[asterisk-users] use more then one sip-provider to dial out
Hi I want to be able to use one provider if I dial 0 before the number and another if I dial 1 before, how can I do that in asterisk 1.6? /ralf Ralf Träskman, IT AdLibris AB, Box 3667, 103 59 Stockholm. Besöksadress: Sveavägen 56C, 111 34, Stockholm - Obs ny address! Dir: +46-(0)8-5460 60 91, mob: +46-(0)70-7548074, vxl: +46-(0)8-5460 60 00, fax: +46-(0)8-5460 60 99 r...@adlibris.commailto:and...@adlibris.com, www.adlibris.comhttp://www.adlibris.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem with nortel 2002 disconecting
We have 40 nortel ip2002 phones connected to asterisk 1.6, the problem I have is that the phone looses the connections with the server and then drops calls, we can reconnect but the customers don't like it. Anyone has the same problem? /ralf Ralf Träskman, IT AdLibris AB, Sveavägen 56C, 111 34 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.commailto:r...@adlibris.com www.adlibris.comhttp://www.adlibris.com/ P Please consider the environment before printing this e-mail g ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip phone cant hear the caller
Hi Im using a sip phone SPA921, and the one that calls me can hear me but I cant hear them, when I make the call I can hear them. Im running asterisk 1.6 behind a firewall, I have port 1-2 for rtp and 5060 for sip forward to my asterisk. Any tips? Regards /ralf Ralf Träskman, IT AdLibris AB, Sveavägen 56C, 111 34 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.commailto:r...@adlibris.com www.adlibris.comhttp://www.adlibris.com/ P Please consider the environment before printing this e-mail ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] unistim channel problem
Hi [Feb 17 07:59:45] WARNING[21539]: channel.c:3477 ast_request: No channel type registered for 'USTM' [Feb 17 07:59:45] WARNING[21539]: app_dial.c:1502 dial_exec_full: Unable to create channel of type 'USTM' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) I get this after I restart my asterisk 1.6, it all worked yesterday. I have the unistim module loaded. Could it be that I have set keepaliave in unistim.conf to 500, I had to do that outerwise my phones would show server unreachable after approx 2 minutes. What can I do? /ralf Ralf Träskman, IT AdLibris AB, Sveavägen 56C, 111 34 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.commailto:r...@adlibris.com www.adlibris.comhttp://www.adlibris.com/ P Please consider the environment before printing this e-mail ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] unistim and transfer calls
I have added t in dialplan exten = 1234,1,Dial(USTM/2...@c,40,t) so now i can transfer, but when the caller the extension I transfer to hangs up asterisk dumps an I have to start it up again. Any thoughts? /ralf From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ralf Träskman Sent: den 10 februari 2009 16:00 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] unistim and transfer calls Hi When i try to transfer calls from my ip2002 phone in asterisk 1.6, I can make the transfer and it rings on the extension I transfer to, but when I accept the call, asterisk dumps. How can I get it to work? And how do I save the dump error? Regards /ralf Ralf Träskman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.commailto:r...@adlibris.com www.adlibris.comhttp://www.adlibris.com/ P Please consider the environment before printing this e-mail ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] unistim and transfer calls
Hi When i try to transfer calls from my ip2002 phone in asterisk 1.6, I can make the transfer and it rings on the extension I transfer to, but when I accept the call, asterisk dumps. How can I get it to work? And how do I save the dump error? Regards /ralf Ralf Träskman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.commailto:r...@adlibris.com www.adlibris.comhttp://www.adlibris.com/ P Please consider the environment before printing this e-mail ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] no need to dial areacode
Hi To dial an outside line i have to dial 0. I want to have that when we dial local numbers, that is we are in the 08 area, I don't want to have to dial 08, how to set this up in asterisk 1.6? Regards /ralf Ralf Träskman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.commailto:r...@adlibris.com www.adlibris.comhttp://www.adlibris.com/ P Please consider the environment before printing this e-mail ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] no need to dial areacode
Hi
Yes i have tried to get them to dial the whole number to, but no luck. Ill try
your suggestions.
/ralf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gordon Henderson
Sent: den 5 februari 2009 14:50
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] no need to dial areacode
On Thu, 5 Feb 2009, Ralf Träskman wrote:
Hi
To dial an outside line i have to dial 0. I want to have that when we
dial local numbers, that is we are in the 08 area, I don't want to
have to dial 08, how to set this up in asterisk 1.6?
Are your local numbers a fixed length? If so, this might work:
; Local numbers - 8 digits long:
exten _,1,Noop(Local number)
exten _,n,Dial(${out}/08${EXTEN})
etc.
If you have other numbers of varying length, then this might not work...
This generally works OK in the UK for local area dialling. My local numbers are
5 or 6 digits long, so ...
exten = _XX,1,Macro(dialOut,01364${EXTEN})
exten = _X,1,Macro(dialOut,01364${EXTEN})
Not perfect, but it works OK.
I do try to persuade my customers to always dial the full number though, 10 or
11 digits starting with 0, because that's what they need to do on a mobile...
Gordon
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[asterisk-users] musiconhold realtime queue
Hi I have asterisk 1.6 and running queues with realtime mysql. I am trying to set another musiconhold then default but I cant get it to work, I have an musiconhold entry in my queue_table, but don't know what to put in there and where to put the file. Regards /ralf Ralf Träskman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.commailto:r...@adlibris.com www.adlibris.comhttp://www.adlibris.com/ P Please consider the environment before printing this e-mail ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] musiconhold realtime queue
Hmm i hope i do it in realtime, how can I tell? /ralf From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David fire Sent: den 5 februari 2009 15:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] musiconhold realtime queue do you condifure the new musiconhold in the music on hold config file (or in realtime) ? David 2009/2/5 Ralf Träskman r...@adlibris.commailto:r...@adlibris.com Hi I have asterisk 1.6 and running queues with realtime mysql. I am trying to set another musiconhold then default but I cant get it to work, I have an musiconhold entry in my queue_table, but don't know what to put in there and where to put the file. Regards /ralf Ralf Träskman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.commailto:r...@adlibris.com www.adlibris.comhttp://www.adlibris.com/ P Please consider the environment before printing this e-mail ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] musiconhold realtime queue
Thanks I got it working now /ralf From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David fire Sent: den 5 februari 2009 15:55 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] musiconhold realtime queue hi to add a new music on hold you need to add it to musiconhold.conf or in the realtime table. see the file you will know how to add a new music on hold. and then you can make it realtime. David 2009/2/5 Ralf Träskman r...@adlibris.commailto:r...@adlibris.com Hmm i hope i do it in realtime, how can I tell? /ralf From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David fire Sent: den 5 februari 2009 15:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] musiconhold realtime queue do you condifure the new musiconhold in the music on hold config file (or in realtime) ? David 2009/2/5 Ralf Träskman r...@adlibris.commailto:r...@adlibris.com Hi I have asterisk 1.6 and running queues with realtime mysql. I am trying to set another musiconhold then default but I cant get it to work, I have an musiconhold entry in my queue_table, but don't know what to put in there and where to put the file. Regards /ralf Ralf Träskman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.commailto:r...@adlibris.com www.adlibris.comhttp://www.adlibris.com/ P Please consider the environment before printing this e-mail ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Start asterisk on boot
Hi We runs asterisk 1.6 on a ubuntu 8.04 server. How can I get asterisk to start at boot? I have created an file named asterisk in /etc/event.d and put in this # This service maintains Asterisk from the point the system is # started until it is shut down again. description Asterisk daemon start on runlevel-2 stop on shutdown respawn exec //usr/sbin/asterisk -f But it doesn't work. Regards /ralf Ralf Träskman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.commailto:r...@adlibris.com www.adlibris.comhttp://www.adlibris.com/ P Please consider the environment before printing this e-mail ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Start asterisk on boot
Hi That didnt work either, do i have to set some permissions? /ralf From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: den 26 januari 2009 09:52 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Start asterisk on boot 2009/1/26 Ralf Träskman r...@adlibris.commailto:r...@adlibris.com Hi We runs asterisk 1.6 on a ubuntu 8.04 server. How can I get asterisk to start at boot? I have created an file named asterisk in /etc/event.d and put in this # This service maintains Asterisk from the point the system is # started until it is shut down again. description Asterisk daemon start on runlevel-2 stop on shutdown respawn exec //usr/sbin/asterisk -f But it doesn't work. Regards /ralf 1. Copy relevant file from contrib directory into /etc/init.d directory (while renaming it asterisk) 2. Then sudo update-rc.d asterisk defaults and it's done Ralf Träskman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.commailto:r...@adlibris.com www.adlibris.comhttp://www.adlibris.com/ P Please consider the environment before printing this e-mail ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Forwarding calls and trasfer calls
Hi
Where do i put this, and what shall i change do make it work for me?
/ralf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Klaus Darilion
Sent: den 20 januari 2009 18:17
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Forwarding calls and trasfer calls
features.conf for transfers
for call forwardin you need some application logic.
e.g.
_**21**. = {
Set(NUM=${EXTEN:6}); // contains the new target
// now store this number somewhere, e.g. astdb, odbc ...
...
}
context fromPstn {
1234 = {
// check if user has actived forwarding
// retrieve NUM from astdb or ODBC
if(${EXISTS(${NUM})}) {
Dial(DAHDI/g1/${NUM});
} else {
Dial(SIP/${EXTEN});
}
}
}
regards
klaus
Ralf Träskman schrieb:
Hi
How do i set up so that everyone can dial, for example **21** to forward
all calls to a cellphone or another extension and how do I enable so
that cals can be transferd between extentions.
I use asterisk 1.6 and have my phones in unistim.conf and my extensions
in extensions.conf.
Regards
/ralf
Ralf Träskman, IT
AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden
Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99
r...@adlibris.com mailto:r...@adlibris.com www.adlibris.com
http://www.adlibris.com/
P *Please consider the environment before printing this e-mail*
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[asterisk-users] Forwarding calls and trasfer calls
Hi How do i set up so that everyone can dial, for example *21* to forward all calls to a cellphone or another extension and how do I enable so that cals can be transferd between extentions. I use asterisk 1.6 and have my phones in unistim.conf and my extensions in extensions.conf. Regards /ralf Ralf Träskman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.commailto:r...@adlibris.com www.adlibris.comhttp://www.adlibris.com/ P Please consider the environment before printing this e-mail ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] adding numbers in dialplan
Hi When we ned to call 112 (emergency number) we need to add 0379 before 112 and 464 after for it to work, how do I do that In my dialplan? The caller should only dial 112 on the phone. Regards /ralf Ralf Träskman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.commailto:r...@adlibris.com www.adlibris.comhttp://www.adlibris.com/ P Please consider the environment before printing this e-mail ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] adding numbers in dialplan
Hi
Thanks
/ralf
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Ortiz
Sent: den 19 januari 2009 14:27
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] adding numbers in dialplan
sorry try with:
exten = 112,1,Dial(SIP/Provider/0379${EXTEN}464)
2009/1/19 Daniel Ortiz zate...@gmail.commailto:zate...@gmail.com
exten = 112,1,Dial(SIP/Provider/0379464${EXTEN})
bye
2009/1/19 Ralf Träskman r...@adlibris.commailto:r...@adlibris.com
Hi
When we ned to call 112 (emergency number) we need to add 0379 before 112 and
464 after for it to work, how do I do that In my dialplan?
The caller should only dial 112 on the phone.
Regards
/ralf
Ralf Träskman, IT
AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden
Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99
r...@adlibris.commailto:r...@adlibris.com
www.adlibris.comhttp://www.adlibris.com/
P Please consider the environment before printing this e-mail
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[asterisk-users] gxp2000 and no sound asterisk 1.6
Hi I have a grandstream gxp-2000 and trying it on an asterisk 1.6. When I call internally between extensions I can hear the other person in the gxp2000, but when I call externally from the gxp I can't hear the person on the other end, but he can hear me. How do you configure the grandstream 2000 to work on asterisk 1.6? Regards /ralf Ralf Träskman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.commailto:r...@adlibris.com www.adlibris.comhttp://www.adlibris.com/ P Please consider the environment before printing this e-mail ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] gxp2000 and no sound asterisk 1.6
Hi Yes we use voip as external. /ralf -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gordon Henderson Sent: den 14 januari 2009 10:46 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] gxp2000 and no sound asterisk 1.6 On Wed, 14 Jan 2009, Ralf Träskman wrote: Hi I have a grandstream gxp-2000 and trying it on an asterisk 1.6. When I call internally between extensions I can hear the other person in the gxp2000, but when I call externally from the gxp I can't hear the person on the other end, but he can hear me. How do you configure the grandstream 2000 to work on asterisk 1.6? First, upgrade your asterisk to 1.2 ... ;-) What is the external connection? Is it VoIP, PSTN, or ... ? If it's VoIP then it's almost certian to be a NAT problem with your network/router. There's no magic in setting up GXP2000's - they're fairly straightforward, and if you can do phone to phone, (via an asterisk) they're probably OK. Let us know more about the external connection technology... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 404 not found from one ip-adress
Hi Our sip provider has two servers that sends calls to our asterisk 1.6. When server 1 sends call everything is working, but when server 2 sends call I get [Jan 13 14:56:23] NOTICE[16680]: chan_sip.c:16869 handle_request_invite: Call from '' to extension '0840303390' rejected because extension not found. And the provider get an 404 not found error on their side. What can be the problem? Regards /ralf Ralf Träskman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.commailto:r...@adlibris.com www.adlibris.comhttp://www.adlibris.com/ P Please consider the environment before printing this e-mail ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 404 not found from one ip-adress
Hi The provider dont use register, they are running openSER I have this in my sip.conf [outgoing] context=ip-only disallow=all allow=alaw,ulaw canreinvite=yes dtmfmode=rfc2833 host=sip.hub.ip-only.se insecure=very reinvite=yes type=friend [incoming] disallow=all allow=alaw,ulaw context=ip-only type=user Regards /ralf From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovid Bender Sent: den 13 januari 2009 15:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 404 not found from one ip-adress - Original Message - From: Ralf Träskmanmailto:r...@adlibris.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion'mailto:asterisk-users@lists.digium.com Sent: Tuesday, January 13, 2009 4:04 PM Subject: [asterisk-users] 404 not found from one ip-adress Hi Our sip provider has two servers that sends calls to our asterisk 1.6. When server 1 sends call everything is working, but when server 2 sends call I get [Jan 13 14:56:23] NOTICE[16680]: chan_sip.c:16869 handle_request_invite: Call from '' to extension '0840303390' rejected because extension not found. And the provider get an 404 not found error on their side. What can be the problem? Regards /ralf Ralf Träskman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.commailto:r...@adlibris.com www.adlibris.comhttp://www.adlibris.com/ P Please consider the environment before printing this e-mail Raif, What does your sip register statement look like ? It seems that they are sending it to yournum...@youripmailto:yournum...@yourip and you do not have it set up in the context for this carrier. You can call them and ask them to fix it or just add in Exten = 084303390 in the context and then just have a goto to the extension that the first server is sending calls to (maybe the s extension ?). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 404 not found from one ip-adress
Hi Its the same provider and i use dns name in sip.conf [outgoing] context=ip-only disallow=all allow=alaw,ulaw canreinvite=yes dtmfmode=rfc2833 host=sip.hub.ip-only.se insecure=very reinvite=yes type=friend [incoming] disallow=all allow=alaw,ulaw context=ip-only type=user Regards /ralf From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: den 13 januari 2009 15:39 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] 404 not found from one ip-adress Provider 2 is dropping into a new context than Provider 1. The $EXTEN is probably coming in from P1 as XX and P2 as AXX. Check your incoming and default sections of extensions.conf. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ralf Träskman Sent: Tuesday, January 13, 2009 8:04 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] 404 not found from one ip-adress Hi Our sip provider has two servers that sends calls to our asterisk 1.6. When server 1 sends call everything is working, but when server 2 sends call I get [Jan 13 14:56:23] NOTICE[16680]: chan_sip.c:16869 handle_request_invite: Call from '' to extension '0840303390' rejected because extension not found. And the provider get an 404 not found error on their side. What can be the problem? Regards /ralf Ralf Träskman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.commailto:r...@adlibris.com www.adlibris.comhttp://www.adlibris.com/ P Please consider the environment before printing this e-mail ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 404 not found from one ip-adress
Thanks Your tip got my on the right track Regards /ralf -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kristian Kielhofner Sent: den 13 januari 2009 16:32 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 404 not found from one ip-adress On Tue, Jan 13, 2009 at 9:59 AM, Ralf Träskman r...@adlibris.com wrote: Hi The provider dont use register, they are running openSER I have this in my sip.conf [outgoing] context=ip-only disallow=all allow=alaw,ulaw canreinvite=yes dtmfmode=rfc2833 host=sip.hub.ip-only.se insecure=very reinvite=yes type=friend [incoming] disallow=all allow=alaw,ulaw context=ip-only type=user Regards /ralf Ralf, That incoming peer isn't matching anything. They're probably hitting the context defined in [general]. Add another peer/friend match with the other servers IP/hostname and the ip-only context. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem incomming from openser
Hi I have an asterisk 1.6 running, and our provider have an openser on their end. When I get an incoming call I get this on my end [Jan 8 14:51:56] NOTICE[16680]: chan_sip.c:16869 handle_request_invite: Call from '' to extension '0840303395' rejected because extension not found. If I wait approx a minute and try again, the call will go trough. We don't use REGISTER or anything like that. What can be the problem regards /ralf Ralf Träskman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.commailto:r...@adlibris.com www.adlibris.comhttp://www.adlibris.com/ P Please consider the environment before printing this e-mail ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem incomming from openser
Here it is, the part I want to use is the things under [ip-only] /ralf From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David fire Sent: den 8 januari 2009 15:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem incomming from openser hi can you post your extension.conf? thanks David 2009/1/8 Ralf Träskman r...@adlibris.commailto:r...@adlibris.com Hi I have an asterisk 1.6 running, and our provider have an openser on their end. When I get an incoming call I get this on my end [Jan 8 14:51:56] NOTICE[16680]: chan_sip.c:16869 handle_request_invite: Call from '' to extension '0840303395' rejected because extension not found. If I wait approx a minute and try again, the call will go trough. We don't use REGISTER or anything like that. What can be the problem regards /ralf Ralf Träskman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.commailto:r...@adlibris.com www.adlibris.comhttp://www.adlibris.com/ P Please consider the environment before printing this e-mail ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. extensions.conf Description: extensions.conf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem incomming from openser
Hi This is my sip.conf Ralf From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mailinglists Sent: den 8 januari 2009 16:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem incomming from openser Any context you have specified in sip.conf? There the extension is searched for. And if that's not default, it might not find it. br Walter From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ralf Träskman Sent: Thursday, January 08, 2009 3:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Problem incomming from openser Here it is, the part I want to use is the things under [ip-only] /ralf From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David fire Sent: den 8 januari 2009 15:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem incomming from openser hi can you post your extension.conf? thanks David 2009/1/8 Ralf Träskman r...@adlibris.commailto:r...@adlibris.com Hi I have an asterisk 1.6 running, and our provider have an openser on their end. When I get an incoming call I get this on my end [Jan 8 14:51:56] NOTICE[16680]: chan_sip.c:16869 handle_request_invite: Call from '' to extension '0840303395' rejected because extension not found. If I wait approx a minute and try again, the call will go trough. We don't use REGISTER or anything like that. What can be the problem regards /ralf Ralf Träskman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.commailto:r...@adlibris.com www.adlibris.comhttp://www.adlibris.com/ P Please consider the environment before printing this e-mail ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. sip.conf Description: sip.conf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem incomming from openser
Hi I dont understand how to do that, I put in this line to ip-only in extention.conf exten=s,1,Dial No differens Thanks for all help /r From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David fire Sent: den 8 januari 2009 16:46 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem incomming from openser hi try this... add the s extencion to ip-only whit just one line verbose (S ext called) the s extencion is like default extencion just to see what happen... maybe the operator isnt sending all the info to you. Davidf 2009/1/8 Ralf Träskman r...@adlibris.commailto:r...@adlibris.com Hi This is my sip.conf Ralf From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mailinglists Sent: den 8 januari 2009 16:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem incomming from openser Any context you have specified in sip.conf? There the extension is searched for. And if that's not default, it might not find it. br Walter From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ralf Träskman Sent: Thursday, January 08, 2009 3:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Problem incomming from openser Here it is, the part I want to use is the things under [ip-only] /ralf From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David fire Sent: den 8 januari 2009 15:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem incomming from openser hi can you post your extension.conf? thanks David 2009/1/8 Ralf Träskman r...@adlibris.commailto:r...@adlibris.com Hi I have an asterisk 1.6 running, and our provider have an openser on their end. When I get an incoming call I get this on my end [Jan 8 14:51:56] NOTICE[16680]: chan_sip.c:16869 handle_request_invite: Call from '' to extension '0840303395' rejected because extension not found. If I wait approx a minute and try again, the call will go trough. We don't use REGISTER or anything like that. What can be the problem regards /ralf Ralf Träskman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.commailto:r...@adlibris.com www.adlibris.comhttp://www.adlibris.com/ P Please consider the environment before printing this e-mail ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] set monitor_filename
Hi
Hm is this function for recording? The thing I want to do is to be able to see
how many calls there is waiting in a queue, maybe im looking in the wrong
direction?
/ralf
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro
Kauffmann
Sent: den 5 december 2008 05:03
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] set monitor_filename
Ralf Träskman wrote:
Hi
I have this in my queue extension and I see this in asterisk when I call
to the queue, but no file is created in the directory any ideas?
exten =
s,1,Set(MONITOR_FILENAME=/var/spool/asterisk/queuecalls/QSAMPLE-${UNIQUEID})
-- Executing [EMAIL PROTECTED]:1] Set(SIP/0850001175-b7942770,
MONITOR_FILENAME=/var/spool/asterisk/queuecalls/QSAMPLE-1228382046.12)
Regards
/ralf
The basics. Does the queuecalls directory exist? Does the user that *
runs under have write permission in that directory?
The monitor-format parameter MUST be set in queues.conf to enable
recording and to select the format of the recording. In addition, I
believe this is still true in 1.4, if you don't set the monitor-join
parameter to yes you will end up with two files (in out) instead of a
single file with both legs of the call.
Alex
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[asterisk-users] set monitor_filename
Hi
I have this in my queue extension and I see this in asterisk when I call to the
queue, but no file is created in the directory any ideas?
exten =
s,1,Set(MONITOR_FILENAME=/var/spool/asterisk/queuecalls/QSAMPLE-${UNIQUEID})
-- Executing [EMAIL PROTECTED]:1] Set(SIP/0850001175-b7942770,
MONITOR_FILENAME=/var/spool/asterisk/queuecalls/QSAMPLE-1228382046.12)
Regards
/ralf
Ralf Träskman, IT
AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden
Dir: +46-(0)707458074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99
[EMAIL PROTECTED]mailto:[EMAIL PROTECTED]
www.adlibris.comhttp://www.adlibris.com/
P Please consider the environment before printing this e-mail
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[asterisk-users] Asterisk user client for customer service
Hi Is there a user client that a group, like customer service can use? We have today an avaya IP-office with phonemanager pro and I want something equal to phonemanager pro, where you can logon to ques and see how many calls is in that queue and so on. Regards /ralf Ralf Träskman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707458074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 [EMAIL PROTECTED]mailto:[EMAIL PROTECTED] www.adlibris.comhttp://www.adlibris.com/ P Please consider the environment before printing this e-mail ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.0-beta5 voicemail problem
Hi I have set up voicemail, when i call an ext and voicemail kicks in i can leave a message. The problem is that the message is in the tmp directory of the extensions voicemail and when i call to check if there is any new vm there is no messages at all. Even if i move the message to inbox or work or old, the sstem dont detect the messages. Any clues to what i have done wrong? Regards AdLibris /Ralf P Please consider the environment before printing this e-mail ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] realtime queue asterisk 1.6.0-beta5
Hi I have enabled realtime queue in asterisk, but when i enter a queue i get this and then asterisk crashes. Any clues? -- Executing [EMAIL PROTECTED]:1] Answer(SIP/Ralf-08207de0, ) in new stack -- Executing [EMAIL PROTECTED]:2] Ringing(SIP/Ralf-08207de0, ) in new stack -- Executing [EMAIL PROTECTED]:3] Wait(SIP/Ralf-08207de0, 2) in new stack -- Executing [EMAIL PROTECTED]:4] Queue(SIP/Ralf-08207de0, Kundservice) in new stack Segmentation fault Regards /Ralf Med vänliga hälsningar! AdLibris /Ralf Ralf Träskman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)706452438, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 [EMAIL PROTECTED]mailto:[EMAIL PROTECTED] www.adlibris.comhttp://www.adlibris.com/ P Please consider the environment before printing this e-mail ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
