[Asterisk-Users] Unlock / install of Cisco 7940 IP Phone ?
Hello Guys, I just got my new Cisco 7940* IP Phone Unfortunately I can't find out how to setup this phone. Am I really that stupid ;-) ? I've read the Cisco Manuel, but everything I tried did not help anything. 1. When I turn on the phone it will display Configuring VLANConfiguring IP.. This message will not disappear. 2. I can see that the phone has a local IP. I can also access the IP over my LAN with http (only http, telnet does not work) 3 My Main menu will this show Configuring VLANConfiguring IP.. But if I click on settings, network settings it will show me the local IP of the phone Now, my question, what do I do wrong ? how can I get that phone installed with a sip image ? I tried to unlock the phone with **# but that does not do anything. Also, there is no unlock function in the phone menu (phone settings) This is a new Cisco phone, no sip image on it. Thank you for the help Matt ___ --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
AW: [Asterisk-Users] Unlock / install of Cisco 7940 IP Phone ?
Thx guys for the help. I was able to unlock it. I see the following under Network Configuration : 8 TFTP Server 1 On 32 I can see 32 Alternate TFTP NO Shall I type in the IP of the TFTP Server there ? if yes, how ? BTW: Im assuming that TFTP has to be installed on my local windown pc, and then let my phone access it over the local ip, correct ? Thx again for ur help! Grüsse / Best Regards Matt -- « hola! » see...habla espanol :-) -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von [EMAIL PROTECTED] Gesendet: Mittwoch, 7. Juni 2006 23:29 An: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [Asterisk-Users] Unlock / install of Cisco 7940 IP Phone ? To convert the phone to SIP you have to unlock and set the TFTP server to your TFTP server address as explained below. But... you also need to make sure you have a SIP image in the root directory of your TFTP server and also edit the OS79XX.TXT file to contain only the SIP image name (no .bin or .sbin extension). This should tell the phone what image it should be running and will then start a tftp transaction to download the SIP image. Once the SIP image is loaded on your phone, you will have to reset the TFTP Server addresses again so it can then download the SIP configuration files (SIPdefault.cnf and SIPMAC.cnf) good luck! -- Original message -- From: Aaron Daniel [EMAIL PROTECTED] You have to press settings, then **#, and wait a moment to make sure it unlocks. Then you can configure a tftp server to use. The alternative is to configure your dhcp server with a tftp server. On linux, that would be next-server ip/host in the subnet section. TFTP server on windows. On Wed, 7 Jun 2006, Mateo Meier wrote: Hello Guys, I just got my new Cisco 7940* IP Phone Unfortunately I can't find out how to setup this phone. Am I really that stupid ;-) ? I've read the Cisco Manuel, but everything I tried did not help anything. 1. When I turn on the phone it will display Configuring VLANConfiguring IP.. This message will not disappear. 2. I can see that the phone has a local IP. I can also access the IP over my LAN with http (only http, telnet does not work) 3 My Main menu will this show Configuring VLANConfiguring IP.. But if I click on settings, network settings it will show me the local IP of the phone Now, my question, what do I do wrong ? how can I get that phone installed with a sip image ? I tried to unlock the phone with **# but that does not do anything. Also, there is no unlock function in the phone menu (phone settings) This is a new Cisco phone, no sip image on it. Thank you for the help Matt ___ --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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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 ___ --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 ___ --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] VoIP connection US -- EU with ADSL a problem ?
Hello I currently have a softphone (Firefly Soft Phone) here in Miami with a ADSL Connection (3 Mbps/ 384 Kbps) connected to a Asterisk Server in Switzerland with a ADSL (1,2 Mbps/ 200 Kbps). (both ADSL connection are not very busy. Most of the time only 1 phone call is active ) I keep having problem with the connection.. Does anybody know if the above connection and or the distance could be a problem? Thank you for the help Matt ___ --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
AW: [Asterisk-Users] VoIP connection US -- EU with ADSL a problem ?
Pings are stable + / - 150 MS Grüsse / Best Regards Mateo Meier -- « hola! » see...habla espanol :-) -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Sam Tam Gesendet: Samstag, 19. November 2005 22:21 An: 'Asterisk Users Mailing List - Non-Commercial Discussion' Betreff: RE: [Asterisk-Users] VoIP connection US -- EU with ADSL a problem ? Well try to setup some QoS service on both router to let VoIP calls take priority over any others data. Also try to do some pinging test for 1 day or so and see if you are suffering from any packet loss. Packet loss can do a lot of harm to VoIP calls.. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mateo Meier Sent: 19 November 2005 21:14 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] VoIP connection US -- EU with ADSL a problem ? Hello I currently have a softphone (Firefly Soft Phone) here in Miami with a ADSL Connection (3 Mbps/ 384 Kbps) connected to a Asterisk Server in Switzerland with a ADSL (1,2 Mbps/ 200 Kbps). (both ADSL connection are not very busy. Most of the time only 1 phone call is active ) I keep having problem with the connection.. Does anybody know if the above connection and or the distance could be a problem? Thank you for the help Matt ___ --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 ___ --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 ___ --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
[Asterisk-Users] Music Volume ?
Hey guys Anybody knows how to turn up the volume of a Music on Hold Mp3 file ? When I play it on my windows box, volume is perfect.. but when I use it Music on hold.. the volume is very low. Maybe there is a general setting for asterisk volume ? Thx for the help Matt ___ 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
AW: [Asterisk-Users] Music Volume ?
What do you mean ? My etc/asterisk/musiconhold.conf looks like that: [EMAIL PROTECTED] root]# more /etc/asterisk/musiconhold.conf ; ; Music on hold class definitions ; [classes] default = quietmp3:/var/lib/asterisk/mohmp3 ;loud = mp3:/var/lib/asterisk/mohmp3 ;random = quietmp3:/var/lib/asterisk/mohmp3,-z ;unbuffered = mp3nb:/var/lib/asterisk/mohmp3 ;quietunbuf = quietmp3nb:/var/lib/asterisk/mohmp3 ; Note that the custom mode cannot handle escaped parameters (specifically embedded spaces) ;manual = custom:/var/lib/asterisk/mohmp3,/usr/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s Grüsse / Best Regards Mateo Meier - Don't marry for money; you can borrow it cheaper ;-) -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von C F Gesendet: Montag, 7. März 2005 01:43 An: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [Asterisk-Users] Music Volume ? Check out musiconhold.conf you can use loud On Mon, 7 Mar 2005 00:02:27 +0100, Mateo Meier [EMAIL PROTECTED] wrote: Hey guys Anybody knows how to turn up the volume of a Music on Hold Mp3 file ? When I play it on my windows box, volume is perfect.. but when I use it Music on hold.. the volume is very low. Maybe there is a general setting for asterisk volume ? Thx for the help Matt ___ 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 Spam detection software, running on the system zeus.avanzada7.com, has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: Check out musiconhold.conf you can use loud On Mon, 7 Mar 2005 00:02:27 +0100, Mateo Meier [EMAIL PROTECTED] wrote: Hey guys Anybody knows how to turn up the volume of a Music on Hold Mp3 file ? When I play it on my windows box, volume is perfect.. but when I use it Music on hold.. the volume is very low. Maybe there is a general setting for asterisk volume ? Thx for the help MattAsterisk-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 [...] Content analysis details: (0.1 points, 5.0 required) pts rule name description -- -- 0.0 RCVD_BY_IP Received by mail server with no name 0.1 FORGED_RCVD_HELO Received: contains a forged HELO ___ 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
AW: [Asterisk-Users] Call Forwarding to Cell Phone, Pager, etc
Yeh.. doing something like this..
With Capi..
exten =1,1,Answer
and later on...
exten = 1,1,Dial(capi/72045**:MY-OFFICE,18)
exten = 1,2,Goto(2-${DIALSTATUS},1)
exten = 1-NOANSWER,1,Dial(capi/72045**:MY mobile)
exten = 1-CHANUNAVAIL,1,Goto(1,1)
exten = 1-BUSY,1,Dial(capi/72045**:AND AND AND)
BTW:
Replace MY-OFFICE with your office number(same with mobile.. )
Also, replace 72045** with the number your calling from ( if u are using
ISDN / CAPI)
Grüsse / Best Regards
Mateo Meier
-
Don't marry for money; you can borrow it cheaper ;-)
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Nitesh
Divecha
Gesendet: Mittwoch, 2. März 2005 20:56
An: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Betreff: [Asterisk-Users] Call Forwarding to Cell Phone, Pager, etc
Hello all,
Was just wondering if Asterisk can do Call forwarding to cell phones,
pagers, home phone, etc.
For example, if exten 202 is away, he will set his call blasting priority
like first ring the exten for 10 sec if not answered then ring cell number
for 10 sec again if not answered, then ring home and etc.
Like a call blasting priority...
Any help would be appreciated.
Neel
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[Asterisk-Users] Music on hold..Mar error res_musiconhold.c:309 monmp3thread: Request to schedule in the past ?
Hey guys. Im trying to setup Music on Hold. If I transfer a call (with dial) I like to put the call on Music on hold.. Here's what I've tried so far: On my I extensions.conf exten =1,1,WaitMusicOnHold(30) exten =1,2,Dial(SIP/mateo,18) exten =1,3,VoiceMail(1001) I have also added this line to [context].. So it looks like that: ;[context] musiconhold=default Additinaly, when I pikc up the call I've done: [mainmenue] exten = s,1,Answer exten = s,2,SetMusicOnHold(default) exten = s,3,DigitTimeout,5 exten = s,4,ResponseTimeout,10 my musiconhold.conf looks like: ; ; Music on hold class definitions ; [classes] default = quietmp3:/var/lib/asterisk/mohmp3 ;loud = mp3:/var/lib/asterisk/mohmp3 ;random = quietmp3:/var/lib/asterisk/mohmp3,-z ;unbuffered = mp3nb:/var/lib/asterisk/mohmp3 ;quietunbuf = quietmp3nb:/var/lib/asterisk/mohmp3 ; Note that the custom mode cannot handle escaped parameters (specifically embedded spaces) ;manual = custom:/var/lib/asterisk/mohmp3,/usr/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s I did' put any mp3 files on my asterisk box.. but I can see there are some standard file on /var/lib/asterisk/mohmp3 I have also installed mpg123.. not 100% sure if I did everything ok.. but it seems like it: [EMAIL PROTECTED] root]# mpg123 High Performance MPEG 1.0/2.0 Audio Player for Layer 1, 2 and 3. Version 0.59g (97/04/23). Written and copyrights by Michael Hipp. Uses code from various people. See 'README' for more! THIS SOFTWARE COMES WITH ABSOLUTELY NO WARRANTY! USE AT YOUR OWN RISK! usage: mpg123 [option(s)] [file(s) | URL(s) | -] Anybody knows why I get a res_musiconhold.c:309 monmp3thread: Request to schedule in the past error when I try to call asterisk ? Thx for the help ___ 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
AW: [Asterisk-Users] Transfer a call ? Am I lookingfortheflashcommand ?
Hello Jim,
I tryed that with capi.. but no luke. It will hang up the line anyway :-(
exten = s,1,Playback(transfer)
exten = s,2,Flash(capi/72044**:041720,18)
exten = s,3,SendDTMF(${ARG1})
exten = s,4,Hangup()
Any idears why ?
BTW: Whats actually that SendDTMF ? thing ?
Thx for the help..
Grüsse / Best Regards
Mateo Meier
-
Don't marry for money; you can borrow it cheaper ;-)
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Jim Van
Meggelen
Gesendet: Samstag, 26. Februar 2005 07:54
An: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion'
Betreff: RE: [Asterisk-Users] Transfer a call ? Am I
lookingfortheflashcommand ?
[EMAIL PROTECTED] wrote:
Hello Jim,
thx for the answer..
Im happy I found someone that is using flash :)
It's not perfect, but it can be useful.
Am I right, if I transfer a call with flash, the line will be free
afterwards ?
Yep
Would you mind to past me how you did the flash part @the
extention file ? Also, If I use flash, do I have to setup
anything else or just @the extention file ?
Jere's the relevant section of my dial plan:
[macro-cell_user]
exten = s,1,Playback(transfer)
exten = s,2,Flash(zap/1)
exten = s,3,SendDTMF(${ARG1})
exten = s,4,Hangup()
Good luck!
Jim.
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Jim
Van Meggelen Gesendet: Freitag, 25. Februar 2005 05:57
An: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Betreff: RE: [Asterisk-Users] Transfer a call ? Am I looking for
theflashcommand ?
[EMAIL PROTECTED] wrote:
On Fri, 2005-02-25 at 00:50 +0100, Mateo Meier wrote:
Hey Guys
Im trying to forward a call with asterisk to a regular phone.
Something like I get a call on my regular phone, and he's trying
to reach some buddy of mine.. then I tell him wait a sec and push
Flash and get a other dialtone.. then I dial that other number
then hangup the phone, so the one that called will be connected to
where I dialed it to...
Some buddy of mine told me im looking for a function called flash
Only thing Im able to find is:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Flash
Im unsure how to use it now..
Let's say if I forward a call with asterisk as following: exten =
2,1,Dial(capi/720:07812345*,18)
How would I use the flash command to transfer that call above to 078
12345* ? I have no problem transferring a call, but when Im doing
this with the dial command (see above).. then my line will be busy
Been covered before, You can't do that on an analog line. Problem
comes from where you are and what flash would be working on at that
point. If you flash asterisk and get dialtone again, you are getting
the dialtone
from asterisk. At this point the only channel being worked is the one
you are on and flashing it won't help.
What you would need to do is get the other leg of the call to make
the flash.
It might be really handy to be able to specify the trunk to
flash() as an argument. I use flash in my dialplan to
transfer incoming calls to my cell phone when I'm out and
about - frees up the line and reduces attenuation caused by
an analog trombone. It'd be handy to be able to use it to
transfer terminated calls as well.
Of course if you where on a PRI link, you could do hairpinning,
ect or tromboning and get the call taken back by the PSTN and
transferred to the new number.
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Content preview: [EMAIL PROTECTED] wrote: Hello
Jim, thx for the answer.. Im happy I found someone that is using
flash :) It's not perfect, but it can be useful. Am I right, if I
transfer a call with flash, the line will be free afterwards ? [...]
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AW: [Asterisk-Users] Transfer a call ? Am I looking for theflashcommand ?
Hello Jim, thx for the answer.. Im happy I found someone that is using flash :) Am I right, if I transfer a call with flash, the line will be free afterwards ? Would you mind to past me how you did the flash part @the extention file ? Also, If I use flash, do I have to setup anything else or just @the extention file ? Grüsse / Best Regards Mateo Meier - Don't marry for money; you can borrow it cheaper ;-) -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Jim Van Meggelen Gesendet: Freitag, 25. Februar 2005 05:57 An: 'Asterisk Users Mailing List - Non-Commercial Discussion' Betreff: RE: [Asterisk-Users] Transfer a call ? Am I looking for theflashcommand ? [EMAIL PROTECTED] wrote: On Fri, 2005-02-25 at 00:50 +0100, Mateo Meier wrote: Hey Guys Im trying to forward a call with asterisk to a regular phone. Something like I get a call on my regular phone, and he's trying to reach some buddy of mine.. then I tell him wait a sec and push Flash and get a other dialtone.. then I dial that other number then hangup the phone, so the one that called will be connected to where I dialed it to... Some buddy of mine told me im looking for a function called flash Only thing Im able to find is: http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Flash Im unsure how to use it now.. Let's say if I forward a call with asterisk as following: exten = 2,1,Dial(capi/720:07812345*,18) How would I use the flash command to transfer that call above to 078 12345* ? I have no problem transferring a call, but when Im doing this with the dial command (see above).. then my line will be busy Been covered before, You can't do that on an analog line. Problem comes from where you are and what flash would be working on at that point. If you flash asterisk and get dialtone again, you are getting the dialtone from asterisk. At this point the only channel being worked is the one you are on and flashing it won't help. What you would need to do is get the other leg of the call to make the flash. It might be really handy to be able to specify the trunk to flash() as an argument. I use flash in my dialplan to transfer incoming calls to my cell phone when I'm out and about - frees up the line and reduces attenuation caused by an analog trombone. It'd be handy to be able to use it to transfer terminated calls as well. Of course if you where on a PRI link, you could do hairpinning, ect or tromboning and get the call taken back by the PSTN and transferred to the new number. -- -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.4.0 - Release Date: 22/02/2005 ___ 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
WG: AW: [Asterisk-Users] Transfer a call ? Am I looking for theflashcommand ?
Hey.. Your saying I can not use flash with ISDN ? What options to I have to transfer a call directly ? ( So I have a free line afterwords) What interface are you using? ZapBRI? if so you might be able to do the hairpinning if it is supported. Im not using any interface.. But if you know how to do that, let me know and I install that interface. Thx for your answer :) Grüsse / Best Regards Mateo Meier - Don't marry for money; you can borrow it cheaper ;-) -Ursprüngliche Nachricht- Von: Steven Critchfield [mailto:[EMAIL PROTECTED] Gesendet: Freitag, 25. Februar 2005 02:38 An: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Betreff: Re: AW: [Asterisk-Users] Transfer a call ? Am I looking for theflashcommand ? On Fri, 2005-02-25 at 02:21 +0100, Mateo Meier wrote: Hey Steven, It's actully ISDN.. not a analog line :) Will that change anything :) ? Yes as I do not believe flash is something you can do on ISDN at all. What interface are you using? ZapBRI? if so you might be able to do the hairpinning if it is supported. Been covered before, You can't do that on an analog line. Problem comes from where you are and what flash would be working on at that point. If you flash asterisk and get dialtone again, you are getting the dialtone from asterisk. At this point the only channel being worked is the one you are on and flashing it won't help. What you would need to do is get the other leg of the call to make the flash. Of course if you where on a PRI link, you could do hairpinning, ect or tromboning and get the call taken back by the PSTN and transferred to the new number. -- Steven Critchfield [EMAIL PROTECTED] -- Steven Critchfield [EMAIL PROTECTED] ___ 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] Transfer a call ? Am I looking for the flash command ?
Hey Guys Im trying to forward a call with asterisk to a regular phone. Something like I get a call on my regular phone, and he's trying to reach some buddy of mine.. then I tell him wait a sec and push Flash and get a other dialtone.. then I dial that other number then hangup the phone, so the one that called will be connected to where I dialed it to... Some buddy of mine told me im looking for a function called flash Only thing Im able to find is: http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Flash Im unsure how to use it now.. Let's say if I forward a call with asterisk as following: exten = 2,1,Dial(capi/720:07812345*,18) How would I use the flash command to transfer that call above to 078 12345* ? I have no problem transferring a call, but when Im doing this with the dial command (see above).. then my line will be busy Thx for all your help.. ;-)) Matt ___ 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
AW: [Asterisk-Users] Transfer a call ? Am I looking for the flashcommand ?
Hey Steven, It's actully ISDN.. not a analog line :) Will that change anything :) ? Thx for the help, Matt Been covered before, You can't do that on an analog line. Problem comes from where you are and what flash would be working on at that point. If you flash asterisk and get dialtone again, you are getting the dialtone from asterisk. At this point the only channel being worked is the one you are on and flashing it won't help. What you would need to do is get the other leg of the call to make the flash. Of course if you where on a PRI link, you could do hairpinning, ect or tromboning and get the call taken back by the PSTN and transferred to the new number. -- Steven Critchfield [EMAIL PROTECTED] ___ 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
AW: [Asterisk-Users] mp3 to gsm?
Take a look at this URL: http://www.voip-info.org/tiki-index.php?page=Asterisk%20sound%20files Ive used the following command sox inputfile.wav -r 8000 -c 1 outputfile.gsm resample -ql hope this helps Grüsse / Best Regards Mateo Meier - Don't marry for money; you can borrow it cheaper ;-) Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Muhammad Muzzamil Luqman Gesendet: Dienstag, 22. Februar 2005 17:56 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] mp3 to gsm? i have got a music file with extension mp3 and it is not workign with background() is there any way to convert the mp3 to gsm or any other codec? 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] Anybody using X-Lite Softphone ? tryed to forward a call to X-Lite....
Hey Guys Im trying to forward a call from the asterisk mainmenue to my second computer with X-Lite installed.. What I've done so far is this: Installed X-lite @my win PC.. X-Lite configuration: Menu | System Settings | SIP Proxy | default Display Name: mateo01 User Name Authorization User: mateo01 Password: Domain/Realm: 192.168.1.** SIP Proxy: 192.168.1.** 192.168.1.** = IP address of Asterisk and the sip.conf file looks like that: [mateo01] type=friend username=mateo01 callerid=mateo01 1234 host=dynamic secret= disallow=all allow=gsm allow=ulaw allow=alaw context=sip nat=no Now, Im unsure what to do ? whats next ? and what do I type in to extensions.conf instead of the following: exten=2,1,Dial(capi/720:078***) Thx for the help Mateo ___ 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 ECT (explicit call transfer) ?
Hey Guys
Im trying to find out how to transfer a call with ECT (explicit call
transfer) ?
Im currently transferring a call as following:
exten=2,1,Dial(capi/720:07,18)
exten = 2,2,Goto(2-${DIALSTATUS},1)
exten = 2-NOANSWER,1,Dial(capi/720:07979)
exten = 2-CHANUNAVAIL,1,Goto(1,1)
exten = 2-BUSY,1,Dial(capi/720:07979)
If I wanna transfer a call with ECT (call deflection), do I'll do that in
the extensions.conf file ?
Thx for the help
Matt
P.S: I've already looked on google, but could not find any help..
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[Asterisk-Users] Softphone..easy to use ?
Hello Guys Im currently using Asterisk on a Red Hat box with an ISDN Card on it.. Works perfect. Now I like to forward a call to a softphone. (from my asterisk menu) Im very new to this, so unsure what softphone I should use ? Can anybody provide me a link with a good Softphone ? (for windows) How is the quality on software? Do you head any different between softphone and regular phone if a person calls you ? And is it hard to forward a call from asterisk to a softphone ? Thank you for the help Regards from Switzerland Matt ___ 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 forward a call to the same ISDN box ?
Hello Guys Im trying to forward a incoming call from asterisk to a second number (the second phone number is located on the same ISDN BOX ) I did try the following on the extensions.conf exten=2,1,Dial(capi/720XXX1:720XXX2,18) It does work if the second number is a phone number located outside of the building (lets say a mobile phone number).. But I don't know why I cant forward the call if the phone number is on the same ISDN box. Example: Incoming call -- 720XXX1 (Asterisk server pick's up the call) --- (forwarding to) 720XXX2.. Did anybody ever got to manage this problem ? Thank you regards from Switzeeland :) Matt ___ 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 make a call with asterisk from shell ? (or with a .sh file )
Hello Guys Does anybody know how to make a call with asterisk with a shell comment ? I like to connect that comment with a small .sh script.. Is it actually possible to make a call with asterisk true shell or a .sh file ? Thank you for the help and regards from Switzerland Matt ___ 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] Error when starting Asterisk (Loading module chan_capi.so failed!)
Hey Guys Why I try to start Asterisk on my debain box with asterisk c Ill get the following error: [res_musiconhold.so] = (Music On Hold Resource) == Parsing '/etc/asterisk/musiconhold.conf': Found == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' [res_parking.so] = (Call Parking Resource) == Parsing '/etc/asterisk/parking.conf': Found -- Registered extension context 'parkedcalls' -- Added extension '701' priority 1 to parkedcalls -- Added extension '702' priority 1 to parkedcalls -- Added extension '703' priority 1 to parkedcalls -- Added extension '704' priority 1 to parkedcalls -- Added extension '705' priority 1 to parkedcalls -- Added extension '706' priority 1 to parkedcalls -- Added extension '707' priority 1 to parkedcalls -- Added extension '708' priority 1 to parkedcalls -- Added extension '709' priority 1 to parkedcalls -- Added extension '710' priority 1 to parkedcalls -- Added extension '711' priority 1 to parkedcalls -- Added extension '712' priority 1 to parkedcalls -- Added extension '713' priority 1 to parkedcalls -- Added extension '714' priority 1 to parkedcalls -- Added extension '715' priority 1 to parkedcalls -- Added extension '716' priority 1 to parkedcalls -- Added extension '717' priority 1 to parkedcalls -- Added extension '718' priority 1 to parkedcalls -- Added extension '719' priority 1 to parkedcalls -- Added extension '720' priority 1 to parkedcalls == Registered application 'ParkedCall' == Manager registered action ParkedCalls [chan_capi.so] = (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found Oct 27 14:29:21 NOTICE[1024]: chan_capi.c:2626 load_module: CAPI not installed! Oct 27 14:29:21 WARNING[1024]: loader.c:313 ast_load_resource: chan_capi.so: load_module failed, returning -1 Oct 27 14:29:21 WARNING[1024]: chan_capi.c:2801 unload_module: Unable to unregister from CAPI! == Unregistered channel type 'CAPI' Oct 27 14:29:21 WARNING[1024]: loader.c:359 load_modules: Loading module chan_capi.so failed! I do understand that theres a problem with chan_capi.so. But I can locate chan_capi.so without any problems.. asterisk:/etc/asterisk# locate chan_capi.so /usr/lib/asterisk/modules/chan_capi.so Sorry for asking all this.. Thank u very much for the help Matt P.S: The wirred thing is, it seems to me that chan_capi.so is loaded file (more /var/log/messages | grep capi) Oct 27 00:35:35 asterisk kernel: kcapi: driver fcpci attached Oct 27 00:35:35 asterisk kernel: kcapi: driver fcpci detached Oct 27 00:55:47 asterisk kernel: kcapi: capidrv attached Oct 27 00:55:47 asterisk kernel: kcapi: appl 1 up Oct 27 00:55:47 asterisk kernel: capidrv: Rev 1.1.4.1: loaded Oct 27 00:55:47 asterisk kernel: capifs: Rev 1.1.4.1 Oct 27 00:55:48 asterisk kernel: capi20: started up with major 68 Oct 27 00:55:48 asterisk kernel: kcapi: capi20 attached Oct 27 00:55:48 asterisk kernel: capi20: Rev 1.1.4.2: started up with major 68 (middleware+capifs) Oct 27 00:55:48 asterisk kernel: kcapi: driver fcpci attached Oct 27 00:55:48 asterisk kernel: [isdn:__insmod_isdn_S.bss_L2336+1282055/220495493] [isdn:__insmod_isdn_S.bss_L2336+1282036/220495512] [isdn:__insmod_isdn_S.bss_L2336+1314040/220463508] [capifs:__insmod_capifs_O/lib/modules/2.4.21-i586-cdv/kernel/driver+-958112/96] [isdn:__insmod_isdn_S.bss_L2336+1313920/220463628] [capifs:__insmod_capifs_O/lib/modules/2.4.21-i586-cdv/kernel/driver+-958142/96] Oct 27 03:18:57 asterisk kernel: kcapi: capi20 detached Oct 27 03:18:57 asterisk kernel: capi: Rev 1.1.4.2: unloaded Oct 27 03:18:58 asterisk kernel: kcapi: appl 1 down Oct 27 03:18:58 asterisk kernel: kcapi: capidrv detached Oct 27 03:18:58 asterisk kernel: capidrv: Rev 1.1.4.1 : unloaded Oct 27 13:42:59 asterisk kernel: kcapi: capidrv attached Oct 27 13:42:59 asterisk kernel: kcapi: appl 1 up Oct 27 13:42:59 asterisk kernel: capidrv: Rev 1.1.4.1: loaded Oct 27 13:42:59 asterisk kernel: capifs: Rev 1.1.4.1 Oct 27 13:42:59 asterisk kernel: capi20: started up with major 68 Oct 27 13:42:59 asterisk kernel: kcapi: capi20 attached Oct 27 13:42:59 asterisk kernel: capi20: Rev 1.1.4.2: started up with major 68 (middleware+capifs) Oct 27 13:42:59 asterisk kernel: kcapi: driver fcpci attached Oct 27 13:42:59 asterisk kernel: [isdn:__insmod_isdn_S.bss_L2336+1265671/220502045] [isdn:__insmod_isdn_S.bss_L2336+1265652/220502064] [isdn:__insmod_isdn_S.bss_L2336+1297656/220470060] [capi:__insmod_capi_O/lib/modules/2.4.21-i586-cdv/kernel/drivers/+-949920/96] [isdn:__insmod_isdn_S.bss_L2336+1297536/220470180] [capi:__insmod_capi_O/lib/modules/2.4.21-i586-cdv/kernel/drivers/+-949950/96] asterisk:/etc/asterisk# ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update
[Asterisk-Users] installing install isdn4k-utils from source ?
Hey guys.. Im trying to install isdn4k-utils from source.. Can anybody help me ? What I did is: untar the file.. cd isdn4k-utils-CVS-2004-10-07 make config Select from the submenu Card configuration tools I only selected [*] avmcapictrl / capiinit ( not sure if thats correct, I have a AVM ISDN Fitz Card ) then save end exit.. and then Ill get the following error: checking for tcl.h... no checking for tcl header in /usr/include/tcl8.3/tcl.h... no configure: warning: ** configure: warning: ** Unable to find a installed tcl package! configure: warning: ** configure: error: stop make[2]: Entering directory `/root/isdn4k-utils-CVS-2004-10-07' WARNING! Configure in vbox failed, disabling package make[2]: Leaving directory `/root/isdn4k-utils-CVS-2004-10-07' Running configure in doc ... loading cache ./config.cache checking for a BSD compatible install... (cached) /usr/bin/install -c checking for mawk... (cached) gawk checking for sed... (cached) /bin/sed creating ./config.status creating Makefile creating ttyI.man creating isdninfo.man creating isdn_cause.man creating isdn_audio.man creating isdnctrl.man Running configure in FAQ ... loading cache ./config.cache checking for a BSD compatible install... /usr/bin/install -c checking for tar... /bin/tar checking for zip... /usr/bin/zip checking for gzip... /bin/gzip checking for sgml2html... ./configure: line 707: test: too many arguments sgml2html-not-found-or-installed checking for sgml2txt... ./configure: line 738: test: too many arguments sgml2txt-not-found-or-installed configure: error: sgml2html not found please install it. make[2]: Entering directory `/root/isdn4k-utils-CVS-2004-10-07' WARNING! Configure in FAQ failed, disabling package make[2]: Leaving directory `/root/isdn4k-utils-CVS-2004-10-07' make[1]: Leaving directory `/root/isdn4k-utils-CVS-2004-10-07' Anybody knows whats wrong ? Thank you soo much Matt ___ 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
RE: [Asterisk-Users] CAPI and Asterisk (with AVM ISDN Card)
Thank you for your help ;-) I have downloadet from the url you gave me the following file. isdn4k-utils-CVS-2004-10-07.tar.bz2 When I cd into that folder.. and type make conf I get a wirred screen where I can choose from the following: Code maturity level options --- General configuration --- Runtime configuration tools --- Card configuration tools --- Tools for monitoring activity --- Applications --- Documentation --- --- Load an Alternate Configuration File Save Configuration to an Alternate File Any idears ? I spend so hours on this now.. Do you know anyone ( or do you) offer a service where I can pay someone to login and help me install it ? ( capi + my card ) Thank you Grsse / Best Regards Mateo Meier --- Don't marry for money; you can borrow it cheaper ;-) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maurizio Marini Sent: Mittwoch, 20. Oktober 2004 09:04 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CAPI and Asterisk (with AVM ISDN Card) On Wednesday 20 October 2004 00:30, Mateo Meier wrote: Does anybody knows what version of capi is needed ? try the most recent here: ftp://ftp.isdn4linux.de/pub/isdn4linux/CVS-Snapshots it did work fine for me (FC2 and debian sid) I tried to install a capi rpm.. but after the capi rpm installation, there seems to be no /etc/capi.conf cat capi.conf # card fileproto io irq mem cardnr options b1isa b1.t4 DSS10x150 7 - - P2P b1pci b1.t4 DSS1- - - - c4 /usr/sbin/c4.binDSS1- - - - c4 - DSS1- - - - c4 - DSS1- - - - P2MP c4 - DSS1- - - - P2MP c2 c2.bin DSS1- - - - c2 - DSS1- - - - t1isa t1.t4 DSS10x340 9 - 0 t1pci t1.t4 DSS1- - - - fcpci - - - - - - fcclassic - - 0x150 10 - - What kind of capi version do I need ? capi4k-utils ? or just any capi rpm ? download a tarball and install it... Maurizio ___ 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 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] CAPI and Asterisk (with AVM ISDN Card)
Hello Guys Im trying to get Asterisk with my AVM fritz Card (ISDN) to work. ( fedora core 1 ) I did found a easy how to.. it was posted from someone here on this Mailing List Im referring to http://lists.digium.com/pipermail/asterisk-users/2004-June/052118.html MY PROBLEM: I cant get CAPI to work ;-) The how to (http://lists.digium.com/pipermail/asterisk-users/2004-June/052118.html) is assuming you have already installed capi and are able to edit the the /etc/capi.conf File. Does anybody knows what version of capi is needed ? I tried to install a capi rpm.. but after the capi rpm installation, there seems to be no /etc/capi.conf What kind of capi version do I need ? capi4k-utils ? or just any capi rpm ? Thank you so much for your feedback Regards from Switzerland Matt ___ 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
