Re: [asterisk-users] Asterisk Not Starting after YUM Update - Solved
On 13 Feb 2014, at 09:55, Aldo Bergamini aabe...@gmail.com wrote: Hi, I did compile the latest DAHDI and LibPRI, with no success… So I thought about updating the Asterisk package to the last known 1.6.2 release. Now it's crashing at some different point. This is the the strace result: Hi all, thanks for the hints. I did solve the problem… After reinstalling both DAHDI and LibPRI I first did recompile Asterisk, using the 1.6.2 release tarball. This proved to still make fuss, so that I redid the installation, this time with the latest 1.6.0 code. Aside complaining about some incompatible modules, the PBX went up well. I did even reload the binaries for G729, Skype and the fax stuff. They do not activate themselves (as an aside from the update nightmare, I had to replace an ethernet card), but this is probably due to host identification fingerprinting… Does anybody know if it is still possible to receive a 'move to new host' authorisation from Digium for the now unsupported Skype bridge? Thanks and best regards, Aldo -- _ -- 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] Asterisk Not Starting after YUM Update
On 12 Feb 2014, at 23:22, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: Try: # standard asterisk command-line. No verbosity strace -eopen asterisk -U asterisk -c See which module was the one last loaded. -- Tzafrir Cohen Hi, I did compile the latest DAHDI and LibPRI, with no success… So I thought about updating the Asterisk package to the last known 1.6.2 release. Now it's crashing at some different point. This is the the strace result: [root@hermes sources]# strace -eopen asterisk -U asterisk -c open(/etc/ld.so.cache, O_RDONLY) = 3 open(/lib/libssl.so.6, O_RDONLY) = 3 open(/lib/libcrypto.so.6, O_RDONLY) = 3 open(/lib/libc.so.6, O_RDONLY)= 3 open(/usr/lib/libxml2.so.2, O_RDONLY) = 3 open(/lib/libz.so.1, O_RDONLY)= 3 open(/lib/libm.so.6, O_RDONLY)= 3 open(/lib/libdl.so.2, O_RDONLY) = 3 open(/lib/libcap.so.1, O_RDONLY) = 3 open(/lib/libpthread.so.0, O_RDONLY) = 3 open(/lib/libtermcap.so.2, O_RDONLY) = 3 open(/lib/libresolv.so.2, O_RDONLY) = 3 open(/usr/lib/libgssapi_krb5.so.2, O_RDONLY) = 3 open(/usr/lib/libkrb5.so.3, O_RDONLY) = 3 open(/lib/libcom_err.so.2, O_RDONLY) = 3 open(/usr/lib/libk5crypto.so.3, O_RDONLY) = 3 open(/usr/lib/libkrb5support.so.0, O_RDONLY) = 3 open(/lib/libkeyutils.so.1, O_RDONLY) = 3 open(/lib/libselinux.so.1, O_RDONLY) = 3 open(/lib/libsepol.so.1, O_RDONLY)= 3 open(/etc/selinux/config, O_RDONLY|O_LARGEFILE) = 3 open(/proc/mounts, O_RDONLY|O_LARGEFILE) = 3 open(/dev/urandom, O_RDONLY|O_LARGEFILE) = 3 Asterisk 1.6.2.24, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer marks...@digium.com Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = [ Booting... [ Reading Master Configuration ] open(/etc/asterisk/asterisk.conf, O_RDONLY|O_LARGEFILE) = 4 open(/etc/nsswitch.conf, O_RDONLY)= 4 open(/etc/ld.so.cache, O_RDONLY) = 4 open(/lib/libnss_files.so.2, O_RDONLY) = 4 open(/etc/passwd, O_RDONLY) = 4 open(/proc/sys/kernel/ngroups_max, O_RDONLY) = 4 open(/etc/group, O_RDONLY)= 4 Unable to access the running directory (Permission denied). Changing to '/' for compatibility. open(/usr/share/terminfo/x/xterm, O_RDONLY|O_LARGEFILE) = 4 [ Initializing Custom Configuration Options ] open(/etc/asterisk/extconfig.conf, O_RDONLY|O_LARGEFILE) = 4 open(/etc/termcap, O_RDONLY) = 4 open(/root/.asterisk_history, O_RDONLY|O_LARGEFILE) = -1 EACCES (Permission denied) open(/var/run/asterisk.pid, O_WRONLY|O_CREAT|O_TRUNC|O_LARGEFILE, 0666) = 4 open(/etc/asterisk/logger.conf, O_RDONLY|O_LARGEFILE) = 5 open(/var/log/asterisk/messages, O_WRONLY|O_CREAT|O_APPEND|O_LARGEFILE, 0666) = -1 EACCES (Permission denied) Logger Warning: Unable to open log file '/var/log/asterisk/messages': Permission denied open(/var/log/asterisk/event_log, O_WRONLY|O_CREAT|O_APPEND|O_LARGEFILE, 0666) = -1 EACCES (Permission denied) open(/etc/localtime, O_RDONLY|O_LARGEFILE) = 5 open(/etc/localtime, O_RDONLY)= 5 open(/var/log/asterisk/queue_log, O_WRONLY|O_CREAT|O_APPEND|O_LARGEFILE, 0666) = -1 EACCES (Permission denied) [root@hermes sources]# It's a long time I do not compile Asterisk; compiling from root does set a specific user under which to run the binaries? Does it run as root? Another idea I had is if the presence of now outdated binaries for fax, Skype and G729 (all from Digium, but made for Ast 16.0.x) could cause the problem.. Thanks in advance! Aldo-- _ -- 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] Asterisk Not Starting after YUM Update
Hi List, it feels silly, but here I am. My Asterisk box is useless, after running a long delayed yum update (Centos box). * A few details on the box: cat /etc/redhat-release CentOS release 5.10 (Final) arch i686 uname -a Linux hermes 2.6.18-371.4.1.el5 #1 SMP Thu Jan 30 06:09:24 EST 2014 i686 athlon i386 GNU/Linux asterisk -r Asterisk 1.6.2.20, Copyright (C) 1999 - 2010 Digium, Inc. and others. * Starting Asterisk very verbosely seems to load the dialplan, but at some point I get a segmentation fault. This is new to me! […] edited […] chan_agent.so = (Agent Proxy Channel) == Registered custom function 'EXTENSION_STATE' func_extstate.so = (Gets an extension's state in the dialplan) == Registered application 'DAHDIBarge' app_dahdibarge.so = (Barge in on DAHDI channel application) == Registered custom function 'CALLERPRES' == Registered custom function 'CALLERID' func_callerid.so = (Caller ID related dialplan functions) [2014-02-12 22:40:07] NOTICE[13842]: codec_g729a.c:760 load_module: G.729A transcoding module version 1.6.0_3.1.4, Copyright (C) 1999-2009 Digium, Inc. [2014-02-12 22:40:07] NOTICE[13842]: codec_g729a.c:761 load_module: This module is supplied under a commercial license granted by Digium, Inc. [2014-02-12 22:40:07] NOTICE[13842]: codec_g729a.c:762 load_module: Please see the full license text supplied by the accompanying [2014-02-12 22:40:07] NOTICE[13842]: codec_g729a.c:763 load_module: register utility, or ask for a copy from Digium. Segmentation fault The problem seems to come after the callerid module loads: does this make sense? BTW: I do have a G729 pack of licenses (they were actually active without any problem before messing with the update).. What should the clever sysadmin do? Thanks in advance, Aldo -- _ -- 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] Dedicated hangup extension h
On 28 Aug 2013, at 09:50, Grant Bagdasarian g...@cm.nl wrote:
Hi Grant!
I do not know of a way to have multiple 'h' extensions in the same context.
But you can easily make an appropriate context for your custom need!
exten = _X.,1,Playback(invalid)
exten = _X.,n,Hangup
exten = 1000,1,Noop(Go to CURL Hangup)
exten = 1000,n,Goto(CURL_Hangup,${EXTEN},1)
; . your context goes on
; Added Custom Context
[CURL_Hangup]
exten = _X.,1,Playback(welcome)
exten =_X.,n,Read(dtmfinput,15)
exten =_X.,n,Hangup
exten =
h,1,Set(response=${CURL(http://sample.company.local/PostHandler.ashx,var1=${dtmfinput}var2=1000)})
HTH,
Aldo
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Re: [asterisk-users] Conf into a call in progress
On 15 Nov 2012, at 15:44, Michael wrote: Hi Aldo, Thank you very much for answering my question. Can you kindly elaborate on how to do the following or at least where to read about the way to do it? Hi Michael, sure... I am sending you -by direct mail- a diagram that tries to illustrate what I would try to do. (I do not know if this list allows attachments; generally it's not permitted...). send both channels of the active call 111 - 22334455 to a context that joins them in a conference room. AMI has a useful command for that task: Redirect, see here: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Redirect and here: http://www.voip-info.org/wiki/view/Asterisk+manager+Example:+Transfer If you are manipulating a call not from one of the connected terminals (e.g. your phone) you have to take care of both channels. This is what the Redirect command does. It lets you specify what to do with both channels: they can be sent to the same context or each to one context by itself. Finally you are able to make changes on one channel only... through AMI, I would originate the call to 22556677 and join it into the conference. So the plan would be to first send the two channels to a conference room (an ad hoc one), using a first redirect command. This is made to get the conf. room where the three way call will take place AND to be able to call the third party without losing the original call partner's channel... A second Redirect command should detach the user's channel from the conference and send it to a context that connects him/her to the third party, letting the original user offer the 3 way call. If the call is accepted, than a third redirect would send both channels to the conference room created at step 1, where the other party is waiting... The dynamic conference is closed either by the original call party hanging up his/her channel or with a direct AMI hangup command doing the same thing. Clearly this is logically equivalent to a manual transfer of the user's call party into a conference room. Then calling the second call party and transfer him/her to the conference and seeing the user finally dialing him/herself into the conference. You can do that with AMI, provided you have some means to make some sort of UI for the whole process... Thank you very much, Michael You're welcome: hth! Aldo -- _ -- 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] Conf into a call in progress
On 15 Nov 2012, at 14:21, Michael wrote: Hello, Does anyone know if it's possible to setup the following scenario? 1. A specific ext(let's say 111) is on active call with an external number via SIP (let's say 22334455). 2. Via a web GUI, send to asterisk another phone number (22556677) and the ext number (111). 3. Asterisk initiates a call to that number (22556677) and joins it to the call in progress (between 111 and 22334455) in order to establish a 3-party conf call. It's somewhat similar to ChanSpy, but with full conf capabilities and not only whisper to one side. Thanks, Michael Hi Michael, I would use a combination of AMI dialplan programming. Over AMI I would send both channels of the active call 111 - 22334455 to a context that joins them in a conference room. It is a matter of choice if it is better to create an ad hoc/ on the fly conference or use a set of predefined rooms. Next, again through AMI, I would originate the call to 22556677 and join it into the conference. You have to be aware that calling somebody and transferring the channel into a conference may leave the person on the other side of the wire WITHOUT means to exit the conference room and thus to close the call (I did it!!! embarrassing..). So one has to be sure (I am speaking of the old MeetMe app) that the originator's channel enters the conference room as the conference master. So, when that channel closes, all other channels are dumped out of the conference room and the whole thing closes down. HTH, Aldo -- _ -- 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] Impromptu conferencing
On 8 Nov 2012, at 10:46, martin f krafft madd...@madduck.net wrote: For a 3 way conference, all those days phones are able to do this. Yeah, except I want Asterisk to handle that, not my phone (which might lose reception or run out of battery etc.). Martin, I understand that having the feature in the PBX makes it uniform to all phones, makes it available to all users, etc. But it is not easy to do this inside the dialplan: the biggest problem is that you basically miss any kind of GUI to handle a set of steps that can each require a decision, to tell Asterisk what to do e.g. if the third person is busy and sends you to voicemail, having just a dialpad in your hands, with no visual feedback ... Having complex functions in Asterisk is than linked with instruction sets for users (that should memorize them) in form of strange dialing sequences.. The best way to handle your idea is probably done through some kind of application to be used during calls.. Aldo -- _ -- 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] Counting calls in progress from AMI
On 18 Oct 2012, at 17:19, Michelle Dupuis wrote:
I need to do this from the AMI (not the CLI)...I don't *think* a comparable
command exists from the AMI.
As well, I don't want to poll the system for calls so I'm hoping to trap a
call bridged,unbridged type event.
Michelle,
if you do not want to poll Asterisk with an AMI 'Status' event (that returns a
list of open channels) counting the bridged, unbridged events is possible, but
not that easy..
There are two gotchas around this:
One is that a single call, depending of what the dialplan is doing, can involve
several bridged, unbridged events;
The second is that some kind of calls (I call them 'one legged calls': anything
resembling an IVR or a call to get voicemail) do not get any bridged, unbridged
events, at all. The same applies for calls that are sent to a MeetMe
conference: there you see specific MeetMe events for conference rooms.
If you have access to the dialplan (that is if you are in charge of it and you
can modify it) you could add some user defined events, to mark the 'rising' of
a call and its connection to the far side. The dialplan can send pretty much
what you like by using UserEvent.
As an example:
exten = _00.,n,UserEvent(DNIS-Ext,Exten: ${EXTEN},CallerID:
${CALLERID(num)},DNID: ${CALLERID(dnid)},DisplayID:
${PersonalID_Num},ChannelID: ${CHANNEL},RDNIS: ${RDNIS})
The first string (DNIS-Ext) is a marker: anything that you want to receive that
'brands' the user event to your suiting.
The rest is a list of (name, value) pairs, giving the details you might need on
the AMI processing side.
Beware of any loop in the dialplan: it might be that you will get multiple
copies of the same event, if the dialplan execution is such that the same
extension is visited more than once during the processing of a call.
HTH,
Aldo
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Re: [asterisk-users] Question on AMI and ChanIsAvail
On 18 Oct 2012, at 17:58, Jerry Geis wrote: I was wanting to call ChanIsAvail from AMI. I logged in and issues command, Action: Command Command: ChanIsAvail DAHDI/1 my response was this: event_list=0 ret=158 Response: Follows[CR ][LF ]Privilege: Command[CR ][LF ]No such command 'ChanIsAvail DAHDI/1' Is there any way to tell if a channel is available through AMI? Did I format my request wrong? I am on 1.4.43. Thanks, Jerry You can get (on the CLI) the list of AMI commands supported by your Asterisk installation issuing manager show commands. The AMI command that tells you what a terminal is doing at any time is ExtensionState (I guess it's available on 1.4..). But if you add hints to your dialplan, then Asterisk will send an AMI event whenever a terminal in the list of hints is changing state. You would need to listen to all such events and keep track of each line/extension; issuing a list of ExtensionState events only when you launch your tracking process. Cheers, Aldo -- _ -- 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] 100 outgoing calls - 10 at a time - /var/spool/asterisk/outgoing/
On 28 Sep 2012, at 13:27, Leif Madsen leif.mad...@asteriskdocs.org wrote: Generally the preferred method when you're doing this programatically anyways is to use an external script through the Asterisk Manager Interface to generate your calls. Luckily, Russell Bryant has recently create an amioriginate.py[0] script which he's using as an example in the upcoming Asterisk: The Definitive Guide 4e book. [0] https://github.com/russellb/amiutils -- Leif Madsen http://www.oreilly.com/catalog/asterisk Hi Leif, I am happy to hear that a new release of The Book is in the works! I will have a look at Russell's script as soon as I am back at my work chair: there is however something I am very curious about: it is how you ask Asterisk, over AMI, to launch an external command, script, etc. I was (falsely) assuming that you need a channel to launch a script upon.. To be able to trigger 'commands' over AMI, before any channel exists, opens immense possibilities!! TIA, Aldo Sent from my iPhone -- _ -- 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] 100 outgoing calls - 10 at a time - /var/spool/asterisk/outgoing/
On 28 Sep 2012, at 14:24, Leif Madsen leif.mad...@asteriskdocs.org wrote: That's good! I'd hate to be working on something no one wanted :) ;-))) Oh heck ya. You can start up an Asterisk instance and just start doing things with it via your programs. That's the immense power of AMI; it's essentially the Asterisk API. Yes, I know... I have been writing an Objective-C library to control Asterisk over AMI, but I did not see the event that you need to send to just trigger a script, say, to read the log files and get the last calls, or anything similar. I can originate calls with an AMI Originate command and some dialplan glue, but I am missing anything like an Exec AMI command. -- Leif Madsen http://www.oreilly.com/catalog/asterisk TIA, Aldo Sent from my iPhone -- _ -- 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] Check for the voicemail
On 21 Aug 2012, at 19:32, Ruben Rögels ruben.roeg...@jumping-frog.org wrote: Hello, no problem at all, I think this is the tricky part. A smtp dialogue between your email client and a smtp server normally looks like this: user@box:~? netcat mx1.example.com 220 postfix ESMTP mx1.example.com helo me.local 250 mx1.example.com mail from: ruben.roeg...@wiseape.de 250 2.1.0 Ok rcpt to: ruben.roeg...@example.com 450 5.7.1 ruben.roeg...@example.com: Mailbox Full The tricky part is writing or finding a console smtp client that gives you feedback about the 450 error that just happened. Right now I cannot give you a precise way to do that, but I have basic understanding of the technology, so I know that it is possible to do so ;-) I'm looking around in the net, because I think I'll soon have to handle your problem aswell in my company ;-) If I can find solution, I'll post it. regards, Ruben It is not very difficult to write an ad-hoc script in a language like Python and call it instead of the regular sendmail command. Just look up something 'Python smtp send tutorial' and you should get a good starting point. Regards, Aldo Sent from my iPhone -- _ -- 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] Asterisk Timezones
On 12 Apr 2010, at 22:14, asterisk-users-requ...@lists.digium.com wrote: There's system clock, and hardware clock. Whatever you get for the localtime when you do 'date' command is what you're going to get for logs from asterisk. It seems somewhere you have your system set to run in GMT, even though you don't want it to be like that. You will need to consult documentation about properly setting your clock for your timezone. The alternative is to leave your system 'broken', and change your time checks to GMT. Hello David, it was luckily easier than that! I checked with an extension in my DP that reads me the 'Asterisk time' and it was correct. At least after I tinkered a little with the time zones settings of the OS, but still with strange CDR times. So that I went looking into the CDR config file and noticed that I had the default choice of using GMT time for the CSV logs. So, no surprise that Asterisk was doing it... Thanks and best regards, Aldo -- _ -- 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] Asterisk Timezones
Hi all, I have noticed something I can't solve regarding Asterisk (latest 1.6.0.x). My server is set at the GMT+2 timezone. The clock is ok (I can get the correct time at the terminal). But today I got a call at a time where Asterisk should have gone 'off business hours'. All log times are wrong by exactly 2 hours. As if Asterisk would just sit on GMT, ignoring the GMT+2 timezone. I have looked around and I do not have found any information about how to set the log/system timezone. The only place I remember having a reference to timezones is the voicemail config file; but I do not get the link to 'server time'. Any idea? Tia, Aldo -- _ -- 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] Weird Polycom SP 650
Hi, I am seeking help with the installation of a Soundpoint 650 desk phone. Although I have some experience (and a good one! no single issue so far, besides the problem I am trying to solve...) installing a few SP 320/330 units, I am having several issues with my first SP 650. Polycom SP 650 Data: • P/N: 3150-11530-212 • SD Sound • FW: 2.1.2.0078 • Assembly: 2345-12600-001 Rev.G The first thing I have noticed is that I was not able to upgrade the unit's firmware with the one currently available in the support area for this phone. The TFTP setup I used had worked for the upgrade of some additional SP's (SP 320/330); besides the fw files, that I got twice, even if I am not sure that this is necessary. The second strange occurrence is the inability to change the unit's display language (to Italian settings). I was however able to activate the BLF function (through a customisation found online for the sip.cfg config file), joined with the activation of the 'Presence' setting for some custom created entries of the Directory of the phone. Furthermore, once installed at my customer's site I had to fiddle with problems related to DTMF tones. The customer reported that she could not link to voicemail, to get messages. And as a matter of fact when I checked there was no way to dial the password into Asterisk, until I changed the SIP settings for this extension to 'Inband'. Needless to say that the other SP 330 have no similar issue, with 'copy cat' settings in the sip.conf file. What is however a complete disaster is what happens when the user is talking on a call, and for any reason, a second calls is presented to the unit by the Asterisk 1.6 server. The user has its headset speaker muted (and therefore thinks that the call was lost/ended abruptly), yet the party at the other end of the call is still alive and well (aka connected) and has no idea we my user starts blabbering about problems to the call. Does anybody have similar experiences with the 650? There is very little I did differently on this unit than on the other SP 330s that are running without a problem, on the same Asterisk setup.. Any additional questions are more than welcome! Kind regards Aldo -- _ -- 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] DAHDI - BRI - Astribank
Hello List, it is a very long time since I wrote here It has been still in Zaptel times Today I am run into a related problem: I can't get a DAHDI setup to work 100%. I am configuring an Astribank XR00013 (BRI, two ISDN ports). At some degree the installation (latest DAHDI drivers, Asterisk 1.6.0.18 on Centos 5.x) works. I can get incoming calls to the dialplan context that is setup as target. The call quality is good, etc. The problem is on the other side, the outgoing route. There I get just rejected calls. I am including some information on the setup: dahdi-system.conf ** # Autogenerated by /usr/sbin/dahdi_genconf on Mon Nov 30 18:37:14 2009 # If you edit this file and execute /usr/sbin/dahdi_genconf again, # your manual changes will be LOST. # Dahdi Configuration File # # This file is parsed by the Dahdi Configurator, dahdi_cfg # # Span 1: XBUS-00/XPD-00 Xorcom XPD #00/00: BRI_TE (MASTER) AMI/CCS span=1,1,0,ccs,ami # termtype: te bchan=1-2 hardhdlc=3 echocanceller=mg2,1-2 # Span 2: XBUS-00/XPD-01 Xorcom XPD #00/01: BRI_TE AMI/CCS span=2,2,0,ccs,ami # termtype: te bchan=4-5 hardhdlc=6 echocanceller=mg2,4-5 # Global data loadzone= it defaultzone = it ** dahdi-init.conf ** # # Shell settings for Dahdi initialization scripts. # This replaces the old/per-platform files (/etc/sysconfig/zaptel, # /etc/defaults/zaptel) # # The maximal timeout (seconds) to wait for udevd to finish generating # device nodes after the modules have loaded and before running dahdi_cfg. #DAHDI_DEV_TIMEOUT=40 # Override settings for xpp_fxloader #XPP_FIRMWARE_DIR=/usr/share/dahdi #XPP_HOTPLUG_DISABLED=yes ** chan_dahdi.conf ** ; ; DAHDI telephony ; ; Configuration file [trunkgroups] [channels] language=it ;context=from-pstn ;signalling=fxs_ks ;rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no ;echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both ;faxdetect=incoming ;faxdetect=outgoing faxdetect=no ;Include setup-pstn configs ;internationalprefix = 00 ;nationalprefix = 0 ; Span 1: XBUS-00/XPD-00 Xorcom XPD #00/00: BRI_TE (MASTER) ;group=0,11 group=0 context=from-pstn switchtype = euroisdn signalling = bri_cpe_ptmp channel = 1-2 ;group = 63 ; Span 2: XBUS-00/XPD-01 Xorcom XPD #00/01: BRI_TE ;group=0,12 group=1 context=from-pstn switchtype = euroisdn signalling = bri_cpe_ptmp channel = 4-5 ;group = 63 ;group=1 ** This is what I see on the CLI ** -- Executing [...@macro-outcallsrouter:39] Set(SIP/ 23001-000f, CALLERID(num)=221591030) in new stack -- Executing [...@macro-outcallsrouter:40] NoOp(SIP/ 23001-000f, Dial(DAHDI/g0/191,120)) in new stack -- Executing [...@macro-outcallsrouter:41] Dial(SIP/ 23001-000f, DAHDI/g0/191,120) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g0/191 -- DAHDI/1-1 is proceeding passing it to SIP/23001-000f -- Channel 0/1, span 1 got hangup request, cause 63 -- Hungup 'DAHDI/1-1' == Everyone is busy/congested at this time (1:0/0/1) ** And this is from the PRI debug ** -- Making new call for cr 32776 Protocol Discriminator: Q.931 (8) len=32 Call Ref: len= 1 (reference 8/0x8) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) User information layer 1: A-Law (35) [18 01 81] Channel ID (len= 3) [ Ext: 1 IntID: Implicit Other Spare: 0 Preferred Dchan: 0 ChanSel: B1 channel ] [6c 0b 21 80 32 32 31 35 39 31 30 33 30] Calling Number (len=13) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '' ] [70 04 a1 31 39 31] Called Number (len= 6) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '191' ] [a1] Sending Complete (len= 1) q931.c:3134 q931_setup: call 32776 on channel 1 enters state 1 (Call Initiated) Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 1 (reference 8/0x8) (Terminator) Message type: CALL PROCEEDING (2) [18 01 89] Channel ID (len= 3) [ Ext: 1 IntID: Implicit Other Spare: 0 Exclusive Dchan: 0 ChanSel: B1 channel ] [27 01 e8] Notification indicator (len= 3): Ext: 1 Diversion activated (DSS1)
[asterisk-users] Re: Reception Console
[EMAIL PROTECTED] is believed to have said: We are currently writing a reception console for Asterisk - if anyone is interested in beta testing it, feel free to ask. Paul Hales Does it run on *nix (Linux/MacOSX)? Is there a place we can see some information without cluttering the list? TIA Aldo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 25, Issue 119
[EMAIL PROTECTED] is believed to have said: I'd expect it to be in Falcom's best interest to support development efforts as it would open the asterisk market to them. Anyone up for creating a bounty-page for this? I would be more than interested! Anyone else? What would be the steps? Aldo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Which Mac OSX softphone with IAX2 support?
[EMAIL PROTECTED] is believed to have said: Hi, which OSX softphone do you use that supports IAX2 protocol with Asterisk? thanks Mike Hi Mike, look for LoudHush on VersionTracker... HTH Aldo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: courtesy message calling mobile phones
[EMAIL PROTECTED] is believed to have said: Can you explain this? What country? In this case it's not asterisk but the telco that has to do the Answer. To every mobile? or just that provider? Well, it's funny because here, now (Italy; Telecom Italia PSTN calling Wind mobile), I do get the courtesy message saying that they're moving me to voicemail, if I call myself from the office PBX to my mobile Wind number, and the cellphone is switched off. The question I would like to know more about is if there is some way to discriminate between a courtesy message and a carbon-based voice responder (aka 'a person' ;-). Or is there some way to match a prerecorded voice file to the audio stream coming in? This could be great to do -easily- something like the voice commands setups most cellphones offer: you say 'home', the voice file is matched, and you get the call going. Regards, Aldo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone using the GSM gateway from CyberTelecom ?
[EMAIL PROTECTED] is believed to have said: I was not sure of the quality of both these hence i wanted some real user experiences. Thanks anyways... Dan Dan, you're welcome... While I can't speak for the quality of the Dock'n'Talk, as I did never see one, the GSM gateway's audio quality is good. I must add that the place where it is operated now has a very bad record of GSM signal strength, for all operators. Still -may be due to the external antenna- it works.. Best regards, Aldo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone using the GSM gateway from CyberTelecom ?
[EMAIL PROTECTED] is believed to have said: Hi, Sorry for being very late on this thread but i am trying to make a decision on which one to go for. Options are 1. Dock n Talk offered by Voxilla (USD139) 2. GSM Gateway by CyberTelecom (GBP60) I'm having a TDM400P with 1 FXO FXS. I'm interested in implementing DISA my [EMAIL PROTECTED] Need some recommendations for experienced guys out here... Thanks... Dan Hi Dan, I am using the GSM Gateway by CyberTelecom. I guess it's a different model than the one shown on the website; it is marketed as supporting SMS messages.. It is very easy to setup. Just plug all cables in, put the GSM chip in place and you are ready to go. The reason why I asked if anybody had experience with this model is that I can't read the CallerID from the FXO TDM module; and so far I have no idea how to deal with SMS messages. I've got no manual specific to the model I have, so I do not know how to proceed... Best regards, Aldo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Anyone using the GSMgateway from CyberTelecom ?
Benchev is believed to have said: Thanks Aldo, No I do not have a manual and I don't believe such a thing exist. Actually, that GSM gateway is a Dock-N-Talk kind of thing with the exception that the handset is imbedded, so pretty much no need of a manual. Is your Grandstream a HT-488? If so you might be able to simulate the spa3000 case. Please, let me know what happened. Best regards, Benchev Hello Benchev. the unit I have has also a serial port. So while it is really easy, a matter of plugging in cables, sim, power, to set up for receiving and making calls, I have no idea how to send and receive sms messages. Or what should/can be done with the serial port. I guess there must be a use for it, or one could save the effort to put one there. I do have a GS HT-488; but while in the office I was in such a hurry that I did no test. Sorry! I'll be back there next week; I'll let you know how the test will end. Best regards, Aldo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Anyone using the GSMgateway from CyberTelecom ?
Benchev is believed to have said: Hi, Do you have any success receiving the caller id with your TDM400 FXO? I have the same problem when I connect the GSM gateway to a SPA3000 FXO line and thought this a Sipura's problem. On a phone connected to the GSM gateway I can see the callerid, but not on the Sipura's PSTN line ... Thanks, benchev Hello benchev, this is no more and no less the same problem as I do have. It appears it's then not really the TDM400 FXO module. I have another option to test: I do have a similar ATA like the Sipura, but made by Grandstream. It's here at home; I will take it to the office tomorrow and see if it can read the caller id from the GSM gateway. Even my gsm unit does indeed pass the callerID when I connect it to a cheap, dead simple analog phone! BTW: Do you have a manual for the gateway? Best regards, Aldo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Anyone using the GSMgateway from CyberTelecom ?
Hi List Is someone out there using one or more GSMgateway(s) from CyberTelecom ? Me and some friends are interested in buying some of them, but before we would like to ask, how the experiences are others have made. e.g. How easy to setup ? How reliable ? How's the voice quality ? etc. Any input/feedback is welcome. Greets Adibar Hi Adibar, I have one since a couple of weeks. It works for me. Basically you just plug it into an analog interface after installing the GSM chip. The voice quality is good even in my office; a sort of radio waves-black hole. Normally most cellphones just disappear when they are there.. The only problem I have so far is that the TDM400 FXO module does not seem to read the caller id. A regular phone shows it, if I switch connections. It might be a problem of configuration of the TDM card; I have looked in the wiki and googled around, but I do not know how I can change the way a zaptel card reads the callerid. I will try to upgrade to 1.2.x asap to see if this helps. Best regards Aldo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BT102 and ringtones
Hi list,
any success trying to let internal calls ring differently than external
calls on a Grandstream BT102?
My settings, phoneside:
Default Ring Tone:system ring tone
x custom ring tone 1, used if incoming caller ID is *
custom ring tone 2, used if incoming caller ID is #
custom ring tone 3, used if incoming caller ID is *#
This means that I checked the first custom option and entered * as
incoming caller ID
The dialplan side for this phone is:
exten = 2502,1,GotoIf($[${CALLERIDNUM:0:1} = 2]?2:5)
exten = 2502,2,SetCallerID(${CALLERIDNAME} #${CALLERIDNUM})
exten = 2502,3,Goto(10)
exten = 2502,5,SetCallerID(${CALLERIDNAME} *${CALLERIDNUM})
exten = 2502,6,Goto(10)
exten = 2502,10,Noop(${CALLERID})
exten = 2502,11,Dial(SIP/2502,30)
exten = 2502,12,Voicemail(u2502)
exten = 2502,13,Hangup()
exten = 2502,112,Voicemail(b2502)
exten = 2502,113,Hangup()
Knowing that my local (internal) extensions start with 2 the whole
thing means that if 2101 is calling 2502 the phone will see #2101 as
calleridnum .
If somebody else is calling from outiside (say 0211223344) the phone will see
*0211223344 as callerid.
This confirmed on the display of the BT, as well as in the CLI.
Now, oddly enough whatever is checked as ring tone in the websettings is
always played with no regards to the beginning character.
With the setting example above #2101 calling would obtain ringtone one,
even if there is no trace of a *-character in the calleridnum.
Is it me, is it the evil fate, is it the BT102?
I even updated the firmware to Software Version:
Program-- 1.0.6.7Bootloader-- 1.0.1.0HTML-- 1.0.0.49VOC-- 1.0.1.0
TIA,
Aldo
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[Asterisk-Users] Re: BT102 and ringtones
[EMAIL PROTECTED] is believed to have said: Is it me, is it the evil fate, is it the BT102? I even updated the firmware to Software Version: Program-- 1.0.6.7Bootloader-- 1.0.1.0HTML-- 1.0.0.49VOC-- 1.0.1.0 TIA, Aldo The attempt to obtain custom ringtones I described in my previous message bases on the tip seen in the Asterisk wiki: Pattern matching on CallerId You can instruct your phone to use a different ring sound based upon the incoming caller id: If, for example, you select the built-in standard ringtone as your default ringtone in the web interface, and enter 30 for the 1st custom ring tone and 4 for the 2nd custom ring tone then a partial leading match of the callerid will trigger the respecitve sound. Example: 302 -- 1st custom sound, 485 -- 2nd custom sound. Please note that under IP-to-IP mode (?i.e. calling directly phone-to- phone without Asterisk or another call manager?), you need to first enter the IP calling mode and then press *72 (or *90 or *92) to activate the call forwarding. Even configuring the BT102 like it is described in the wiki does (ie using the system ringtone as default; leaving the other BT web based settings like described, with the dialplan as shown) does not have any desired effect. TIA, Aldo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zap, Caller ID problem
[EMAIL PROTECTED] is believed to have said: Do you have the following set in your zapata.conf? callerid=asreceived Dear all, I add my half cent on the subject. I do have the following zapata.conf: * [channels] usecallerid = yes signalling = fxo_ks callerid = A 2302 context = internal channel = 1 signalling = fxo_ks callerid = F 2105 context = internal channel = 2 signalling = fxs_ks context = gsm_gateway callerid = asreceived channel = 4 * Unfortunately while the calls are correctly answered there is no sign of a caller id. I am using an FXO module on a TDM400 card. TIA, Aldo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM - Analog Trunk - CallerID question
Hello list.
I have a question about how to read the incoming calls' callerid on an
FXO interface of a TDM 400 analog card; (it's one of those RED modules).
Now -may this is the complexity adding step..- I have a GSM gateway
attached to this FXO thing; incoming calls are processed as they should.
But both when peeking on the CLI, as well as in the phone display I do
not see the caller id.
Here I copy the very simple zapata.conf contents:
;
; Zapata telephony interface
;
; Configuration file
;
; Zapata configuration for Asterisk server zeta-stargate
;
[channels]
; edited by aaberga % 10.02.06
;cidsignalling=v23 ; Added for UK CLI detection
;cidstart=polarity ; Added for UK CLI detection
usecallerid=yes
signalling = fxo_ks
callerid= 2302
context=internal
channel = 1
signalling = fxo_ks
callerid= 2105
context=internal
channel = 2
signalling = fxs_ks
context = gsm_gateway
callerid=asreceived
channel = 4
I made a couple of attempts activating and moving around the two UK CLI
(this unit should work in UK; I thougt those settings might help)
settings. But nothing changed, except for the fact that the calls were
no more answered. ;-)
The usecallerid = yes and the callerid=asreceived have been added and
removed, but with no success.
This is what I see on the CLI when calling the gsm unit:
-- Starting simple switch on 'Zap/4-1'
Feb 10 16:20:02 NOTICE[7409]: chan_zap.c:5405 ss_thread: Got event 17
(Polarity Reversal)...
Feb 10 16:20:06 NOTICE[7409]: chan_zap.c:5405 ss_thread: Got event 2
(Ring/Answered)...
-- Executing NoOp(Zap/4-1, GSM Gateway Call from: ) in new stack
-- Executing Dial(Zap/4-1, Zap/1|30) in new stack
This is what I have in the relevant context of the dialplan:
[gsm_gateway]
exten = s,1,NoOp(GSM Gateway Call from: ${CALLERID})
exten = s,2,Dial(Zap/1,30)
exten = s,3,Hangup()
exten = t,1,Hangup()
exten = i,1,Hangup()
Can anybody point out what I do in a wrong way?
Thanks in advance..
Aldo
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[Asterisk-Users] Re: Asterisk-Users Digest, Vol 19, Issue 72
[EMAIL PROTECTED] is believed to have said:
Hi,
Is there any syntax we can apply in the extensions to use
the anti-ex-girl(boy)friend technique to multiple callers without having
to replicate the lines?
I mean, can I write the following two lines in only one line?
exten= 12345/100,1,Hangup
exten= 12345/200,1,Hangup
One way that even makes it possible to add new girls to your list
without touching the dialplan would be the following:
exten = 12345,1,DBGet(ex-girlfriend=disposedGirlfriend/{CALLERIDNUM})
exten = 12345,2,Hangup()
; jump to here only if CALLERIDNUM was NOT found in the list
exten = 12345,102,Dial(SIP/Myself)
Regards,
Aldo
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[Asterisk-Users] Chan_BT question WAS: Asterisk with USB
[EMAIL PROTECTED] is believed to have said: Bluetooth enabled phones to talk to Bluetooth headsets; I guess there's a protocol for the phone to talk to any Bluetooth headset, no matter who made it. This protocol would have to include something to allow voice to pass from the phone to the headset and vice versa. It might also include something for dialing out. I suppose chan_bluetooth is emulating a headset, so there's support in the phone for passing voice around. There's also documentation for how to do it. I've never seen a USB headset so I doubt there's any support in the phone for passing voice over that connection. Maybe this is why there's no such thing as chan_usb Then again, I'm no expert on this matter, so maybe I'm plain wrong. Hello, let me ask something related, but on chan_bluetooth. If this driver is faking Asterisk as a headset to the BT cellphone, does this mean that it can only answer incoming calls, but NOT ask the cellphone to dial a number? I am asking because I would like to see if I can use this driver to bypass any hardware gsm-gateway to obtain a 'gsm_trunk'. Would something like Dial(bluetooth/33512345594, 20) work? Of course using the correct channel indication for the chan_bluetooth module... TIA Aldo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Euro-ISDN
[EMAIL PROTECTED] is believed to have said: [setup tool] Sorry, I cannot answer that one. I don't know enough about these cards and their drivers. Armin, thanks alot. One has to do some research and experimentation on his own every now and then; and see if there is anything interesting that might even end up in the wiki... ;-) So in the end there a lot of reasons to go for a 'better' card. Yes, a lot reasons. But actually, it depends on what you need and what you want to do. You are right.. I was asking about ISDN cards to see if there was any simple integration path for some more expensive units (where one might use those with ISDN interfaces first). But now I'll have to fiddle with the newly arrived GSM-gateway first. And this is a nice, little analog unit. Thank you very much, Aldo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Euro-ISDN
[EMAIL PROTECTED] is believed to have said: With BRIstuff you get to use ztcfg, etc. Cannot say anything about mISDN, CAPI... Francesco, thank you; this is important to know Aldo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Euro-ISDN
[EMAIL PROTECTED] is believed to have said: While we are at the subject another couple of simple related question. Are HFC-S cards active? I got one for a very low price, so that I imagine it will be NOT the case... No, these cards are passive. The protocol handling must be done by the hosts CPU / driver. OK; price sometimes tells the truth... What cards do support operation of an ISDN phone set? (I imagine there will be something similar to the FXS-FXO stuff of the analog world in the ISDN land). For that you need cards operated in NT-mode. The driver / protocol must support this, but the card too. HFC-S cards do as far as I know (I don't know if all do). Fine; I looked into the capi.conf file, but (my lack of knowledge!) I did not understand if there is any setting to tell the driver to operate a card in NT-mode. So, does the newest (which I have anyways to download for the next release of our pbx) capi driver support NT mode? When talking about active cards, I only know of Eicon cards supporting this. Well, in the end it's an option to take into consideration. One and a half years ago as I started learning about Asterisk I had an option to buy Eicon cards, but I was unsure about my ability to cope with the complexity of learning the needed bits to get started. So I bought an AVM card Today the situation changed somewhat: may be time for the next step. Armin Thanks a lot Aldo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Euro-ISDN
[EMAIL PROTECTED] is believed to have said: chan_capi does not set the NT-mode. Your cards driver need to do that. E.g. for Eicon DIVA Server cards, you just set the '-x' option with divactrl or set NT-mode in the config wizard. chan_capi does not (need) to know anything about what protocol the card is doing. CAPI is independent here. Ok. Anyway, if the card is set to NT mode, you should specify ntmode=yes in the capi.conf to tell chan_capi to handle the progress better (get progress tones). Fine! One last related subpoint: while Eicon Diva cards have their own setup application, is there anything standard to control the basic setup of generic HFC-S cards? (something similar to the ztconfig tool for analog cards) capi driver? I think you mean chan_capi? Yes it does support TE and NT mode, as well as DSS1, 1TR6, JAPAN, NI1, QSIG, 5ESS, and a lot more protocols. Yes, I meant actually the chan_capi module... Nice to know. The card and the cards driver need to support/provide this. Armin So in the end there a lot of reasons to go for a 'better' card. Thanks alot for all this information! Regards Aldo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Euro-ISDN
[EMAIL PROTECTED] is believed to have said: The active cards do the ISDN protocol stuff on board, so the host CPU/driver does not need to do that - better performance, less interrupts. The AVM cards do not have such DSPs on board, so no echo-cancel. But the Eicon DIVA Server cards do. They do analog Fax/Modem, echo-cancel, DTMF-detection, voice codec, line interconnect/mixing even to other cards, ... with the on boards DSP. chan_capi-cm is supporting most of the features and will do more soon. Armin While we are at the subject another couple of simple related question. Are HFC-S cards active? I got one for a very low price, so that I imagine it will be NOT the case... What cards do support operation of an ISDN phone set? (I imagine there will be something similar to the FXS-FXO stuff of the analog world in the ISDN land). Thanks in advance Aldo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: iDEFISK (mac iax2 softphone) release
[EMAIL PROTECTED] is believed to have said: We are considering it yes, but i don't know how hard or easy it would be. I guess we will first try to make the other versions like we want them to be and then start looking at other os'es. Zoa I have no clue, at all, too So your plan makes sense (hoping you will go for that additional challenge!).. Best regards Aldo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iDEFISK (mac iax2 softphone) release
[EMAIL PROTECTED] is believed to have said: Hey ho, A few days ago we released the linux version of the phone, today we are very happy to have the mac version ready for a little field test. Freely downloadable from http://www.asteriskguru.com/tools/idefisk_mac.php At the same time, we also put a newer version of the windows and linux versions online. Let us know how you feel about it, a more mac look (brushed metal) is coming. Please send us all problems you find, ([EMAIL PROTECTED]), all stories about how it saved or ruined your life are also appreciated. Cheers, Zoa I went looking for it and I will post my remarks ASAP. This message is, however, for a loosely related reason: are you considering a port of your softphone for the Palm platform? Strangely enough I cannot find any kind of softphone for Palm OS. Thank you and best regards Aldo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Small office with all employee's offsite
[EMAIL PROTECTED] is believed to have said: Jason, I'm sure these questions have been answered at some point, but I'm too new to this stuff to know the right words to plug into the search function to find what I need. well, yes of course. I have never touched Asterisk before, but have wanted to for some time. Now I finally think I'm going to bite the bullet, as I have a real-world application for it! You are in for some fun and satisfaction; with some small price to pay... My office consists of two employees, neither of whom work in the office physically. Here is what I'd like to do. Hopefully someone can tell me what I need to do/buy/configure/install to make it work... As a minimum set up you will need a CPU plus an interface to your incoming phone lines and most likely to an extension line in the main office. I want all calls to come into the Asterisk box in the main office. Obvious. I want all incoming calls to be recorded (not as concerned about outgoing calls) Can be done from the dialplan. Both employees have regular POTS telephone lines (one fellow has a land line and a cell, the other has just a land-line). I'd like callers to be presented with a short menu of options, the behavior of which might change depending on the time of day (for instance, at night, I'd like both the sales and support calls to go to one employee, while during the day I'd like sales to go to one person, and support to go to another. I'd also like to have an answering machine (built into Asterisk?) pick up calls that go unanswered. Can be done from the dialplan. Voicemail is an Asterisk application. I guess that's about it. I looked at the Digium TDMxx cards, but don't really know what I need in the way of FXO's and FXS's to pull off what I want to do. That's a very good option. As an added bonus, if someone knows of a VOIP adapter that allows one to plug an analog phone into it AND accept both VOIP and normal phone calls to the same phone, that would be cool (and might make things easier to configure, without making each extension 100% dependent on VOIP). You could look into products from Sipura or from Grandstream. Thanks in advance. I'm really looking forward to finally doing something with Asterisk, one of the most exciting projects I've looked at for a while!! But the very best advice I can give you is to start getting used to the Asterisk wiki and get the O'Reilly book on Asterik: it will be your friend. That's the small price to be paid. I found it worth. Regards Aldo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 13, Issue 25
[EMAIL PROTECTED] is believed to have said:
--
Message: 9
Date: Thu, 4 Aug 2005 07:43:51 +0200 (CEST)
From: Bastian Scholz [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] chan_capi upgrade
To: asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain;charset=iso-8859-1
Hi,
The old (reused) diaplan tries to dial out with:
exten = _3.,1,Dial,CAPI/mynumber:b${EXTEN}|90
The Dialstring has changed and use the same syntax as ZAP now.
From the README
- ATTENTION! the dialstring syntax now uses the zaptel dialstring syntax
It used to be: Dial(CAPI/[EMAIL PROTECTED]outgoingMSN:[b|B]destination)
Now it is: Dial(CAPI/ggroup/[b|B]destination)
Or: Dial(CAPI/contrcontroller/[b|B]destination)
Greets
Bastian Scholz
--
Bastian,
great news!
Thanks, this was what I was looking for...
Best regards,
Aldo
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[Asterisk-Users] chan_capi upgrade
Dear list,
today I installed a new asterisk machine, bound to replace my current pbx.
I am using a Fritz ISDN card; on the old machine with the drivers coming
together with the super-old rpm asterisk installation of SUSE 9.2.
The new machine is finally on asterisk 1.0.9, with chan_capi 0.5.4; now
I am doing a nightly test.
Apparently I can receive calls, but I can't dial out. I seem to remember
reading some place that there was a change in the dial app of the new
chan_capi; unfortunately I did not find (may be it's too late now for
the two neurons still awake) the message or readme where the change was
mentioned, even using heavy 'find scans' in my message database.
No luck on the wiki, either...
The old (reused) diaplan tries to dial out with:
exten = _3.,1,Dial,CAPI/mynumber:b${EXTEN}|90
; Cellular Phone numbers
exten = _3.,2,Goto(s-${DIALSTATUS},1)
exten = _3.NOANSWER,1,Hangup(); Hangup
exten = _3.BUSY,1,Hangup(); Hangup
The console says:
Aug 3 23:59:34 NOTICE[6284]: app_dial.c:764 dial_exec: Unable to
create channel of type 'CAPI'
capi info works:
mini-CDC*CLI capi info
Contr1: 2 B channels total, 2 B channels free.
Not surprising as I do get incoming calls...
Where can I find the new details on the dial with capi?
Thanks
Aldo
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[Asterisk-Users] Re: zaptel make problems
[EMAIL PROTECTED] is believed to have said: The error is the same, afaik. What I can't understand is why the make is entering in the directory '/ usr/src/linux-2.6.11.4-21.7-obj/i386/default'; I am by far not expert, but I would expect it to go fiddle with a '586' directory. Just a guess, your simlink is pointing to the incorrect linux source directory. Go into /usr/src/linux and do a ls -l, see where the linux simlink is pointing to. If it's incorrect, then do a rm linux and delete it. Recreate with a ln -s /usr/src/yourlinuxversionhere Doug Doug, thanks for the recipe! The weird thing is that a similar setup of SUSE 9.3, albeit not updated, did not show this quirk. Now, it should however be easy even for me to fix things. Thanks again Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: zaptel make problems
[EMAIL PROTECTED] is believed to have said: and watch linus himself rant about how this is incorrect to do (yet all the distros do it) :P Well, this is reassuring for a newbie like me. Even the pros (as anybody building a distro ought to be, and most of the times, really is) can do obvious errors... Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaptel make problems (long)
I know that this subject has been treated in the past! As a matter of fact reading some old messages about compiling zaptel I made a couple of tests after the first compiling failure to understand why I can't compile on a specific machine, but I do not know how to handle the results. The machine has SUSE 9.3, and an updated kernel (2.6.11.4-21.7-default; as shown below). YAST (the graphical updater/installer/ect) tells me I have an installed kernel version 2.6.11.4-21.7-i586. It tells me furthermore that the kernel sources are in sync with the compiled kernel. Now I tried to compile zaptel both with the simple 'make' as well as with the make linux26. I get errors in the two cases. Simple make: [EMAIL PROTECTED]:~/asterisk sources/zaptel-1.0.9 uname -r 2.6.11.4-21.7-default [EMAIL PROTECTED]:~/asterisk sources/zaptel-1.0.9 make clean rm -f torisatool makefw tor2fw.h rm -f zttool rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo rm -f *.ko *.mod.c .*o.cmd rm -f gendigits tones.h rm -f libtonezone* rm -f tor2ee rm -f core [EMAIL PROTECTED]:~/asterisk sources/zaptel-1.0.9 make cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA - DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA - DZAPTEL_CONFIG=\/etc/zaptel.conf\makefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA - DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA - DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA - DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o -lm -L. libtonezone.a cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA - DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA - DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -c ztspeed.c cc -o ztspeed ztspeed.o make -C /lib/modules/`uname -r`/build SUBDIRS=/home/aaberga/asterisk sources/zaptel-1.0.9 modules make[1]: Entering directory `/usr/src/linux-2.6.11.4-21.7-obj/i386/default' make -C ../../../linux-2.6.11.4-21.7 O=../linux-2.6.11.4-21.7-obj/i386/ default sources/zaptel-1.0.9 make[3]: *** No rule to make target `sources/zaptel-1.0.9'. Stop. make[2]: *** [sources/zaptel-1.0.9] Error 2 make[1]: *** [sources/zaptel-1.0.9] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.11.4-21.7-obj/i386/default' make: *** [linux26] Error 2 [EMAIL PROTECTED]:~/asterisk sources/zaptel-1.0.9 make linux26: [EMAIL PROTECTED]:~/asterisk sources/zaptel-1.0.9 uname -r 2.6.11.4-21.7-default [EMAIL PROTECTED]:~/asterisk sources/zaptel-1.0.9 make clean rm -f torisatool makefw tor2fw.h rm -f zttool rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo rm -f *.ko *.mod.c .*o.cmd rm -f gendigits tones.h rm -f libtonezone* rm -f tor2ee rm -f core [EMAIL PROTECTED]:~/asterisk sources/zaptel-1.0.9 make linux26 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA - DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA - DZAPTEL_CONFIG=\/etc/zaptel.conf\makefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA - DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA - DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA - DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o -lm -L. libtonezone.a cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA - DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA - DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -c ztspeed.c cc -o ztspeed ztspeed.o make -C /lib/modules/`uname -r`/build SUBDIRS=/home/aaberga/asterisk sources/zaptel-1.0.9 modules make[1]: Entering directory `/usr/src/linux-2.6.11.4-21.7-obj/i386/default' make -C ../../../linux-2.6.11.4-21.7 O=../linux-2.6.11.4-21.7-obj/i386/ default sources/zaptel-1.0.9 make[3]: *** No rule to make target
[Asterisk-Users] Re: Sipura 3000 Question
[EMAIL PROTECTED] is believed to have said: I don't know if it is a phone like issue or not, but try the SPA-3000 setup at http://geekgazette.com. -Kerry Kerry, thanks for the hint. A first try did not get better results, but I was doing it very quickly.. Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Sipura 3000 Question
[EMAIL PROTECTED] is believed to have said: In the advanced options there are a few options for hang-up detection including tone detection, and silence detection. They also have parameters to adjust timing and sensitivy. IIRC, they are not enabled by default. Nathan, thanks: this is something I still have to try systematically. Has anybody hints to give as where to find understandable (by the uninitiated..) documentation of what the telecoms do in Europe with regards to signaling the hangup condition? Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura 3000 Question
Dear list, I am playing with Sipura 3000 since last week. Through the wiki pages I could get running it reasonably well. My setup is that of a Sipura, linked with a local analog cordless phone, a local PSTN line and the setup to link to an asterisk server located at a remote static ip address. I can dial the cordless phone from other extensions located at the asterisk server; I can dial out from the cordless phone trough the Sipura - Asterisk link, using the PTSN line on the other port of the Sipura. So far, so good. BUT: While I can receive a phone call arriving on the PSTN port, it is correctly routed to the cordless phone on the other spa port with the faked callerid trick found in the wiki, the spa does not seems to detect the end of the call. So after the other party ends the call, I end up with an open SIP channel on the asterisk server, and what is way worse, the SPA can not accept or dial out any other call on the PSTN line. I have to manually reset it (and restart the asterisk server to get rid of the zombie SIP channel). The point is in other words how to setup the end of a call detection. I assume that the phone line I am using is set up with italian (or european / etsi) standards. How should I setup the end of call detection for this kind of pstn line? Thanks for any help, Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 9, Issue 104
[EMAIL PROTECTED] is believed to have said: http://www.petitiononline.com/mod_perl/signed.cgi?IAXPhone Sorry for the 170 or so who have already signed. This list supposedly has 10,000 or more subscribers. 170 isn't very impressive. Please sign! Just signed; more hardware side support to the IAX protocol can only be a good thing. I hope more signatures will arrive! Rgds Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: How can I make base calls with X-Lite via Asterisk?
[EMAIL PROTECTED] is believed to have said: I installed Asterisk in a default way. I ran over many manuals and FAQ's on asterisk.org. However, I found that many exaples included in them were equipment-dependent. I do not know how to configure my Asterisk for my X-Lite. Is anybody willing to help me? Regards, Abe Abe if you look on the wiki, searching the page where the Xlite settings are described, you'll find all the details you need. http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20xten%20xlite HTH Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and XLite on same machine (OSX)?
Dear all, I have tried to run an asterisk instance together with XLite on a single machine (a PowerBook). The intent is to take advantage of IAX connections to easily cross NATs while traveling. While the IAX setup proved 'easy', just having to fiddle a little with working configs at both sides, I did not succeed so far in getting XLite to connect to the local Asterisk server, AND be able to make a call. I used the pbx setup successfully with a BT102 attached to the ethernet port of my machine, after activating 'Internet Sharing' in the Sharing prefs. I am connecting trough a wireless link to the local ADSL. iax2 show registry correctly shows my pbx back in the office... So I am quite sure there is something just on the local sip setup. The best I could do was to see XLite logged in; but as soon as I dial any extension I get a 'call not approved' error message in the XLite display. The sip configuration that got me to this first is with Asterisk listening on the wireless eth interface, whereas XLite is on an second IP interface (internal ethernet board). [Even if this setup could work, I would have to change it as it needs to have an ethernet cable plugged in the eth port.] Happily I found and checked the instructions on the wiki under 'Localhost gateway'. Essentially instead of trying to have two different IP interfaces for pbx and softphone the plan is to have different port numbers (5060 and 5061). Strangely enough I can obtain again the login of the softphone, but I still get a 'call not approved' for any dialed number. Activating sip debug peer does not show anything while dialing; and no error/message shows in the Diagnostic Window of XLite. I would be happy with any hint on how to solve this. TIA Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk and XLite on same machine (OSX)?
[EMAIL PROTECTED] is believed to have said: Dear all, I have tried to run an asterisk instance together with XLite on a single machine (a PowerBook). The intent is to take advantage of IAX connections to easily cross NATs while traveling. While the IAX setup proved 'easy', just having to fiddle a little with working configs at both sides, I did not succeed so far in getting XLite to connect to the local Asterisk server, AND be able to make a call. [...] Strangely enough I can obtain again the login of the softphone, but I still get a 'call not approved' for any dialed number. Activating sip debug peer does not show anything while dialing; and no error/message shows in the Diagnostic Window of XLite. I solved the problem! It's completely unrelated to Asterisk. XLite does not support multiple accounts: I must have misinterpreted the meaning of the list of different proxies that can be found under SIP. As I already had a setting for use inside the office LAN, I wanted to leave it untouched; therefore I was adding more configurations in the next configuration entry points (Proxy 1, Proxy 2, etc). As soon as I simply edited the first configuration I got online in no time. So for any reason XLite seems either to have a bug with multiple configurations or just not support more than one different extension. (ok to me: it's a free 'lite' version; I can't really complain!) In the end I will simply get into my office extension over IAX2. Fine enough... Rgds Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: IPSwitchBoard BETA
[EMAIL PROTECTED] is believed to have said: It dosn't run under the mono framework. There, now you have an answer :-) Oh, well: sad enough... Thanks for the answer. Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: IPSwitchBoard BETA
[EMAIL PROTECTED] is believed to have said: Hi Kong, IAX2 support has been added in release 0.65 of IPSwitchBoard, which is ready for download at http://www.voip-info.org/tiki-index.php?page=IPSwitchBoard+BETA, Zap is next (I just need to get a Zap card). Thorben Hi Thorben, IPSwitchBoard looks like a very interesting complement to an Asterisk installation. Do you know how to have it running on a Linux machine? I am not familiar with .net things: would it be possible to run it on mono? TIA Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: IPSwitchBoard BETA
[EMAIL PROTECTED] is believed to have said: I am not sure that it will run on Mono, for now I only support it on Windows. (I will test it on Mono later). One thing (or so) at a time is indeed a good attitude ;-) ... As soon as you'll feel to be the time to see what happens under Mono please let us know. Thanks Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SUSE 9.2 and Zaptel channels
[EMAIL PROTECTED] is believed to have said: Suse 9.2 uses udev. Look for README.udev in you zaptel source directory and follow the instructions. Regards, Alex Thanks Alex! Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SUSE 9.2 and Zaptel channels
[EMAIL PROTECTED] is believed to have said: I have a fairly current CVS build of asterisk running on SuSE 9.2. You need to get rid of the stuff that gets installed with the system and then install the zaptel stuff. Works fine for me, but I do get warnings about unsupported modules and tainting of the kernel. The wiki has an entry on SuSE: http://www.voip-info.org/wiki-Asterisk+Linux+SuSE Thanks Tim! Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: IAX Registration being lost
[EMAIL PROTECTED] is believed to have said: I've posted this question twice without a single reply. Does that mean no one knows the answer, or no one cares to answer? I've been having an issue with an IAX2 trunk setup in Asterisk. Setup the trunk fine and it registers and works fine. I'm able to make outgoing calls from any extension and I'm happy :). Then my Internet disconnects and it won't register anymore. Simply says Request Sent forever. I can reinstate the trunk by removing it, rebooting the server (maybe a reload would also work) and then setting it up again. It then registers fine again. I'm imagining it has something to do with the fact that my home has a dynamic IP address and it's changing when the connection drops out. Of course it may also be nothing to do with it. I'd appreciate any help anyone can give. Hi there, while I can't help with the aboc problem in the scenario of dynamic IPs. I have whatI think a similar question when using static IP addresses. I have now disabled the registration to my iaxtel number from the production asterisk I am using at work, as it was regularly loosing the registration and thus trying to reregister. The * box is running the stock (aka old!) Asterisk that comes on the SUSE 9.2 cds (Asterisk CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a). My first hypothesis was that the iax2 instability could be a problem in an old release. But I was quite surprised to see the same behaviour on a different test machine (older PowerMac, running Yellow Dog Linux/Asterisk 1.0.6, compiled from sources). After starting asterisk the registration to iaxtel is ok, at first. But if I check at the CLI with iax2 show registry at random after a few minutes, most of the time asterisk is trying to re-register. I am planning to test the same process on the next machine (the mytical mini-ITX/Suse 9.2 for which I pestered the list asking for compilation and zaptel startup help in the past week). BTW: I have the PowerMac at home on a fast link (10 mbit/s) with slowly changning dynamic IPs; the pbx at work is on a slower ADSL link (128/ 640kbit/s) but with static IP address. So, I add the question: who is having no registration problems on a iaxtel link? Thanks, Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SUSE 9.2 and Zaptel channels
Of course I am not a kernel expert, so .. please be patient. I am investigating on my zaptel/zapata problem. As the main error message asterisk quits on mentions '/dev/zap/channel': No such file or directory I went peeking over there. [Asterisk Verbose Error Mar 13 20:43:35 WARNING[5779]: chan_zap.c:763 zt_open: Unable to open '/ dev/zap/channel': No such file or directory Mar 13 20:43:35 ERROR[5779]: chan_zap.c:6208 mkintf: Unable to open channel 1: No such file or directory here = 0, tmp-channel = 1, channel = 1 Mar 13 20:43:35 ERROR[5779]: chan_zap.c:9155 setup_zap: Unable to register channel '1' Mar 13 20:43:35 WARNING[5779]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 ] As a matter of fact this is what is listed in /dev/: [EMAIL PROTECTED]:~/sources/voip/zaptel-1.0.6 ls /dev [] z2ram zap1 zap2 zap3 zap4 zapchannel zapctl zappseudo zaptimer zero zkshim zqft0 I edited the list to avoid a huge message. As said I am not a Linux low level expert ... still it's striking that Asterisk does not find a pesudo file /zap/channel and there is something similar, ie zapchannel. Anyways the zaptel modules are there: [EMAIL PROTECTED]:~/sources/voip/zaptel-1.0.6 cat /proc/modules wcfxo 11808 0 - Live 0xdf2a8000 wcfxs 27680 0 - Live 0xdf2b1000 zaptel 176772 2 wcfxo,wcfxs, Live 0xdf2ba000 Does anybody have compiled the whole asterisk set 1.0.6 on SUSE 9.2? Thanks Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel problems, Asterisk 1.0.6
Hi list, I am still attempting to start an asterisk 1.0.6 fresh installation. There are some problems with the zap channel: == Parsing '/etc/asterisk/zapata.conf': Found Mar 13 10:49:38 WARNING[5278]: chan_zap.c:763 zt_open: Unable to open '/ dev/zap/channel': No such file or directory Mar 13 10:49:38 ERROR[5278]: chan_zap.c:6208 mkintf: Unable to open channel 4: No such file or directory here = 0, tmp-channel = 4, channel = 4 Mar 13 10:49:38 ERROR[5278]: chan_zap.c:9155 setup_zap: Unable to register channel '4' Mar 13 10:49:38 WARNING[5278]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' Mar 13 10:49:38 WARNING[5278]: loader.c:440 load_modules: Loading module chan_zap.so failed! [EMAIL PROTECTED]:~ Ouch ... error while writing audio data: : Broken pipe I reduced the sample zapata.conf file to: [channels] signalling = fxs_ks context = incoming channel = 4 Everything else is commented out. The zaptel.conf file is as simple as the first one; just this: loadzone = us defaultzone = us fxsks = 1-2 fxoks = 4 I have circulated the channel that I try to activate (1,2 are fxs modules in a TDM400 board; 4 is a fxo module of the same board), but nothing changes. So I have no real clue of any more steps that I am missing (the modprobe with zaptel, wcfxs and wcfxo did run silently). I even restarted the machine after modprobing to let asterisk try to get the kernel modules by itself (I do not know if this idea has any meaning). Thanks in advance Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zapping around
Dear list, I am trying to learn how to use Zap-things in Asterisk. While loading Asterisk verbosely I get this error: [chan_zap.so]Warning, flexibel rate not heavily tested! = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Mar 12 17:19:01 WARNING[5563]: chan_zap.c:763 zt_open: Unable to open '/ dev/zap/channel': No such file or directory Mar 12 17:19:01 ERROR[5563]: chan_zap.c:6208 mkintf: Unable to open channel 1: No such file or directory here = 0, tmp-channel = 1, channel = 1 Mar 12 17:19:01 ERROR[5563]: chan_zap.c:9155 setup_zap: Unable to register channel '1-2' Mar 12 17:19:01 WARNING[5563]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' Mar 12 17:19:01 WARNING[5563]: loader.c:440 load_modules: Loading module chan_zap.so failed! [EMAIL PROTECTED]:~ Ouch ... error while writing audio data: : Broken pipe Now, modprobing zaptel is ok: [EMAIL PROTECTED]:~ sudo /sbin/modprobe zaptel [EMAIL PROTECTED]:~ Same with wcfxo, wcfxs [EMAIL PROTECTED]:~ sudo /sbin/modprobe wcfxo [EMAIL PROTECTED]:~ sudo /sbin/modprobe wcfxs [EMAIL PROTECTED]:~ although here I do have a first question. The card I am using is a TDM400 with two FXS module, an empty slot and an FXO module. The wiki mentions a wctdm module that I do not find (modprobing it just fails). Am I missing something, or can I use the older set of kernel modules? [info: I did get the 1.0.6 zaptel, libpri and asterisk archives from the Digium site; I did compile everything under SUSE 9.2, thus with a stock 2.6.8-24-default kernel; I did use the make linux26 command in the install process of zaptel. ] Now the card seems to react to my fiddling: the three green leds corresponding to the installed module positions do turn on as soon as I type the wcfxo or wcfxs modprobe command. The zaptel config file is as follows: ** zaptel.conf ** the stock file as generated by the compile process, with the addition of these lines # edited by aaberga % 12.3.05 loadzone = us defaultzone = us fxsks = 1-2 #fxoks = 4 ** zaptel.conf ** The zapata.conf file is as follows: ** zapata.conf ** the stock file as generated by the compile process, with the addition of these lines ; edited by aaberga % 12.3.05 ;signalling = fxs_ks ;context = incoming ;channel = 4 signalling = fxo_ks context=internal channel = 1-2 ** zapata.conf ** I am obviously missing and/or misdoingsomething; can anybody help? Thanks in advance Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Compiling Asterisk On SUSE 9.2
Dear all, I have tried to compile * 1.0.6 (downloaded from the digium site, in the right sequence - zaptel, libpri, asterisk) on two different machines running SUSE 9.2. The problem comes during some preliminary checks: checking for ar... /usr/bin/ar checking for tgetent in -ltermcap... no checking for tgetent in -ltinfo... no checking for tgetent in -lcurses... no checking for tgetent in -lncurses... no configure: error: termcap support not found make: *** [editline/libedit.a] Error 1 Now I got the termcap rpm and afaik it's installed (now). Is there anything obvious I should try? Thanks in advance, Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Compiling Asterisk On SUSE 9.2
[EMAIL PROTECTED] is believed to have said: you need to install ncurses-dev Thanks! Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Compiling Asterisk On SUSE 9.2
[EMAIL PROTECTED] is believed to have said: It's telling you that you have no curses devel package installed. B Thanks! Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Message Waiting over a IAX trunk
[EMAIL PROTECTED] is believed to have said: I have Asterisk set up at 2 offices, connected via an IAX trunk. My delema is one person is always moving between offices. I have the dial plan set up to ring phones at both offices but his voicemail box is at office A. His phone at office A has the message indicator, however, he wants to also have the message indicator at office B. Has anyone figured out a way to set the phone registered at Office B to pick up the message waiting indicator from Office A voicemail over the trunk? If everything else fails, you send them to the second location by email or a scriplet and use another scriplet to trim already read messages in the 'other place'. But may be there some less kludgy way to do this.. Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Suse Compiling: next err
Hi all, sorry bothering again. I am still stuck in compiling asterisk. Learning (or trying to) from the first problem and first hint, when I got this error: gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o ast_expr.o dsp.o chanvars.o indications.o autoservice.o db.o privacy.o astmm.o enum.o srv.o dns.o aescrypt.o aestab.o aeskey.o utils.o editline/libedit.a db1-ast/libdb1.a stdtime/ libtime.a -ldl -lpthread -lncurses -lm -lresolv -lssl /usr/lib/gcc-lib/i586-suse-linux/3.3.4/../../../../i586-suse-linux/bin/ ld: cannot find -lssl collect2: ld returned 1 exit status make: *** [asterisk] Error 1 I thought that I was missing some ssl libraries. So I added openssl. But still I had no luck. As there are many 'ssl'-containing package descriptions I ended, after some trial and more errors, adding most development libraries. After all the disk is big enough; for some unknown (to me) reason I started getting strange compile errors, even in the zaptel and libpri compilation that was previously completing well. The funny thing is that the errors seemed to be related to the call of UI widget functions. g Now after a couple of hours of reinstallation of a fresh system I do not dare to add heaps of packages without knowledge of what is really needed. So: which ssl/crypto package does containg the right library? Thanks in advance, Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] WOW: solved (was: compiling and ssl)
OK: I rushed for help too soon! The analogy with the earlier problem (ncurses installed but not ncurses- devel) struck back; I was lacking something related to openssl... So I tried with openssl-devel and everything worked fine. Sorry bothering the list, cheers Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Suse Compiling: next err
[EMAIL PROTECTED] is believed to have said: Over here in Fedora Core land - libssl.a and libssl.so live in /usr/lib are are installed there from the RPM openssl-devel. Not sure how that translates to SuSE. Don Luckily quite closely to Fedora Core land! Thanks Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: More NAT questions -- SOLVED
[EMAIL PROTECTED] is believed to have said: Hi, all Got it to work finally. Thanks to all. Had to add [general] externip=xxx.xxx.xxx.xxx ;ip address of your nat firewall (public ip) localnet=192.168.0.0/24; the local subnet where the asterisk box is Actually, I had 'externip' before, but I have added 'localnet' one. I also had to do port forwarding on the NAT near to PHONE 2 to pass port 5060 to the phone. This is needed if you ever want to call this phone. I can e-mail my sip.conf to anyone who is interested. Rudolf Rudolf, yes, please. I fiddled with the problem but I could not solve it... TIA Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: CallTransfer
[EMAIL PROTECTED] is believed to have said: I am using BT102s and some generic voip phone. On the BT102 the transfer button will put the call on hold and give you a new line to call an extention with, however nothing happens when I call an extention. On the generic voip phone the transfer button does nothing. Well, I am using BT 102's and the transfer button works. We have to push transfer and then dial the number we want to transfer the call to. It worked on the stock phones; I updated the fw to 1.0.5.20 for other reasons (a problem with message waiting indication). HTH Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 7, Issue 296
[EMAIL PROTECTED] is believed to have said: I am using slackware 10.1 (kernel 2.4.29) and I am getting the following when I issue gcc -v Dimitris, while I never compiled chan_capi I thought you would need a 2.6 kernel to use it. HTH Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voice Message Matching?
Hi all, is there a way to sense the automated announce messages that are sent by cell phone operators? I would like to switch to my own voicemail system if I dial a coworker's cell ph. number and I am connected to the provider voicemail announce (or if the cellphone is unavailable without voicemail). It's like pattern matching, but with voice. Any way Asterisk could do this? TIA Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: capiECT problem
[EMAIL PROTECTED] is believed to have said:
On Wed, Feb 16, 2005 at 08:58:41PM +0100, Robert Rozman wrote:
Hi,
I'm trying to get capiECT working. I'd like to transfer call to
another ISDN
router connected extension and free channel from router to Asterisk.
I get incoming call on CAPI and would liek to transfer it to dialed local
extension - 400 in this case:
[outbound-capi-local]
exten = _4XX,1,NoOp(Transferring to local PBX ISDN number ${EXTEN} on msn
CAPI/${CALLERIDNUM})
exten = _4XX,2,capiHOLD
exten = _4XX,3,capiECT,${CALLERIDNUM:1}:${EXTEN}
When I dial 400, another extension rings, shows right callerid (1st
argument
to capiECT), but incoming call gets constant sound and obviously loses
connection. But capi channel is freed. When I lift handset of 400
extension,
asterisk s starts to anounce number that was sent as callerid ...
Any help, hint or working example for capiECT ?
Try to Answer the call first.
--
Tho/\/\as
I have the same problem.. I have actually tried to pass calls to non ISDN
'receivers', eg GSM cellphones.
This makes me wonder if I understand your hint; is this what you are
suggesting?
[outbound-capi-local]
exten = _4XX,1,NoOp(Transferring to local PBX ISDN number ${EXTEN} on
msn CAPI/${CALLERIDNUM})
exten = _4XX,2,Answer(); - ADDED LINE
exten = _4XX,3,capiHOLD
exten = _4XX,4,capiECT,${CALLERIDNUM:1}:${EXTEN}
Thanks,
Aldo
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[Asterisk-Users] Re: Asterisk-Users Digest, Vol 7, Issue 216
[EMAIL PROTECTED] is believed to have said: Hey Everyone, I downloaded and installed the X-Lite softphone the other day (the lite version) and cannot seem to get it to work well. Don't get me wrong, it registers with my asterisk server and everything seems to work well, except the call quality really is horrible. I thought it may be the place I was trying it at (DSL) so I took it to the office and tried it right next to the asterisk box and had the same luck. My laptop is the Dell XPS, so power, ram, etc are not problems, and loading it onto my desktop system revealed the same results. There was also no difference between a NAT implementation and a regular (live IP) implementation of the software. I am getting stuttering speech, cutouts, etc all the time. Running my Cisco 7960 at the same locations and it works fantastic with no issues at all. Is anyone else using this softphone or does anyone know of a better softphone or some hints on configuration that may make X-Lite work better..? TIA Richard, what codec are you using? I get good results on MacOS X (client side) with the ulaw codec. On my sip.conf side I do first disable all codecs; then I enable ulaw and alaw. On the x-lite side I do actually select G711u as the chosen codec. HTH Aldo PS: --- from : sip.conf disallow = all ; Disallow all codecs allow=ulaw allow=alaw allow=gsm ; GSM consumes far less bandwidth than ulaw --- end-from ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Best Grandstream firmware to use?
[EMAIL PROTECTED] is believed to have said: http://fm.grandstream.com/gs/ Diego Aguirre FWD# 459696 Thanks! Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Best Grandstream firmware to use?
[EMAIL PROTECTED] is believed to have said: Diego Aguirre wrote: I'm using 1.0.5.18 with no problems. 1.0.5.18 has an issue when registering (boot) and re-registering (after register expiration, 1 hour) that appears an 403 for a minute on the display (during this time the phone refuse calls) and then it comes back to normal operation. Didn't this happen with your phones ? Upgraded to 1.0.5.20 yesterday and I think this issue was fixed. Bye, Leonardo Leonardo, where did you get this firmware release? The Grandstream shows just 1.0.5.16 ... Thanks in advance Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Grandstream Bugetone 101 mwi
[EMAIL PROTECTED] is believed to have said:
In order to get the message button to work - programme it with the
extension number for your voice-mail. On your BT-100's phone web page -
it looks something like..
Voice Mail UserID:[300] (User ID/extension for 3rd party voice
mail system)
So if I push the 'Message' button - I effectively dial '300' (ie the
same as picking up the handset and dialing '300'). In my
extensions.conf file - the appropriate line is...
Aha! Now I understand...
I had completely misinterpreted the meaning of this field. My fault..
I thought one had to indicate the voice mailbox id (ie the id you define
in the voicemail.conf file) and NOT (as it really has to be...) the
extension number you have to dial to get your voicemails.
In other words my BT's had each a differente number in this field,
whereas one just has to put in whatever extension you dial to get to the
exten = 300,1,VoicemailMain(s${CALLERIDNUM})
exten = 300,2,Hangup
in the dialplan.
As simple as that, then...
Thanks a lot,
Aldo
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[Asterisk-Users] Re: Budgetone and MWI
[EMAIL PROTECTED] is believed to have said:
Budgetone and MWI
The message button can be programmed to dial an extension that checks
voicemail
exten = 160,1,Voicemailmain(${CALLERIDNUM})
Thanks, this is what I was thinking about. Still, how do you get the BT
to dial 160?
In my Asterisk setting I have the same mailbox numbers reused for the
voice mailboxes.
As a matter of fact when one presses the message button the BT dials the
number indicated as the voice mailbox number.
Thus in my case a small in the main extension pattern matching gives me
what I was looking for (just press the button and you get your voice mails..)
This is how it was before:
; EXT. 2XXX
; generic dialer
exten = _2XXX,1,Dial(SIP/${EXTEN},20)
exten = _2XXX,2,Voicemail(u${EXTEN})
exten = _2XXX,3,Hangup()
exten = _2XXX,102,Voicemail(b${EXTEN})
exten = _2XXX,103,Hangup()
And this is how I changed it:
exten = _2XXX,1,GotoIf(${CALLERIDNUM}=${EXTEN}?5)
exten = _2XXX,2,Dial(SIP/${EXTEN},20)
exten = _2XXX,3,Voicemail(u${EXTEN})
exten = _2XXX,4,Hangup()
exten = _2XXX,5,Voicemailmain(s${CALLERIDNUM})
exten = _2XXX,103,Voicemail(b${EXTEN})
exten = _2XXX,104,Hangup()
I reloaded the dialplan and it works! Asterisk is astonishingly flexible...
If one has different number sets for extensions and voice mailboxes there
could be still a wayout: add key/value pairs in the database and look
them up to know what mailbox to look for in 'label 5'...
Best regards,
Aldo
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[Asterisk-Users] Re: Remote Voicemail Retrieval...
[EMAIL PROTECTED] is believed to have said: Other than doing an IVR type arrangement or a phone number dedicated to VM access is there a way to do this? On my old POTS line I used to be able to call my line and simply punch * during unavailable message playback to go to the equivalent of voicemailmain(). Is there a way to do this in *? Hmm doesn't that example you mention sound an awful lot like an IVR setup? Essentially, once you answer the line, just make sure there is an extension that will send you to voicemailmain. Well, a simple thing (if you know from where you want to call in) you could detect eg the callerID of your cellphone and add a matching for '*' to send you to your voicemails... HTH Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Budgetone and MWI
Aldo Bergamini is believed to have said:
This is how it was before:
; EXT. 2XXX
; generic dialer
exten = _2XXX,1,Dial(SIP/${EXTEN},20)
exten = _2XXX,2,Voicemail(u${EXTEN})
exten = _2XXX,3,Hangup()
exten = _2XXX,102,Voicemail(b${EXTEN})
exten = _2XXX,103,Hangup()
And this is how I changed it:
exten = _2XXX,1,GotoIf(${CALLERIDNUM}=${EXTEN}?5)
exten = _2XXX,2,Dial(SIP/${EXTEN},20)
exten = _2XXX,3,Voicemail(u${EXTEN})
exten = _2XXX,4,Hangup()
exten = _2XXX,5,Voicemailmain(s${CALLERIDNUM})
exten = _2XXX,103,Voicemail(b${EXTEN})
exten = _2XXX,104,Hangup()
I reloaded the dialplan and it works! Asterisk is astonishingly flexible...
Well, sort of...
Pressing the message button sends me to voicemailmain. However, I go to
voicemailmain even if from ext. 2102 I dial 2101...
This is clearly a problem.
To understand what is going on I changed the 'dialer' fragment as follows:
; EXT. 2XXX
; generic dialer
exten = _2XXX,1,Answer
exten = _2XXX,2,NoOp(${CALLERID})
exten = _2XXX,3,GotoIf(${CALLERID}=${EXTEN}?7)
exten = _2XXX,4,Dial(SIP/${EXTEN},20)
exten = _2XXX,5,Voicemail(u${EXTEN})
exten = _2XXX,6,Hangup()
exten = _2XXX,7,Voicemailmain(s${CALLERIDNUM})
exten = _2XXX,104,Voicemail(b${EXTEN})
exten = _2XXX,105,Hangup()
Now I have noticed two things I do not understand:
a) I do not see the caller id printed on the console; why?
b) as it is now the GotoIf never matches; even if I am pressing the
message id button on one of the Budgetones; or if I dial my own extension..
I am running this version of Asterisk:
Asterisk CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a built by
[EMAIL PROTECTED] on a i686 running Linux
Thanks for any hint...
Aldo
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[Asterisk-Users] Re: Budgetone and MWI
[EMAIL PROTECTED] is believed to have said: I don't mean to be rude to everyone who responded to this question, but I think that everyone is answering the wrong question. The point is that the message waiting indicator doesn't light up, at all, ever. All that happens when messages are waiting is that the display blinks and the phone gives a stutter dialtone. That's it. There is no light under the button - there should be, but there isn't. The blinking phone designers should have put those stupid blinking red leds - that only flash on boot up - under the message button and flashed the display during boot up. But they didn't and we're stuck with it. Such is life. Stephen R. Besch I noticed this strange factoid as soon as I got MWI to work. Does anybody know what then is the use for the message button? Thanks, Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Grandstream Bugetone 101 mw
[EMAIL PROTECTED] is believed to have said: Hahawell the MWI is the blinking blue LCD. The message button is reserved for future use Hang in there. There will soon to be some No the message button call the number you configure in the web interface. Presumably voicemail, but could be your mistress. Then it's easy to get voicemail checking to work... Thanks! Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Budgetone 10x mwi
[EMAIL PROTECTED] is believed to have said:
Ronald, it's the context listed in voicemail.conf (I got caught on this
as well)
I really wish Asterisk was better documented; it's bullshit the way it
stands at the moment.
Cheers,
Dean
Dean,
so if I have two contexts defined in voicemail.conf, like:
[general]
[local]
{voicemailboxes]
I do have to set [EMAIL PROTECTED] in the Budgetone webconfiguration page? I
guess BT 102 does not make a difference to 101...
Thanks,
Aldo
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[Asterisk-Users] Asterisk and echo
Since a couple of days I using an Asterisk server. I noticed something obvious to anybody dealing with telephony since any longer time than myself; echo is nasty... Is it correct to say that the difference between a conversation between SIP phone, Asterisk, an ISDN BRI line and a GSM phone is entirely digital, whereas I do have a remote analog portion of the sound link only if I dial an analog landline subscriber? At times I do hear a bad echo of my own voice when calling old POTS users (while they do not report any echo), while so far I did not have any echo with cellphones. Can I do anything with echo suppression on my side? Is it beyond the features of my cheap (convenient..) Fritz PCI card? Thanks, Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Inbound calls (similar problem; ISDN - chan_capi)
[EMAIL PROTECTED] is believed to have said: Hey Dan!! Give us a clue as to what hardware/setup network provider you have there, and we might be able to help :) Paul Hello Paul, hello everybody! I have, too, an inbound call problem. I am using an ISDN Fritz Card PCI 2.00, together with chan_capi 3.5.x . As I call my number I get (finally) a reaction from the pbx. But still no complete call passing. Activating capi debug this is what I see on the CLI: Jan 6 15:49:55 WARNING[1112791984]: pbx.c:1868 ast_pbx_run: Channel 'CAPI[contr1/221591030]/0' sent into invalid extension 's' in context 'default', but no invalid handler -- DISCONNECT_IND ID=002 #0x0a68 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x3400 Now I do have a capi_conf file that in my opinion should send inbound calls to a different context: [interfaces] msn=0221591030 incomingmsn=221591030 controller=1 devices=2 softdtmf=1 callgroup=1 context=from-chan_capi Can anybody help me understanding if the incoming call gets answered and if so why it is sent to a default context? What is an invalid handler? Thanks in adavance Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 6, Issue 76
[EMAIL PROTECTED] is believed to have said: Change the order of the lines and reload (or better restart Asterisk): [interfaces] msn=0221591030 incomingmsn=221591030 softdtmf=1 callgroup=1 context=from-chan_capi devices=2 controller=1 Cheers, Philipp Philipp, THANKS! I got past this stage by chance, as I did follow without knowing it your advice: at some point I restarted Asterisk. As you suggested it worked! The next funny thing is that while Asterisk started getting inbound calls, but it had trouble going to the context specified in the capi.conf file. It was (and is) insisting about a default context, that I did not have. In the end I did create such a default context and inbound calls got passed there. Curious behaviour... I will try to get past this quirk with the different ordering of lines. It's anyways a fine satisfaction seeing the own (first) digital pbx running! Thanks again, Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Inbound calls (similar problem; ISDN - chan_capi)
Sorry for the post with the bad subject - AAB [EMAIL PROTECTED] is believed to have said: Change the order of the lines and reload (or better restart Asterisk): [interfaces] msn=0221591030 incomingmsn=221591030 softdtmf=1 callgroup=1 context=from-chan_capi devices=2 controller=1 Cheers, Philipp Philipp, THANKS! I got past this stage by chance, as I did follow without knowing it your advice: at some point I restarted Asterisk. As you suggested it worked! The next funny thing is that while Asterisk started getting inbound calls, but it had trouble going to the context specified in the capi.conf file. It was (and is) insisting about a default context, that I did not have. In the end I did create such a default context and inbound calls got passed there. Curious behaviour... I will try to get past this quirk with the different ordering of lines. It's anyways a fine satisfaction seeing the own (first) digital pbx running! Thanks again, Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CAPI Question
Dear list, I am starting to setup an asterisk pbx, using a Fritz ISDN card through chan_capi (0.3.5). The underlying OS is SUSE 9.2; I installed asterisk with the RPMs supplied on the DVD. While I can dial out (I had successful outside calls), through the ISDN card, so far I could not answer a phone call on the card. My capi.conf file is quite fantasyless: ; ; CAPI config ; ; [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] msn=221591030 incomingmsn=221591030 controller=1 devices=2 softdtmf=1 callgroup=1 context=from-chan_capi ;accountcode= ;echosquelch=1 ;echocancel=yes ;echotail=64 ;deflect=02-fastweb !! Now if I try from one SIP extension to call my self (on 0221591030) , I can obtain the 'ringing' of the office number. But I was not able to see Asterisk answering the call. There is a second ISDN physical phone connected, as well as a Zyxel ISDN router (with two analog phones attached). Everything rings but the internal SIP extension... After some fiddling I did activate capi debugging; here is what prints out on the CLI during an attempt: gamma-stargate*CLI capi info Contr1: 2 B channels total, 2 B channels free. gamma-stargate*CLI reload Jan 5 16:52:40 NOTICE[1110690736]: indications.c:397 ast_unregister_indication_country: Removed default indication country 'it' -- CONNECT_CONF ID=002 #0x0041 LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 -- INFO_IND ID=002 #0x17cc LEN=0015 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x800d InfoElement = default -- INFO_IND ID=002 #0x17cd LEN=0016 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x18 InfoElement = 89 -- CONNECT_IND ID=002 #0x17ce LEN=0049 Controller/PLCI/NCCI= 0x201 CIPValue= 0x10 CalledPartyNumber = a1221591030 CallingPartyNumber = 21 81221591030 CalledPartySubaddress = default CallingPartySubaddress = default BC = 80 90 a3 LLC = default HLC = 91 81 AdditionalInfo = default Jan 5 16:52:57 NOTICE[1088080816]: chan_capi.c:1932 capi_handle_msg: CONNECT_IND ID=002 #0x17ce LEN=0049 Controller/PLCI/NCCI= 0x201 CIPValue= 0x10 CalledPartyNumber = a1221591030 CallingPartyNumber = 21 81221591030 CalledPartySubaddress = default CallingPartySubaddress = default BC = 80 90 a3 LLC = default HLC = 91 81 AdditionalInfo = default Jan 5 16:52:57 ERROR[1088080816]: chan_capi.c:2051 capi_handle_msg: did not find device for msn = 221591030 -- INFO_IND ID=002 #0x17cf LEN=0025 Controller/PLCI/NCCI= 0x201 InfoNumber = 0x70 InfoElement = a1221591030 Jan 5 16:52:57 ERROR[1088080816]: chan_capi.c:1198 find_pipe: unable to find a pipe for PLCI = 0x201 MN = 0x17cf Jan 5 16:52:57 NOTICE[1088080816]: chan_capi.c:1302 pipe_msg: INFO_IND ID=002 #0x17cf LEN=0025 Controller/PLCI/NCCI= 0x201 InfoNumber = 0x70 InfoElement = a1221591030 -- INFO_IND ID=002 #0x17d0 LEN=0016 Controller/PLCI/NCCI= 0x201 InfoNumber = 0x18 InfoElement = 8a Jan 5 16:52:57 ERROR[1088080816]: chan_capi.c:1198 find_pipe: unable to find a pipe for PLCI = 0x201 MN = 0x17d0 Jan 5 16:52:57 NOTICE[1088080816]: chan_capi.c:1302 pipe_msg: INFO_IND ID=002 #0x17d0 LEN=0016 Controller/PLCI/NCCI= 0x201 InfoNumber = 0x18 InfoElement = 8a -- DISCONNECT_IND ID=002 #0x17d3 LEN=0014 Controller/PLCI/NCCI= 0x201 Reason = 0x0 Jan 5 16:52:57 ERROR[1088080816]: chan_capi.c:1198 find_pipe: unable to find a pipe for PLCI = 0x201 MN = 0x17d3 -- INFO_IND ID=002 #0x17d4 LEN=0015 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x8001 InfoElement = default -- DISCONNECT_CONF ID=002 #0x0042 LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 -- DISCONNECT_IND ID=002 #0x17d5 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x3400 gamma-stargate*CLI Unfortunately I do not understand where the problem is. The context inside extension.conf that I would like to obtain the call is the following: ; *** INBOUND CONTEXT: DIAL INTERNAL PHONES [from-chan_capi] ; reach the internal dialplan context! include = incoming include = nba_plan ; *** INCOMING CONTEXT: FROM FRITZ! CARD [incoming] ;
[Asterisk-Users] Re: ISDN/SS7 book?
[EMAIL PROTECTED] is believed to have said: some time ago, I asked the list of a good book for learning ISDN and SS7. I don't need to know how to write a channel driver or something; I just want to know more about the possibilities and what's really sent back and forth. I was told the book ISDN and SS7: Architectures for Digital Signaling Networks by Uyless Black (ISBN 0132591936) was a good choice, but this seems sold out. Does anyone know about another book about the subject? thanks roy Roy, if you look up on Amazon you'll find it used. HTH Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk and Capi
Bruno Hertz is believed to have said: Hi Aldo don't know about Suse, but I have a working setup with asterisk 1-0 stable, chan_capi 0.3.5 and fcpci-suse9.1-3.11-02 on Debian Sarge, though not prepackaged but all self compiled. Looking at your log messages, chan_capi obviously is installed, but the load of app_capiCD.so fails due to an undefined symbol capidebug. [...] So either your modules.conf is messed up, or there's a problem with the chan_capi package itself, which you should then report to Suse. But take a look at your modules.conf. I myself have autoload enabled, and all works automagically. Maybe you have it disabled, and the module load order is affected by this ? Also, you can check if you like wether the symbol is actually defined in chan_capi.so: # nm chan_capi.so | grep capidebug 00010720 B capidebug If you see a 'U' instead of a 'B' there, your chan_capi package is messy. Regards, Bruno. Bruno, thanks for your hints! They were precious as I could easily trace the state of my box. The drivers were OK, but I was loading other ISDN drivers on the one hand and missing the config file for chan_capi. Quite a messy situation. So I looked into the wiki and the example file from the chan_capi driver and came up with a neutral config file. Now my box happily reports to see the Fritz! card, with the two B channels sitting idle (I am away from the office and just did send the updated config files to the unconnected box to see if they were the right approach..). So thanks again for the hints (and have a fine new year) Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk and Capi
Dear list, I tried to install the CAPI support RPM from the SUSE 9.2 disks. YaST tells me it is happy with the process. The Asterisk release I am using is the one that comes packaged in RPM format, also included in the distribution. Still starting asterisk with the usual asterisk -vvvc I see that something goes wrong. [app_capiCD.so]Dec 23 19:21:45 WARNING[1076850816]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/app_capiCD.so: undefined symbol: capidebug Dec 23 19:21:45 WARNING[1076850816]: loader.c:423 load_modules: Loading module app_capiCD.so failed! A dumb (newbie) question how to solve the problem: is there some other package that I have to install to be able to start asterisk with the chan_capi? Best regards and seasonal greetings, Aldo SW versions: SUSE 9.2 Asterisk - 1.0.0.2004.0814.-1.i586.rpm Asterisk - capi.0.3.5-3.i586.rpm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Getting started with Asterisk
[EMAIL PROTECTED] is believed to have said: Hello , I'll just started with asterisk and I would liket to to dial between your two phones with to cisco ATA 186 , but I have a problem The two cisco ATA and the server in the same networks and i have the ring in the phone but i'am not able to place a call Between the twe phone . In attachement the sip.conf and a log file Any suggestement . Regards RAbii Rabii, I can't help as this is the same problem I have with the very same sample configurations. Here I am using the OSX install of Asterisk (with an older release). What I saw poking around is that on the server machine at port 5060, if I try peeking in with telnet, nobody is listening. So either the two configs (sip.conf and extensions.conf) are not enough or some configuration info in some other file is messing up. What is the minimum set of configuration files needed to operate a SIP only Asterisk setup? TIA Aldo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Getting started with Asterisk
[EMAIL PROTECTED] is believed to have said: Telnet uses TCP, SIP listens on UDP, use netstat instead. B Bob, thanks for the hint! I should have imagined that SIP could not use a tcp protocol... Aldo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] which ISDN Card?
[EMAIL PROTECTED] is believed to have said: Hello, I am a newbie with asterisk; I¥ve searching the mailinglist, www.voip-info.org, isdn4linux web... But I don¥t know which isdn card to buy. I want the * box to be able to send faxes, and obviusly to send and receive calls. 1) What do you recomend me? 2) Would AVM ISDN Fritz Card PCI V2.0 work? and Eicon Diva ISDN Modem PCI? 3) Do you know any cheap site to buy? Many thanks. Rubens, where are you located? I have just ordered from an italian reseller the AVM Fritz Card, for about 70 Euros. As I could not locate any reseller info I just wrote to AVM (from their website, http://www.avm.de/en/index.php3) and got a couple of references to local dealers very quickly. AFAIK the Fritz Card is a popular and cheap ISDN solution (at least for Europe). I will see in a couple of days how the jump from theory to reality really looks like. HTH Aldo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
