to unknown [Nov 26 13:49:38] NOTICE[7062]: channel.c:2493 __ast_read: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],1 of format ulaw since our native format has changed to unknown [Nov 26 13:49:38] NOTICE[7062]: channel.c:2493 __ast_read: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],1 of format ulaw since our native format has changed to unknown [Nov 26 13:49:38] NOTICE[7062]: channel.c:2493 __ast_read: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],1 of format ulaw since our native format has changed to unknown ** SIP.CONF Example Line *** [trunk0] authuser=191691245XX username=191691245XX fromuser=191691245XX secret=12345 fromdomain=richmond-1.vtnoc.net host=richmond-1.vtnoc.net dtmf=auto dtmfmode=inband insecure=port,invite qualify=yes type=peer canreinvite=yes call-limit=2 context=viatalk -- /Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialing an external number and then passing it to an extension...
Okay, you need to explain to me a little more about why you're calling the
list before connecting it to an extension.
So for me, I use a .CALL file but I assume your setup does the same thing.
It will call a number and once ANSWERED, pass it to an extension. Let's say
we pass it to a LOCAL channel and have that then passed to the extension.
Have the LOCAL be a bunch of Dial() strings. Humm, that will only go to the
first line that gets answered--it wont check to see if it's a fax or
answering machine. So that doesn't work.
You could so some complex thing with the manager. Connect a single call to
an extension that decides if it's an answering machine, fax, or human. If
human, it can do a GOTO command. If not human, it can set some type of
GLOBAL var and hangup. When the Manager detects the EVENT HANGUP it can
see if it made that call recently, if it did, then it can go to the next
number in the list. Repeate. That does it, but it's not great.
I'd like to know more about why you're doing things the way you're doing
them. There might be a better way.
On 9/20/07, Carlos Chavez [EMAIL PROTECTED] wrote:
I am in need of some guidance regarding the following problem:
I need to dial an external number from a list(PSTN)
I need to check if the number is busy, no answer or fail
If any of the above are met then I try another number from a list
If none of the above happen then I first need to determine if the line
answering is a fax machine or an answering machine
If fax or answering machine then hangup and try next number
If human then connect to an internal extension
An outbound callcenter suite is overkill since we only need two or
three
calls at a time. Can something like this be done using the Originate
command
on AMI? The main problem I have is that if I dial an external call and it
fails for some reason how do I know? Is there something like
${DIALSTATUS}
that can give me the result of that part of the call?
We plan to have a web interface that will fire the call when you click
a
button. That will fire an event that connects to the manager interface
and
uses originate to dial the external call and then dial the internal
extension
if all conditions are met. The numbers will be in a database.
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
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Re: [asterisk-users] call limit
Setting call-limit=1 in sip.conf will limit the number of incomming (and outgoing?) channels on your SIP device to the number you specifiy (1 in this case). If you want to allow more outgoing, but only 1 incomming, you could do that with some GROUP() checking. Problem is that when there isn't an available channel, Asterisk will return CHANUNAVIABLE or something like that. It's not very helpful. GROUP() checking will allow you to provide a more informative answer to the person who's calling. On 9/21/07, Vieri [EMAIL PROTECTED] wrote: Hi, I would like to know if the following is possible: * how to accept only one call at a time on a particular SIP extension (softphone). I'm referring to incoming calls. Can it be done on the server side or just on the client? ie. all other incoming calls will just be dropped while the extension is busy. In other words I would like to simulate having just one phone line available. I tried using call-limit=1 in sip.conf. Is this the right way? * how to accept only one incoming call at a time for a whole group? If there's an active call on any one of the extensions, drop the other incoming calls. Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online. http://smallbusiness.yahoo.com/webhosting ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /Nick ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging to external speaker like in airports etc...
Once upon a time it cost $20/hr over a 9600 baud link to read stuff like this and people tended to think before they asked questions, I'm afraid they didn't, I vividly remember asking inane questions at 1200baud over uucp ;-) Ditto. I still have newsgroup posts of mine asking about setting up Wildcat 4.0 BBS with some very silly questions attached. Those were the days before we had PPP. But luckily I think my internet fee was only $3/hr for dialup :) ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error messages related to mysql in asterisk CLI
It would surprise me to see mySQL configured incorrectly, but it's always a possibility. Look at the mysql server var called 'wait_timeout'. phpMyAdmin shows it under system vars. http://blog.taragana.com/index.php/archive/mysql-tip-mysql-server-has-gone-away-or-lost-connection-to-server-during-query-fix/ On 9/22/07, Jody Gugelhupf [EMAIL PROTECTED] wrote: hi there :) i get this error in the asterisk CLI: Sep 21 20:31:49 ERROR[3774]: cdr_addon_mysql.c:159 mysql_log: cdr_mysql: Unknown connection error: (2006) MySQL server has gone away ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GROUP() issues for me
I've got a macro that tries to find the first available SIP trunk to send
outgoing calls on. It tracks the usage of the lines (since each trunk has a
call-limit of 2) by using GROUP(). My problem is that once a call switched
to ANSWER state, ``group show channels`` stops listing it and then my Macro
starts screwing up because it's sending calls to a line that sometimes is
full even though GROUP() shows it as being less than 2. I'm tempted to send
this to the Asterisk Dev team just because I believe it's an issue of the
GROUP information being released when Asterisk consolidates the channels
(removes all the MASQ channels) once the call is answered. But maybe it's
something else so I'll ask here first.
The dialplan setup:
exten = 555,1,Dial(Local/1234567890)
exten = _NXXNXX,1,Macro(which-line,${EXTEN})
[macro-which-line]
exten = s,1,set(GROUP()=${DIALSTRING})
exten = s,n,Dial(${DIALSTRING}/1${ARG1})
Things are a bit more complex, but it's all just logic. The extensions
above should give a decent representation of what's going on. I think each
time you switch extensions, Asterisk creates a MASQ channel and that's
what's causing the issue since the GROUP() is set only at the end, inside
the macro. Are there any EVENTS for unlocking of GROUPs? Anything I can do
to better show where this is happening?
I'd love some help if anyone has a guess.
--
/Nick
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Re: [asterisk-users] GROUP() issues for me
exten = 555,1,Dial(Local/1234567890/n) note the /n I'm going to try this in a bit (can't hurt anything, might as well), but I'd like to understand you're reasoning. You're dialing an extra extension? I'm also going to be trying this with Asterisk 1.6 TRUNK to see if it's even a current issues in the development branch but I wont have a chance untill tomorrow sometime. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.24 simultaneous call limits.
Just thinking about it quickly, it's always possible it has nothing to do with Asterisk. There are many instances where I run into issues with a poorly configured servers when they have even a little bump in HTTP traffic. This was years ago though, and it was an issue to do with a web server and not Asterisk, but look into your kernel's configuration. Sometimes the kernel's settings are setup for a normal USER and not designed to handle the memory allocation a server demands. The fix for me back then was something to do with the MAXIMUM PAGE REQUESTS or SIZE maybe. Basicly the kernel couldn't keep track of all the HTTP processes. Now that I'm reading this over I doubt it's your problem because Asterisk doesn't fork. But while we're at it, tell me a bit more about your system. What operating system (and version)? The problem could also be with your method of load generation, but I wouldn't know that since I've never tried load testing a system. Lastly, I know FreeBSD started incorporating a basic DDoS protection a few years back and maybe that's also in some of these newer Linux distros. They would detect a flood and start to limit the bandwidth. These are just ideas, I don't really like any of them. Sometimes the kernel will report issues to SYSLOGD. Might want to check your error and message logs. cat /proc/meminfo On a Linux box will give you memory limits and how close you are to them. They're not exactly what I was looking for, but maybe that will help. All TCP connections require the Kernel to page the information but I can't seem to find out how to access that limit if any. On 9/20/07, Wai Wu [EMAIL PROTECTED] wrote: Hi everyone, I am running into wall today with simultaneous call limits. I have two Asterisk machines (fast 3GHz C2D with 2GB of ram). I tried to create a lot of sip calls from one machine to the other by issuing AMI Originate commands to one machine. The machine that makes calls plays a message (demo-intruct) upon the other machine answer. The machine receives the calls just waits for 40 seconds then hangs up. Throught the manager connection, I was creating 10 calls per-second. I also have sip phone registered with the calling machine. At around 150 to 200 calls. When I call the machine that's making all the calls, most of the calls couldn't go through. For the ones that went through, most of them will drop off within seconds of the call. But here is catch. When I run 'top', the cpu is idling 97%. My question is. Is there a limit on the number of simultaneous calls Asterisk can handle? I know I have very fast systems. Shouldn't they be able to handle that many calls? What is your take? Thnx ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /Nick ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Remote extension search?
I've heard about this, but I really can't seem to find anything on it. I've got a strange setup that exists only because of firewall issues, and everything about it seems fine. The setup: SIP clients - Asterisk (office) - IAX - Asterisk (colocation) - SIP PSTN Termination All the extensions I want to be able to dial are on the colocation box. What I'd really like is for the office asterisk box to forward all extension requests it doesn't know about to the colocation Asterisk box. I think this is refered to as Trunking. I only need to do this in a single direction, if that's any easier to setup. Are there any good documents on VOIP-Info or another site on setting up something like this? The office Asterisk's job is just to act as a SIP to IAX gateway. I've got a work-a-round that will work, but I thought I'd learn the proper method. -- /Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Events
At least with my Manager API, I have the ability to simply set a default event handler and using that I can dump all events as the pass though. Then I setup a case switch and act on the ones I want. But the manager events I like are LINKED and HANGUP. http://www.voip-info.org/wiki/view/asterisk+manager+events On 8/15/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi All, Can anybody send me a complete list of sip events. i know only 3 of those whihc are register, message-summary, message notification. message-summary event is causing some problems actually. My client sends a bad-event response if it recieves a message-summary event in a NOTIFY sip packet. I have little knowledge of sip events. So if anybody knows a good link plz share. And if u know how to fix the bad event message then plz tell also. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Events
Ah, I correct myself. I see, you wanted to know the headers for each SIP packet. Makes a lot more sense now. On 8/15/07, Anthony Francis [EMAIL PROTECTED] wrote: http://www.faqs.org/rfcs/rfc3261.html Rizwan Hisham wrote: Hi All, Can anybody send me a complete list of sip events. i know only 3 of those whihc are register, message-summary, message notification. message-summary event is causing some problems actually. My client sends a bad-event response if it recieves a message-summary event in a NOTIFY sip packet. I have little knowledge of sip events. So if anybody knows a good link plz share. And if u know how to fix the bad event message then plz tell also. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com http://www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan / AGI autoanswer question
So besides the missing ) on line 1, I have some other comments: 1) You should replace your priority numbers with 'n'. Just so much easier to know that the issue isn't with priority numbers. And typing 'dialplan show context' is a nice way to see if everything is setup correctly. The 'n' is a personal choice, but the longer your application the better. 2) I thought I read somewhere that AGI was now auto-answering the channel. But I guess that's not right. AGI will auto-answer the channel if something causes it to do so. If you want to post your AGI code without any database commands, I'll glance at it. humm, I guess that's all I see. Everything else seems fine. It may be good to check the ChannelStatus once in a while just to debug where the channel is getting answered. http://gundy.org/asterisk/agi.html ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Load balancing SIP trunks?
I have 10 SIP trunks that I'd really like to round-robin load balance. Currently I have a macro that switches between available lines, but there really must be a function in Asterisk to do this on its own. So my question is just that, are there any easy ways for Asterisk to either balance between SIP trunks or even just a built in function to find the next available SIP trunk. I think using a macro to test the state of each trunk is silly, but it's the only method I've found. -- /Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan / AGI autoanswer question
First, it seems I have to have a 2 - 3 second wait before the AGI call in order to get valid CID data. Usually 2 seconds suffices for this one setup but during that time the caller has had two rings before the local extension has even begun to ring. Is there something I am doing wrong that causes it to take so long to get the CID? Check out the CDR configuration. I do my CDR via MySQL and I don't think that does buffering, but I know for sure the normal CSV format (and standard configuration file) has options for buffering before saving. I can't really think how that would change recieving the CDR information during a call, but it's possible something along those lines. I'm making it up really since the more I think about it the less that sounds possible. But search the documentation on the CDR or maybe ask the Asterisk Dev group about when the CDR fields get filled. It might even be possible that CDR infromation isn't accessable untill the line has been answered and that's the delay. Finally, if I call from a remote site, it goes to voicemail, I hang up before leaving a message, and then quickly call right back again I get what sounds like a fax tone. I'm not specifying anything about faxes in the extensions.conf and zapata.conf has all the fax stuff commented out. My voicemail extension looks like What is the channel type to the remote site? IAX, SIP, analog? I've seen alarm systems that pick up analog lines they're attached to if thre are back to back calls. That's the best I can come up with there. Maybe someone else has a better idea. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Macro Overlap
I've got 4 SIP phone lines with a call-limit of 2 for each. I've written a handy macro to allow my users to dial a phone number and the macro will figure out the next available line to use by first checking if the GROUP() is over 2 and then checking to see if ChanIsAvail() as a backup, and if it can't use the line for either reason it goes to the next line. The problem is that there are enough situations that the Macro gets called twice without much time seperation. Both macros check the group() number, it comes back as free, they check the line availability and it's open, and they try dialing. But because they both started at more or less the same instant, they've both at the same stage in the macro and sometimes (maybe 10% of the time) a macro will try dialing on a line that's already in use. My question is this. Is it possible to tell Asterisk to execute part of a macro as a block without allowing any other commands to be processed during that time? Some way to LOCK the dialplan (as you'd do in SQL). I want my macro to be able to execute the part of the code that checks line status and then sets the GROUP() without allowing any other dialplans from running during that time. Anyone know if this is a current feature? -- /Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Macro Overlap
I'm going to try it out, but I'm not very hopefull although it's exactly what's needed. My macro contains a Dial() command and my concern is that the dialplan isn't considered done untill Dial() returns. But I'm going to try it. Will report back shortly. On 8/7/07, Jared Smith [EMAIL PROTECTED] wrote: On Tue, 2007-08-07 at 11:13 -0700, Nicholas Blasgen wrote: My question is this. Is it possible to tell Asterisk to execute part of a macro as a block without allowing any other commands to be processed during that time? You'll want to check out the MacroExclusive() application. It does exactly what you're looking for. If I remember correctly, it's new in Asterisk 1.4. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outbound dialing
Not specific to the SPA3102, but just normal outbound dialing is as follows:
exten = _1NXXNXX,1,Dial(trunk type/name/${EXTEN})
or if you want to require people to dial 9, then:
exten = _91NXXNXX,1,Dial(trunk type/name/${EXTEN})
or if you're like me and you're used to a cell phone and don't like dialing
the 1:
exten = _NXXNXX,1,Dial(trunk type/name/1${EXTEN})
On 8/7/07, Tim Johnson [EMAIL PROTECTED] wrote:
Hello all. I am just getting back into Asterisk and I am setting up my
Linksys SPA3102. I have incoming calls working fine, as is the phone
plugged into the unit. My problem is I cannot get the SPA3102 to dial
a phone number automatically. I can call the extention of the PSTN and
I get a second dialtone, and I can then manually dial. I'd like to be
able to have Asterisk pass the number I dialed to the SPA and have it
dialout. I've played with dialplans on the SPA I've found during my
googling, but I think it might be something I am missing in my
extentions.conf file. Any ideas?
Tim
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[asterisk-users] SIP Max Channels Setup
I'm running Asterisk without FreePBX or any of the other managers. I'm trying to figure out how to set the maximum number of channels allowed on a single line? I'd just rather not have Asterisk try the line when I know I'll recieve a CONGESTION back from the SIP phone provider (ViaTalk in this case). Is there a configuration option I can't find that sets the maximum number of connections a SIP channel can handle at a given moment? I expect the line to be something simple, but I can't find it detailed on the Wiki. -- /Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
