[Asterisk-Users] IAX2 Rejected connection attempt (voiptalk.org)
Hi all, Have a problem that I have been battling with for a few days now with help from voiptalk.org support.but I thought someone here might have seen this before. I have an asterisk box running on a real non nat'ed ip address with an incoming number from voiptalk.org on IAX2. The problem I am seeing with or without firewall rules in place (port 4569 udp open or all ports open ie firewall rules flushed) is rejected connection attempt from xxx.xxx.xxx.xxx which is voiptalks IAX server... The real killer here is that this was all working but my inter asterisk sip connections were not transmitting voice, after changing canreinvite=no to canreinvite=yes in sip.conf the sip connections started sending/receiving voice ok but the incoming calls were being rejected. I can also dial out on voiptalks IAX connection happily. And just to be sure I changed the canreinvite back the problem remains. I have all but rebooted the box. I have restarted asterisk and checked everything again, but I just can't see why this is happening. Cheers, Mark ___ 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
Re: [Asterisk-Users] Re: IAX2 Rejected connection attempt (voiptalk.org)
Tony Mountifield wrote: Mark Benson <[EMAIL PROTECTED]> wrote: Hi all, Have a problem that I have been battling with for a few days now with help from voiptalk.org support.but I thought someone here might have seen this before. I have an asterisk box running on a real non nat'ed ip address with an incoming number from voiptalk.org on IAX2. The problem I am seeing with or without firewall rules in place (port 4569 udp open or all ports open ie firewall rules flushed) is rejected connection attempt from xxx.xxx.xxx.xxx which is voiptalks IAX server... Where is this message coming from? Asterisk? kernel IPtables? Also, where is it appearing? e.g. /var/log/messages, console, or Asterisk log file. What does your iax.conf look like? Cheers Tony The message is on the asterisk console - this is all I see even if iax2 debug is on and verbose is 30+ iax.conf looks like this (more or less - comments removed) [general] bindport=4569 allow=all ; same as bandwidth=high disallow=lpc10 jitterbuffer=no [voiptalk] type=peer username= secret=xx context=default host=iax.voiptalk.org [08700nn] type=peer username=08700nn context=default host=iax.voiptalk.org Last two items as per voiptalks' instructions (user and pass and 0870 no removed for list) ___ 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
Re: [Asterisk-Users] Reccomendation for reliable handsets
I have been using an IN1002 generic handset (supposed to be an unbranded cisco copy but I am skeptical) for a few months (6months+) now, and it seems pretty stable - however I haven't found a reliable supplier Also there is almost no support for them.. I have switched to the grandstream budgetone 102 and they seem pretty good too. You can pretty much plug in and forget it with both phones. They do lock up occasionally (once a month to once every 3 months). I have yet to upgrade the firmware on the grandstreams... Mark Brett, Gary wrote: Sorry to move this up the list again, but does anybody have any advice on this -Original Message- From: Brett, Gary [mailto:[EMAIL PROTECTED] Sent: 02 February 2005 10:49 To: 'asterisk-users@lists.digium.com' Subject: [Asterisk-Users] Reccomendation for reliable handsets Hi there I'm sure this question has been raised a number of times before, but unfortunately I do not have direct access to the archives I am about to roll out Asterisk to a few companies and would like to hear your experiences about the various handsets/phones that are Asterisk compatible I am primarily looking for 2 options, the first being a cheaper model which will provide reliability whilst still maintaining a reasonable feature set, and a reliable model from the more expensive range with more features But the definite focus here is on reliability and ease of maintenance Any help or advice would be greatly appreciated; I would really like to hear your experiences/recommendations Cheers Gary ___ 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 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 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
Re: [Asterisk-Users] Re: IAX2 Rejected connection attempt (voiptalk.org)
Cheers Tony, That sorted it! - have passed this info onto voiptalk to update their help pages You wouldn't care to add an explanation as to why user works over friend? Cheers, Mark Tony Mountifield wrote: Mark Benson <[EMAIL PROTECTED]> wrote: Tony Mountifield wrote: Mark Benson <[EMAIL PROTECTED]> wrote: Have a problem that I have been battling with for a few days now with help from voiptalk.org support.but I thought someone here might have seen this before. I have an asterisk box running on a real non nat'ed ip address with an incoming number from voiptalk.org on IAX2. The problem I am seeing with or without firewall rules in place (port 4569 udp open or all ports open ie firewall rules flushed) is rejected connection attempt from xxx.xxx.xxx.xxx which is voiptalks IAX server... Where is this message coming from? Asterisk? kernel IPtables? Also, where is it appearing? e.g. /var/log/messages, console, or Asterisk log file. What does your iax.conf look like? The message is on the asterisk console - this is all I see even if iax2 debug is on and verbose is 30+ OK, that says that it is Asterisk that is rejecting the connection, not any firewall. iax.conf looks like this (more or less - comments removed) [general] bindport=4569 allow=all ; same as bandwidth=high disallow=lpc10 jitterbuffer=no [voiptalk] type=peer username= secret=xx context=default host=iax.voiptalk.org [08700nn] type=peer username=08700nn context=default host=iax.voiptalk.org Last two items as per voiptalks' instructions (user and pass and 0870 no removed for list) The [08700nn] section should be type=user, not type=peer. That is almost certainly the cause of your problem. Also, some items are not required, and will be ignored (e.g. [voiptalk] doesn't need context, and [08700nn] doesn't need username or host). Here is my working setup: [08700nn] type=user notransfer=yes context=voiptalk-incoming [voiptalk] type=peer username= secret=xx host=iax.voiptalk.org notransfer=yes ;trunk=yes qualify=yes Hope this helps! Cheers Tony ___ 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] How to xfer calls or is my setup wrong?
I am having problems transferring calls from one sip extension to another - the extensions use various phones hardware/software. From what I can tell I should just be able to press # and then dial an extension to blind xfer a call right? How do I do attended xfer? Either the phones (for this test I have tried xlite and budgetone102) are not sending DTMF correctly or something else is amiss... The call comes in from an external number via IAX2 (0870xxx) which I can answer on any of the ringing extensions no problem. But when I need to xfer that call I am more or less stuck. I have read various posts and something about *8# ? seemed to partially work one on the grandstream but I haven't been able to reproduce that... The CLI doesn't show anything odd... Any ideas? ___ 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
Re: [Asterisk-Users] How to xfer calls or is my setup wrong?
I have this in my extensions.conf exten => 08700xx,1,Dial(SIP/test1&SIP/test2&SIP/test3,30,t) To ring a group of internal extensions for any call coming in on that number And exten => 100,1,Dial(SIP/test1,20,Trt) exten => 100,2,Voicemail(u100) exten => 100,3,Hangup() exten => 100,102,Voicemail(b100) exten => 100,103,Hangup() For each extension... Altus Snyman wrote: What asterisk version I know we had a problem with one of the cvs We couldn't use the transfer buttons,but # worked What about the Dail(SIP/111,12,tT) in your extensions.conf On Tue, 2005-02-08 at 13:50, Mark Benson wrote: I am having problems transferring calls from one sip extension to another - the extensions use various phones hardware/software. From what I can tell I should just be able to press # and then dial an extension to blind xfer a call right? How do I do attended xfer? Either the phones (for this test I have tried xlite and budgetone102) are not sending DTMF correctly or something else is amiss... The call comes in from an external number via IAX2 (0870xxx) which I can answer on any of the ringing extensions no problem. But when I need to xfer that call I am more or less stuck. I have read various posts and something about *8# ? seemed to partially work one on the grandstream but I haven't been able to reproduce that... The CLI doesn't show anything odd... Any ideas? ___ 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 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 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
Re: [Asterisk-Users] How to xfer calls or is my setup wrong?
I put dtmfmode=rfc2388 into the sip.conf definitions for each sip client and now asterisk is recognising the # key press - guess it wasn't hearing the dtmf tones... Now blind xfer works - how do I do attended xfer? I have read posts about it being in the CVS version - I am running the 1.0.3 release... Altus Snyman wrote: What asterisk version I know we had a problem with one of the cvs We couldn't use the transfer buttons,but # worked What about the Dail(SIP/111,12,tT) in your extensions.conf On Tue, 2005-02-08 at 13:50, Mark Benson wrote: I am having problems transferring calls from one sip extension to another - the extensions use various phones hardware/software. From what I can tell I should just be able to press # and then dial an extension to blind xfer a call right? How do I do attended xfer? Either the phones (for this test I have tried xlite and budgetone102) are not sending DTMF correctly or something else is amiss... The call comes in from an external number via IAX2 (0870xxx) which I can answer on any of the ringing extensions no problem. But when I need to xfer that call I am more or less stuck. I have read various posts and something about *8# ? seemed to partially work one on the grandstream but I haven't been able to reproduce that... The CLI doesn't show anything odd... Any ideas? ___ 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 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 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] Music on hold is a durge
I have just setup music on hold by downloading and installing mpg123 r Now I have music on hold but it sounds terrible - clipping, buzzing, digital distortion, and its too loud (which probably isn't helping) and I'm just running it thru the 'default' line in music onhold.conf line default => quietmp3:/var/lib/asterisk/mohmp3, with the default mp3s. This is a standard 1.0.3 box, running headless (no x desktop) on FC2. On a P4 2.4GHz Any ideas? ___ 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
Re: [Asterisk-Users] newbie questions
Voip to voip (in whatever form that takes sip-sip sip-iax iax-iax) is what I am using asterisk for. I would have thought mandrake would have been ok - but haven't used it for a while. I'm running FC2 (fedora core2) and asterisk complies and runs without any problems. Dont fear make. Apps, for the most part, compile really easily on linux. Follow the instructions to the letter and you shouldn't go wrong. Its often as simple as typing make waiting a bit for stuff to stop happening and then typing make install. Asterisk prompts you with various other options like make help and make samples (or something like that) so its pretty straight forward. You don't need any cards for asterisk. No phone cards, no sound card just whatever allows you to connect to your lan and/or the internet. g729 - isn't required. There are plenty of other codecs you can use for free. Accounting, cdr, ser etc - I haven't got that far myself either. Shaoul Jacobson - TELLINK wrote: Hi, I want to use asterisk as a VoIP-VoIP pbx (just IP, no PSTN/ISDN cards) 1. the distro I downloaded a "free mandrake 10.0 - 3 CD's) but some packages seem missing (some C or C++ or python ...) (buy the full version ) maybe the latest fedora is more complete ? or easier to complete with rpmfind (I am green to linux too, but I open my windows & gates to the tux) (bsd, debian are a bit too tech for me yet, no flaming please.) I prefer ready made rpm's or alike than compile AT THIS TIME. (I promise to improve over time) 2. download any rpm ? or I must download sources and 'make install' ? (I found one iso, but it seemed to require a pstn card) (RTFM a second / third time could is always a good option) 3. pure VoIP is it ok to use it in pure VoIP mode without any 'phone cards' ? all (most) settings & samples I see include such cards. Needed or not ? 4. g729 not free. It seems that requires some licensing to digium. Can that be without using any 'card' (just VoIP) ? How to control the licenses then ? (I e-mailed them the question, but got no answer) accounting, cdr's, ... that's for later (first I have to be able to phone) regards, Shaoul Jacobson ___ 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 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] Music on hold distorted
Yesterday I setup music on hold by downloading and installing mpg123 r Now I have music on hold but it sounds terrible - clipping, buzzing, digital distortion, and its too loud (which probably isn't helping) and I'm just running it thru the 'default' line in music onhold.conf line default => quietmp3:/var/lib/asterisk/mohmp3, with the default mp3s. This is a 1.0.3 box, running headless (no x desktop) on FC2. On a P4 2.4GHz I have listened to the music on hold from both xlite and a budgetone 102 and it sounds the same from both. Any ideas? ___ 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
Re: [Asterisk-Users] Music on hold distorted
I installed mpg123.0.59s and that was nasty so installed 0.59r but it was still distorted, eventually deleted s and reinstalled r and after a few mins the music on hold sorted itself out - it just happend as I was testing it after reinstalling - weird - I had looked at the phone/asterisk settings but found nothing odd - anyhoo all sorted now. Cheers, Mark Scott Herrick wrote: Mark, I have heard this problem. I'm not exactly sure what the cause is but check for any duplex mismatches between the phone and the * box. Hope this helps. Scott H Mark Benson wrote: Yesterday I setup music on hold by downloading and installing mpg123 r Now I have music on hold but it sounds terrible - clipping, buzzing, digital distortion, and its too loud (which probably isn't helping) and I'm just running it thru the 'default' line in music onhold.conf line default => quietmp3:/var/lib/asterisk/mohmp3, with the default mp3s. This is a 1.0.3 box, running headless (no x desktop) on FC2. On a P4 2.4GHz I have listened to the music on hold from both xlite and a budgetone 102 and it sounds the same from both. Any ideas? ___ 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 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 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] App_conference in dial plan?
Hi all, I've been trying to get meetme working for a while now (complie problems - will probably try again later on another machine) but have given up and started looking at alternatives. I've managed to get app_conference compiled and installed - show modules shows its there in asterisk, but I don't know how too actually use it in the dial plan... The info on voip-info doesn't explain its usage very well... The dial plan example doesn't (to my mind anyway) specify an extention to call for conferencing... ; Make as many of these contexts as you have seperate conference bridges ; change "conferencename" in each [conf-conferencename] exten => join,1,System(/opt/asterisk/bin/conference-announce conferencename in) exten => join,2,Conference(conferencename/S/1) exten => h,1,System(/opt/asterisk/bin/conference-announce conferencename out) [confhelper] ; make one of these extensions per seperate conference bridge exten => conf-conferencename,1,Conference(conferencename/S/1) exten => in,1,Answer() ; if I use Playback here instead of BackGround, asterisk crashes exten => in,2,BackGround(conf-announce) exten => in,3,ResponseTimeout(5) exten => in,4,Hangup() exten => out,1,Answer() exten => out,2,BackGround(conf-leave) exten => out,3,ResponseTimeout(5) exten => out,4,Hangup() how do I setup up app_conference to respond to an extention? Just a real simple example to get me started would be appreciated... I've tried a few things along the lines of the example meetme extention ie exten => 901,1,app_conference(901||1234) or exten => 901,1,cmd_conference(901||1234) But I guess its expecting too much to think that this would fireup app_conference Thanks in advance for any help. Cheers, Mark ___ 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
Re: [Asterisk-Users] App_conference in dial plan?
exten => 901,1,Conference(Internal Test Conference/S/1) Looks like it does the job... Mark Benson wrote: Hi all, I've been trying to get meetme working for a while now (complie problems - will probably try again later on another machine) but have given up and started looking at alternatives. I've managed to get app_conference compiled and installed - show modules shows its there in asterisk, but I don't know how too actually use it in the dial plan... The info on voip-info doesn't explain its usage very well... The dial plan example doesn't (to my mind anyway) specify an extention to call for conferencing... ; Make as many of these contexts as you have seperate conference bridges ; change "conferencename" in each [conf-conferencename] exten => join,1,System(/opt/asterisk/bin/conference-announce conferencename in) exten => join,2,Conference(conferencename/S/1) exten => h,1,System(/opt/asterisk/bin/conference-announce conferencename out) [confhelper] ; make one of these extensions per seperate conference bridge exten => conf-conferencename,1,Conference(conferencename/S/1) exten => in,1,Answer() ; if I use Playback here instead of BackGround, asterisk crashes exten => in,2,BackGround(conf-announce) exten => in,3,ResponseTimeout(5) exten => in,4,Hangup() exten => out,1,Answer() exten => out,2,BackGround(conf-leave) exten => out,3,ResponseTimeout(5) exten => out,4,Hangup() how do I setup up app_conference to respond to an extention? Just a real simple example to get me started would be appreciated... I've tried a few things along the lines of the example meetme extention ie exten => 901,1,app_conference(901||1234) or exten => 901,1,cmd_conference(901||1234) But I guess its expecting too much to think that this would fireup app_conference Thanks in advance for any help. Cheers, Mark ___ 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 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
Re: [Asterisk-Users] UK SIP Provider
voiptalk.org seem to be pretty reliable for both incoming and outgoing calls... I've been using them for at least 6 months for light volume calls. Steve Foy wrote: Hi, I'm looking for a reliable provider to use mainly for outgoing calls in the UK, incoming isn't so much of a worry as I think I'm going to accept them over ISDN. Cheers! Steve ___ 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] Call forward...
Hi,
I'm trying to setup a call forwarding rule so that when an extention
doesn't answer the call is forwarded to my mobile.
I'm using voiptalk.org for incoming and outgoing calls and SIP phones
for extentions (so all IP based - no real phone lines).
I tried this (from voip-info.org wiki)...
exten => 1234,1,dial(sip/1234,20)
exten => 1234,2,playback(pls-wait-connect-call)
exten => 1234,3,Setvar(NewCaller=${CALLERIDNUM})
exten => 1234,4,SetCIDNum(0${CALLERIDNUM})
exten => 1234,5,dial(${TRUNK}c/9871234321,20,r)
exten => 1234,6,SetCIDNum(${NewCaller})
exten => 1234,7,voicemail2([EMAIL PROTECTED])
exten => 1234,101,voicemail2([EMAIL PROTECTED])
exten => 1234,102,hangup
Mine looks like this...
exten => 08700688nnn,1,Dial(SIP/operator,1,t)
exten => 08700688nnn,2,playback(pls-wait-connect-call)
exten => 08700688nnn,3,Setvar(NewCaller=${CALLERIDNUM})
exten => 08700688nnn,4,SetCIDNum(0${CALLERIDNUM})
exten => 08700688nnn,5,Dial(${TRUNK}/07961106nnn,20,r)
exten => 08700688nnn,6,SetCIDNum(${NewCaller})
exten => 08700688nnn,7,Voicemail(u100)
exten => 08700688nnn,8,Hangup()
exten => 08700688nnn,101,Voicemail(b100)
exten => 08700688nnn,102,Hangup()
(where nnn is a real number)
The sip channel is set to time out quickly for testing.
And I don't appear to have the pls-wait-connect-call audio file - but
that isn't an issue for the time being...
The IAX2/0870n is the extention/device that calls go out on via
voiptalk... (my call provider)...
If I include the c/ in the TRUNK line I get...
-- Executing Dial("IAX2/[EMAIL PROTECTED]:4569-1",
"c/07961106nnn|20|r") in new stack
May 18 10:37:40 WARNING[24416]: channel.c:1957 ast_request: No channel
type registered for 'c'
May 18 10:37:40 NOTICE[24416]: app_dial.c:927 dial_exec_full: Unable to
create channel of type 'c' (cause 66)
Asterisk shows this from the moment the sip channel is considered not to
have answered (1 sec)...
-- Nobody picked up in 1000 ms
-- Executing Playback("IAX2/[EMAIL PROTECTED]:4569-1",
"pls-wait-connect-call") in new stack
May 18 10:20:26 WARNING[24416]: file.c:486 ast_openstream_full: File
pls-wait-connect-call does not exist in any format
May 18 10:20:26 WARNING[24416]: file.c:790 ast_streamfile: Unable to
open pls-wait-connect-call (format ilbc): No such file or directory
May 18 10:20:26 WARNING[24416]: app_playback.c:83 playback_exec:
ast_streamfile failed on IAX2/[EMAIL PROTECTED]:4569-1 for
pls-wait-connect-call
-- Executing SetVar("IAX2/[EMAIL PROTECTED]:4569-1",
"NewCaller=01202843nnn") in new stack
-- Executing SetCIDNum("IAX2/[EMAIL PROTECTED]:4569-1",
"001202843nnn") in new stack
-- Executing Dial("IAX2/[EMAIL PROTECTED]:4569-1",
"/07961106nnn|20|r") in new stack
May 18 10:20:26 WARNING[24416]: channel.c:1957 ast_request: No channel
type registered for ''
May 18 10:20:26 NOTICE[24416]: app_dial.c:927 dial_exec_full: Unable to
create channel of type '' (cause 66)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing SetCIDNum("IAX2/[EMAIL PROTECTED]:4569-1",
"01202843nnn") in new stack
-- Executing VoiceMail("IAX2/[EMAIL PROTECTED]:4569-1",
"u100") in new stack
-- Playing '/var/spool/asterisk/voicemail/default/100/unavail'
(language 'en')
Again - I'm not worried about the audio file warning - I can fix that
later... I guess this is the important bit...
-- Executing Dial("IAX2/[EMAIL PROTECTED]:4569-1",
"/07961106nnn|20|r") in new stack
May 18 10:20:26 WARNING[24416]: channel.c:1957 ast_request: No channel
type registered for ''
May 18 10:20:26 NOTICE[24416]: app_dial.c:927 dial_exec_full: Unable to
create channel of type '' (cause 66)
== Everyone is busy/congested at this time (1:0/0/1)
The call then drops into voicemail...
I've tried various permuations but still no call is made to the mobile
number. Any ideas?
Cheers,
Mark
___
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
Re: [Asterisk-Users] Call forward...
I should mention that I have tried using the call forward function of the sip phones, but a) this means configuring the phones and some are remote and behind firewalls and b) It doesn't work... Cheers, Mark ___ 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] Call forwarding...
Sorry for posting this again, but it seems to have become attached to
another thread. Guess I replied to another message instead of starting a
new one...
Hi,
I'm trying to setup a call forwarding rule so that when an extention
doesn't answer the call is forwarded to my mobile.
I'm using voiptalk.org for incoming and outgoing calls and SIP phones
for extentions (so all IP based - no real phone lines).
I tried this (from voip-info.org wiki)...
exten => 1234,1,dial(sip/1234,20)
exten => 1234,2,playback(pls-wait-connect-call)
exten => 1234,3,Setvar(NewCaller=${CALLERIDNUM})
exten => 1234,4,SetCIDNum(0${CALLERIDNUM})
exten => 1234,5,dial(${TRUNK}c/9871234321,20,r)
exten => 1234,6,SetCIDNum(${NewCaller})
exten => 1234,7,voicemail2([EMAIL PROTECTED])
exten => 1234,101,voicemail2([EMAIL PROTECTED])
exten => 1234,102,hangup
Mine looks like this...
exten => 08700688nnn,1,Dial(SIP/operator,1,t)
exten => 08700688nnn,2,playback(pls-wait-connect-call)
exten => 08700688nnn,3,Setvar(NewCaller=${CALLERIDNUM})
exten => 08700688nnn,4,SetCIDNum(0${CALLERIDNUM})
exten => 08700688nnn,5,Dial(${TRUNK}/07961106nnn,20,r)
exten => 08700688nnn,6,SetCIDNum(${NewCaller})
exten => 08700688nnn,7,Voicemail(u100)
exten => 08700688nnn,8,Hangup()
exten => 08700688nnn,101,Voicemail(b100)
exten => 08700688nnn,102,Hangup()
(where nnn is a real number)
The sip channel is set to time out quickly for testing.
And I don't appear to have the pls-wait-connect-call audio file - but
that isn't an issue for the time being...
The IAX2/0870n is the extention/device that calls go out on via
voiptalk... (my call provider)...
If I include the c/ in the TRUNK line I get...
-- Executing Dial("IAX2/[EMAIL PROTECTED]:4569-1",
"c/07961106nnn|20|r") in new stack
May 18 10:37:40 WARNING[24416]: channel.c:1957 ast_request: No channel
type registered for 'c'
May 18 10:37:40 NOTICE[24416]: app_dial.c:927 dial_exec_full: Unable to
create channel of type 'c' (cause 66)
Asterisk shows this from the moment the sip channel is considered not to
have answered (1 sec)...
-- Nobody picked up in 1000 ms
-- Executing Playback("IAX2/[EMAIL PROTECTED]:4569-1",
"pls-wait-connect-call") in new stack
May 18 10:20:26 WARNING[24416]: file.c:486 ast_openstream_full: File
pls-wait-connect-call does not exist in any format
May 18 10:20:26 WARNING[24416]: file.c:790 ast_streamfile: Unable to
open pls-wait-connect-call (format ilbc): No such file or directory
May 18 10:20:26 WARNING[24416]: app_playback.c:83 playback_exec:
ast_streamfile failed on IAX2/[EMAIL PROTECTED]:4569-1 for
pls-wait-connect-call
-- Executing SetVar("IAX2/[EMAIL PROTECTED]:4569-1",
"NewCaller=01202843nnn") in new stack
-- Executing SetCIDNum("IAX2/[EMAIL PROTECTED]:4569-1",
"001202843nnn") in new stack
-- Executing Dial("IAX2/[EMAIL PROTECTED]:4569-1",
"/07961106nnn|20|r") in new stack
May 18 10:20:26 WARNING[24416]: channel.c:1957 ast_request: No channel
type registered for ''
May 18 10:20:26 NOTICE[24416]: app_dial.c:927 dial_exec_full: Unable to
create channel of type '' (cause 66)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing SetCIDNum("IAX2/[EMAIL PROTECTED]:4569-1",
"01202843nnn") in new stack
-- Executing VoiceMail("IAX2/[EMAIL PROTECTED]:4569-1",
"u100") in new stack
-- Playing '/var/spool/asterisk/voicemail/default/100/unavail'
(language 'en')
Again - I'm not worried about the audio file warning - I can fix that
later... I guess this is the important bit...
-- Executing Dial("IAX2/[EMAIL PROTECTED]:4569-1",
"/07961106nnn|20|r") in new stack
May 18 10:20:26 WARNING[24416]: channel.c:1957 ast_request: No channel
type registered for ''
May 18 10:20:26 NOTICE[24416]: app_dial.c:927 dial_exec_full: Unable to
create channel of type '' (cause 66)
== Everyone is busy/congested at this time (1:0/0/1)
The call then drops into voicemail...
I've tried various permuations but still no call is made to the mobile
number. Any ideas?
Cheers,
Mark
I should mention that I have tried using the call forward function of
the sip phones, but a) this means configuring the phones and some are
remote and behind firewalls and b) It doesn't work...
___
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
Re: [Asterisk-Users] Call forwarding...
Er... set the trunk variable to what? I thought it was a built in variable... Peter Bowyer wrote: Have you set the TRUNK variable in the [globals] section of extensions.conf? Looks like you didn't. Peter ___ 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
Re: [Asterisk-Users] Call forwarding...
I have been able to get it working by explicitly setting the dial command... So should the trunk variable be the divice to dial out on? Mark Benson wrote: Er... set the trunk variable to what? I thought it was a built in variable... Peter Bowyer wrote: Have you set the TRUNK variable in the [globals] section of extensions.conf? Looks like you didn't. Peter ___ 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 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
Re: [Asterisk-Users] Call forwarding...
Thanks, Staring to see where I was going wrong. Now I know the explicit dial string (as you say I tried that in the dial plan and it worked) I can mess around with the trunk variable. Cheers! Peter Bowyer wrote: On 18/05/05, Mark Benson <[EMAIL PROTECTED]> wrote: Er... set the trunk variable to what? I thought it was a built in variable... No, it's not. Looking at your dialplan extract, you need to set TRUNK to the name of the trunk to place the outgoing call on. eg TRUNK=IAX/voiptalk You might need to mess around to get the dialstring to end up in the right format for the provider you're using, also. Or imbed it directly in the dialplan. Peter ___ 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
Re: [Asterisk-Users] play gsm files in windows
http://www.nch.com.au/wavepad/ Brett, Gary wrote: Does anybody know of a WINDOWS application (preferably freeware) that will simply playback asterisk GSM sound files, I don't want to record them, just playback the ones that are currently there. Any help would be greatly appreciated cheers ___ 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 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
Re: [Asterisk-Users] Portable USB headset for VoIP
How about using a bluetooth headset? You would just need a bluetooth dongle for the laptop to provide the wireless connection for the headset... Mark (i'm in the process of trying this with an old usb bluetooth dongle (trying to find a suitable driver and manufacturers appears to have disapeared) and a cheap headset - i'll let you know how I get on.) Forrest W. Christian wrote: I'm trying to find a voip-suitable USB headset (I.E. headphones + microphone) which I can use with my laptop while I'm traveling and using Firefly or another softphone. I'm currently using a Logitech headset which works well (except the echo it generates toward the other caller when I turn up the gains too high), but it just doesn't carry well - in fact, I can't carry it in my laptop case any more just becuase it doesn't fit and it was getting very beat up. I'd like to find something which folds up and is designed for travel. It has to be USB sicne I don't have a MIC in (just line) on my laptop. Any ideas? -forrest ___ 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 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
Re: [Asterisk-Users] Portable USB headset for VoIP
If you plan to go this route don't buy a bluetooth adaptor that uses the XTNDconntect software. I've never been able to get it to work properly and there are no updates since last year (from the hardware vendor at least). Its an Innovision Wavelinker USB bluetooth module. I can discover and pair the headset, but thats it... nothing... Have ordered a new bluetooth dongle - so will try that and see what happens. Basically, you should just be able to pair the headset and then select it as an audio device in the multimedia settings in control panel (in windows) - when you disconnect the machine should revert back to the default hardware for audio playback. Mark Benson wrote: How about using a bluetooth headset? You would just need a bluetooth dongle for the laptop to provide the wireless connection for the headset... Mark (i'm in the process of trying this with an old usb bluetooth dongle (trying to find a suitable driver and manufacturers appears to have disapeared) and a cheap headset - i'll let you know how I get on.) Forrest W. Christian wrote: I'm trying to find a voip-suitable USB headset (I.E. headphones + microphone) which I can use with my laptop while I'm traveling and using Firefly or another softphone. I'm currently using a Logitech headset which works well (except the echo it generates toward the other caller when I turn up the gains too high), but it just doesn't carry well - in fact, I can't carry it in my laptop case any more just becuase it doesn't fit and it was getting very beat up. I'd like to find something which folds up and is designed for travel. It has to be USB sicne I don't have a MIC in (just line) on my laptop. Any ideas? -forrest ___ 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 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 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
Re: [Asterisk-Users] Portable USB headset for VoIP
I've been able to get it working with some diffent software. Once you have the devices paired and a headset profile available (the XTNDconnect software didn't have much in the way of bluetooth profiles available - but the widcomm software has loads - including the headset profile) you should see a bluetooth audio option in your multimedia settings in the control pannel (the drop downs for prefered audio devices) select bluetooth audio for audio in and out, apply that. Then connect to the headset. When I did this I didn't hear much except a few bleeps in the headset. Just press the button on the headset as well to connect and hey presto - iTunes on my headset! This works perfectly with xlite to make and receive calls. Hope this helps! Mark Mark Benson wrote: If you plan to go this route don't buy a bluetooth adaptor that uses the XTNDconntect software. I've never been able to get it to work properly and there are no updates since last year (from the hardware vendor at least). Its an Innovision Wavelinker USB bluetooth module. I can discover and pair the headset, but thats it... nothing... Have ordered a new bluetooth dongle - so will try that and see what happens. Basically, you should just be able to pair the headset and then select it as an audio device in the multimedia settings in control panel (in windows) - when you disconnect the machine should revert back to the default hardware for audio playback. Mark Benson wrote: How about using a bluetooth headset? You would just need a bluetooth dongle for the laptop to provide the wireless connection for the headset... Mark (i'm in the process of trying this with an old usb bluetooth dongle (trying to find a suitable driver and manufacturers appears to have disapeared) and a cheap headset - i'll let you know how I get on.) Forrest W. Christian wrote: I'm trying to find a voip-suitable USB headset (I.E. headphones + microphone) which I can use with my laptop while I'm traveling and using Firefly or another softphone. I'm currently using a Logitech headset which works well (except the echo it generates toward the other caller when I turn up the gains too high), but it just doesn't carry well - in fact, I can't carry it in my laptop case any more just becuase it doesn't fit and it was getting very beat up. I'd like to find something which folds up and is designed for travel. It has to be USB sicne I don't have a MIC in (just line) on my laptop. Any ideas? -forrest ___ 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 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 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 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 comple on FC2
I wouldn't normally post this to the asterisk mailing list but I'm really stuck... I've been trying to get meetme working on and off for a few months now but I always hit a brick wall when trying to compile. I keep seeing this... 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 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o zttool.o zttool.c cc -o zttool zttool.o -lnewt _*You do not appear to have the kernel sources for your current kernel installed.*_ make: *** [linux26] Error 1 I'm running FC2. I have the current kernel source, its linked to /usr/src/linux and all seems ok (from what I can gather at least). Also this is the latest stable release of zaptel. Any ideas? I'm guessing this is a problem with FC and the kernel source and dirs etc but I'm stumped. Cheers, Mark ___ 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
Re: [Asterisk-Users] Zaptel comple on FC2
Er.. sorry should have thought about this one a bit more - so bogged down in what isn't working that I forgot that I'm actually trying to complie ztdummy on a 2.6 kernel... Mark Benson wrote: I wouldn't normally post this to the asterisk mailing list but I'm really stuck... I've been trying to get meetme working on and off for a few months now but I always hit a brick wall when trying to compile. I keep seeing this... 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 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o zttool.o zttool.c cc -o zttool zttool.o -lnewt _*You do not appear to have the kernel sources for your current kernel installed.*_ make: *** [linux26] Error 1 I'm running FC2. I have the current kernel source, its linked to /usr/src/linux and all seems ok (from what I can gather at least). Also this is the latest stable release of zaptel. Any ideas? I'm guessing this is a problem with FC and the kernel source and dirs etc but I'm stumped. Cheers, Mark ___ 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 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
Re: [Asterisk-Users] Polycom 500...
Are you using inband DTMF? There are other options but I don't know much about the polycom phones. I have noticed that sometimes when accessing voicemail, it will 'miss' some dtmf tones if they are too short. This doesn't explain the number changing, unless your dial plan is putting in the leading zero. Carlos Chavez wrote: I am having a strange problem with a couple of Polycom IP 500 phones. I know this is not related to Asterisk, but maybe someone here had the same problem. I configured my phones following the documentation at voip-info.org and they are working very well. The only problem I have is that when I dial an extension like 1100 the phone changes that to 0110 and obviously the call fails. I have to dial slowly to get the 1100. Does this have anything to do with the dialplan? -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ 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 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
Re: [Asterisk-Users] A Few Questions
Perhaps he has a server that does other things besides asterisk and can't reformat it? Or perhaps he has a server in a remote location and buiness constraints make it difficult to take the time to get to it and spend a whole day doing a reinstall? Mark Dean Collins wrote: Dear Sir, Lol, with respectthat is the dumbest idea I have heard today. Your decision to not to invest time is a fallacy. [EMAIL PROTECTED] will have you up and running in 30 minutes or your money (it's free) back. Kind Regards, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of C. Hatton Humphrey Sent: Monday, 6 June 2005 10:09 AM Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] A Few Questions On 6/5/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: You should checkout [EMAIL PROTECTED], Thanks for the link - unfortunately [EMAIL PROTECTED] won't quite do the trick for me here as it is a complete OS replacement from what I can tell... I can't do that. I have too much time and money invested in the box that I'm running Asterisk on to wipe it and reload. Besides that, I already have Asterisk installed and running; maybe my next step should be to get AMP working on it (which would entail getting a webserver and whatever other requirements AMP has). ___ 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 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 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] Attended xfer
I've been going round in circles looking for user documentation for asterisk. There is plenty of documentation for setting it up (which I've done) but what I can't seem to find is an explanation of how to actually use it, specifically with regard to transferring calls. I am running 1.0.5 and can happily blind xfer from extension to extension, but I can't blind xfer. I have read various snippets about #2 or #8 or other such key combos, but nothing seems to let me do attended xfer. I have budgetone 102s that have an xfer button that puts incoming calls on hold but then sits with a dial tone and no matter what extension I dial it does nothing. Blind xfer with the # key works ok. From xlite I can blind xfer without problem but no attended xfer. Can anyone tell me what I should be doing to do attended xfer? As for the alternative to attended xfer, parking calls, I'm guessing this is just a case of blind xfering calls to a parking extension? Cheers for any help. Mark ___ 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
Re: [Asterisk-Users] Attended xfer
CVS in a production environment? Is that advisable? [EMAIL PROTECTED] wrote: I am running 1.0.5 and can happily blind xfer from extension to extension, but I can't blind xfer. I have read various snippets about #2 or #8 or other such key combos, but nothing seems to let me do attended xfer. From xlite I can blind xfer without problem but no attended xfer. For attendant transfer you should use CVS Head, in Asterisk stable is not implemented that feature! ___ 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 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] CVS in production env (Attended xfer)
Yesterday I asked about a user manual - ie a user guide to actually using asterisk (now on how to set it up) the doc project (v2) isn't anywhere near complete and is the closest thing I could find. Does anyone know of such a doc? The reason I ask is that while a lot of this may be obvious to many people its not to someone new to asterisk and there is a lot of info to trawl through, most of which is related to configuring asterisk. Again, the question of how to do attended xfers - how? (I now know I need asterisk from CVS) but what key presses? Then there is the issue of CVS in a production environment? I'm guessing people are actually doing this, but it goes against my better judgment. Does a roadmap for asterisk exist anywhere? If there was a roadmap I wouldn't need to ask when the next stable release will be available. By stable I don't mean that I think the CVS code isn't going to be reliable, (but that is a concern) but that the code isn't going to be changing constantly. ___ 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
Re: [Asterisk-Users] asterisk -vvvvvvvgrc?
asterisk -r attempts to connect to a running asterisk process (rather than starting another one) -v means be verbose (the more v's the more verbose) -c provide a control console for asterisk -g remove resource limit on core size - a debugging thing maybe? To find all this out for yourself and more try: man asterisk Muhammad Muzzamil Luqman wrote: what does the parameter -vvvgrc meanand are there any others as well? Kindest Muhammad Muzzamil Luqman ___ 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 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
Re: [Asterisk-Users] Sound of breathing
I've noticed that too (its not just when having phone sex either! :-). It depends on the phone being used (or is that abused)? I have a budgetone that really picks it up and a generic IN1800 (or something like that) that doesn't pick it up much at all. And when using soft phones, it depends on where the mic is (no jokes please). dean collins wrote: Yes stop using asterisk for phone sex !!! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hariharan Gopalan Sent: Tuesday, February 22, 2005 6:57 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Sound of breathing while using iax and a soft phone, the sound of breathing comes through so clearly that it has started bothering me. Earlier I was amazed at the quality, but now feel it is irritating. Wondering if there is a way to cut it down. I am in the process of exploring using iax for a call center, but this sound of breathing is a disappointment. Thanks Hari ___ 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 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 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
Re: [Asterisk-Users] MusicOnHold
Can you not just remove the sym link to the mpg123 process so asterisk doesn't find it therefore no music on hold? When I was trying to get music on-hold working I had to compile and sym link the mp123 executable - when it wasn't present I had no music on hold... Mark MF Hulber wrote: I'm looking for a simple way to disable MusicOnHold in my environment. I'm not really interested in having it and it causes too many problems with hanging mpg123 processes and memory management errors. The problem is, so many other modules seem to depend on it. I can't just cause a noload of MusicOnHold and be done. Does anyone have a simple solution? A solution that doesn't require a recompile is preferred but I'll appreciate and listen to any. MARK. ___ 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 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
Re: [Asterisk-Users] CallTransfer
I get the impression that the transfer/flash/recall etc etc buttons don't always work - it seems to depend on what phone/firmware you are using. And possibly the version of asterisk. 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. I have resorted to using # for blind xfer and *2 for attended xfer. Herman Cremer wrote: Hi I was wondering if there are any special settings that I need to be able to transfer calls. Whenever I press the 'recall' button, I just here a click, and no ring-tone to transfer. in my debug log I get this : -- Feb 24 09:09:27 DEBUG[19216]: Exception on 10, channel 1 Feb 24 09:09:27 DEBUG[19216]: Got event Pulse Start(14) on channel 1 (index 0) Feb 24 09:09:27 DEBUG[19216]: Exception on 10, channel 1 Feb 24 09:09:27 DEBUG[19216]: Got event Event 65585(65585) on channel 1 (index 0) Feb 24 09:09:27 DEBUG[19216]: Pulse dial '1' -- Any ideas ? Herman ___ 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 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
Re: [Asterisk-Users] Re: CallTransfer
I'm only just getting my head round asterisk - so the phones themselves have taken a back seat - I have only recently upgraded the phones to r .16 - so maybe they do work now. I'll test as soon as all my users have moved their phones over to the asterisk server. I only found out about the r .22 release today. Who writes the firmware? I'm guessing it isn't grandstream as they are only on .16 and they are running .22 as beta - its not important - just curious. And sorry Herman - I would have replied sooner but work got in the way - yeah features.conf - thats what I was gonna say. dean collins wrote: Yep, works for me to. I'm using revision 1.0.5.22 software and the [EMAIL PROTECTED] solution. I think it has worked since revision .16 Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aldo Bergamini Sent: Thursday, February 24, 2005 7:55 AM To: asterisk-users@lists.digium.com Subject: [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 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 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
Re: [Asterisk-Users] SIP Phone with headset
Grandstream are supposed to be releasing a BT103 ? Its a 100 series phone with headphone jack... when, I couldn't say though. Thibault Lamy wrote: Hi there, Do anyone have any experience with SIP phone that support a headset ? We have Budgetone phones but we need headsets. What would you advise ? Thanks Thibault ___ 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 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] ztdummy compile again
Hi,
I'm still strugling with getting an easy to use conference system
implemented. I did have app_conference running, but today I upgraded
asterisk to 1.0.9 and it stopped working. I've tried following the
instructions for compiling app_conference on 1.0.7 but it didn't work.
So I went back to ztdummy (I've not had any luck getting this to compile
on FC2).
Anyhoo, I've tried again and once again ztdummy fails to compile and the
various disparate instructions on what is needed to get it running are
not helping.
If I run make linux26 then the zaptel drivers start to compile but then
spews out a load of errors.
Anyone have any ideas?
SNIP===
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
./makefw pciradio.rbt radfw > radfw.h
Loaded 42096 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 libtonezone.a -lm
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 -o ztspeed.o -c ztspeed.c
cc -o ztspeed ztspeed.o
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o zttool.o zttool.c
cc -o zttool zttool.o -lnewt
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"zttest.c -o zttest
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o fxotune.o fxotune.c
cc -o fxotune fxotune.o -lm
/lib/modules/2.6.5-1.358/build
make -C /lib/modules/2.6.5-1.358/build SUBDIRS=/usr/src/zaptel modules
make[1]: Entering directory `/usr/src/linux-2.6.5-1.358'
Makefile:434: .config: No such file or directory
CC [M] /usr/src/zaptel/zaptel.o
In file included from /usr/src/zaptel/zconfig.h:9,
from /usr/src/zaptel/zaptel.c:40:
include/linux/config.h:4:28: linux/autoconf.h: No such file or directory
In file included from /usr/src/zaptel/zaptel.c:40:
/usr/src/zaptel/zconfig.h:10:27: linux/version.h: No such file or directory
/usr/src/zaptel/zconfig.h:68:41: missing binary operator before token "("
In file included from include/linux/kernel.h:11,
from /usr/src/zaptel/zaptel.c:42:
include/linux/linkage.h:5:25: asm/linkage.h: No such file or directory
In file included from include/linux/types.h:13,
from include/linux/kernel.h:13,
from /usr/src/zaptel/zaptel.c:42:
include/linux/posix_types.h:47:29: asm/posix_types.h: No such file or
directory
In file included from include/linux/kernel.h:13,
from /usr/src/zaptel/zaptel.c:42:
include/linux/types.h:14:23: asm/types.h: No such file or directory
In file included from include/linux/kernel.h:13,
from /usr/src/zaptel/zaptel.c:42:
include/linux/types.h:18: error: syntax error before "__kernel_dev_t"
include/linux/types.h:18: warning: type defaults to `int' in declaration
of `__kernel_dev_t'
include/linux/types.h:18: warning: data definition has no type or
storage class
include/linux/types.h:21: error: syntax error before "dev_t"
include/linux/types.h:21: warning: type defaults to `int' in declaration
of `dev_t'
include/linux/types.h:21: warning: data definition has no type or
storage class
include/linux/types.h:22: error: syntax error before "ino_t"
include/linux/types.h:22: warning: type defaults to `int' in declaration
of `ino_t'
include/linux/types.h:22: warning: data definition has no type or
storage class
include/linux/types.h:23: error: syntax error before "mode_t"
include/linux/types.h:23: warning: type defaults to `int' in declaration
of `mode_t'
include/linux/types.h:23: warning: data definition has no type or
storage class
include/linux/types.h:24: error: syntax error before "nlink_t"
include/linux/types.h:24: warning: type defaults to `int' in declaration
of `nlink_t'
include/linux/types.h:24: warning: data definition has no type or
storage class
include/linux/types.h:25: error: syntax error before "off_t"
include/linux/types.h:
Re: [Asterisk-Users] ztdummy compile again
When you say kernel development do you mean kernel sources (which I
have) or some other development tools/libs?
and a kernel build config file? make mrproper ? make oldconfig ? I've
done that much at least...
Mark
Kevin Collins wrote:
Looks like you don't have kernel development installed and a basic kernel build config file generated.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Benson
Sent: Friday, September 23, 2005 8:55 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] ztdummy compile again
Hi,
I'm still strugling with getting an easy to use conference system
implemented. I did have app_conference running, but today I upgraded
asterisk to 1.0.9 and it stopped working. I've tried following the
instructions for compiling app_conference on 1.0.7 but it didn't work.
So I went back to ztdummy (I've not had any luck getting this to compile
on FC2).
Anyhoo, I've tried again and once again ztdummy fails to compile and the
various disparate instructions on what is needed to get it running are
not helping.
If I run make linux26 then the zaptel drivers start to compile but then
spews out a load of errors.
Anyone have any ideas?
SNIP===
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
./makefw pciradio.rbt radfw > radfw.h
Loaded 42096 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 libtonezone.a -lm
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 -o ztspeed.o -c ztspeed.c
cc -o ztspeed ztspeed.o
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o zttool.o zttool.c
cc -o zttool zttool.o -lnewt
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"zttest.c -o zttest
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o fxotune.o fxotune.c
cc -o fxotune fxotune.o -lm
/lib/modules/2.6.5-1.358/build
make -C /lib/modules/2.6.5-1.358/build SUBDIRS=/usr/src/zaptel modules
make[1]: Entering directory `/usr/src/linux-2.6.5-1.358'
Makefile:434: .config: No such file or directory
CC [M] /usr/src/zaptel/zaptel.o
In file included from /usr/src/zaptel/zconfig.h:9,
from /usr/src/zaptel/zaptel.c:40:
include/linux/config.h:4:28: linux/autoconf.h: No such file or directory
In file included from /usr/src/zaptel/zaptel.c:40:
/usr/src/zaptel/zconfig.h:10:27: linux/version.h: No such file or directory
/usr/src/zaptel/zconfig.h:68:41: missing binary operator before token "("
In file included from include/linux/kernel.h:11,
from /usr/src/zaptel/zaptel.c:42:
include/linux/linkage.h:5:25: asm/linkage.h: No such file or directory
In file included from include/linux/types.h:13,
from include/linux/kernel.h:13,
from /usr/src/zaptel/zaptel.c:42:
include/linux/posix_types.h:47:29: asm/posix_types.h: No such file or
directory
In file included from include/linux/kernel.h:13,
from /usr/src/zaptel/zaptel.c:42:
include/linux/types.h:14:23: asm/types.h: No such file or directory
In file included from include/linux/kernel.h:13,
from /usr/src/zaptel/zaptel.c:42:
include/linux/types.h:18: error: syntax error before "__kernel_dev_t"
include/linux/types.h:18: warning: type defaults to `int' in declaration
of `__kernel_dev_t'
include/linux/types.h:18: warning: data definition has no type or
storage class
include/linux/types.h:21: error: syntax error before "dev_t"
include/linux/types.h:21: warning: type defaults to `int' in declaration
of `dev_t'
include/linux/types.h:21: warning: data definition has no type or
storage class
Re: [Asterisk-Users] ztdummy compile again
Kevin,
I've got the source package in /usr/src/linux-2.6.5-1.358
I also have sym links to it from /usr/src/linux and /usr/src/linux-2.6
and a sym link /lib/modules/2.6.5-1.358/build to /usr/src/linux (as
mentioned on voipinfo). If I ommit this last sym link then the complier
complains about only needing the lib headers and not the full kernel (or
something like that).
Mark
(I've checked and the .config is in the directory)
Unless you can think of something obvious I'll clean out the src
directory and redownload the kernel source and see if that helps.
Kevin Collins wrote:
Mark,
Have you checked to make sure your kernel source is in the following directory :
/usr/src/linux-2.6.5-1.358'
Makefile:434: .config: No such file or directory
It just seems to be complaining about not finding your kernel development source environment.
Kevin
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Benson
Sent: Friday, September 23, 2005 11:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] ztdummy compile again
When you say kernel development do you mean kernel sources (which I
have) or some other development tools/libs?
and a kernel build config file? make mrproper ? make oldconfig ? I've
done that much at least...
Mark
Kevin Collins wrote:
Looks like you don't have kernel development installed and a basic kernel build config file generated.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Benson
Sent: Friday, September 23, 2005 8:55 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] ztdummy compile again
Hi,
I'm still strugling with getting an easy to use conference system
implemented. I did have app_conference running, but today I upgraded
asterisk to 1.0.9 and it stopped working. I've tried following the
instructions for compiling app_conference on 1.0.7 but it didn't work.
So I went back to ztdummy (I've not had any luck getting this to compile
on FC2).
Anyhoo, I've tried again and once again ztdummy fails to compile and the
various disparate instructions on what is needed to get it running are
not helping.
If I run make linux26 then the zaptel drivers start to compile but then
spews out a load of errors.
Anyone have any ideas?
SNIP===
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
./makefw pciradio.rbt radfw > radfw.h
Loaded 42096 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 libtonezone.a -lm
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 -o ztspeed.o -c ztspeed.c
cc -o ztspeed ztspeed.o
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o zttool.o zttool.c
cc -o zttool zttool.o -lnewt
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"zttest.c -o zttest
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o fxotune.o fxotune.c
cc -o fxotune fxotune.o -lm
/lib/modules/2.6.5-1.358/build
make -C /lib/modules/2.6.5-1.358/build SUBDIRS=/usr/src/zaptel modules
make[1]: Entering directory `/usr/src/linux-2.6.5-1.358'
Makefile:434: .config: No such file or directory
CC [M] /usr/src/zaptel/zaptel.o
In file included from /usr/src/zaptel/zconfig.h:9,
from /usr/src/zaptel/zaptel.c:40:
include/linux/config.h:4:28: linux/autoconf.h: No such file or directory
In file included from /usr/src/zaptel/zaptel.c:40:
/usr/src/zaptel/zconfig.h:10:27: linux/version.h: No such file or directory
/usr/src/zaptel/zconfig.h:68:41: missing binary operator before token "("
In file included from
[Asterisk-Users] Phones no longer register - except one?
Hi I've got an interesting problem. A few days ago (maybe even a week or two) all my sip phones lost registrations with my asterisk box. All that is but one. The asterisk box is out on the internet, I have two phones at my location and 1 at another separate location. The only phone that remains registered is an Integrated Networks IN002 (or something like that). This is at my location. I also have a grandstream GXP-2000 that will not register. This is also at my location. I have tried xlite and sjphone (on my desktop and mobile phone (via wireless) respectivley) to test, These also fail to register. I have a Budgetone 102 at another location which also fails to register. There is nothing on the command line apart from the IN002 phone registering and talking to the * server. It dials in and out fine. The only thing I can see that mine and the remote location have in common is the ISP that provides DSL (plusnet), and I was wondering if they were limiting traffic as they have recently announced their own telephony service. But I doubt it and if that was the case then why does the IN002 register and not the budget tone? Its a crazy paranoid theory, but I can't think of anything else. The * is 1.0.9 and was working perfectly when I upgraded from a CVS version. Any ideas - I must be missing something obvious - but I've not changed anything since the upgrade. Any why one phone? Mark ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
