RE: [Asterisk-Users] CISCO ATA 188
-Original Message- Anyone here with experience on the Cisco ATA 188 and *? Is it as good as ATA 186? AFAIK the only difference was the built-in switch so you can plug your PC in the back, right ? Should be 'as good'. Florian ___ 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] DTMF Issues with SJPHONE
What ver of SJPHONE? Thanks for the voicemail stuff :-) - Original Message - From: Girish Gopinath [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, February 28, 2004 7:48 PM Subject: RE: [Asterisk-Users] DTMF Issues with SJPHONE Has anyone had a similar issue with Asterisk Voicemail being unable to detect the digits sent from an SJ Phone connection. I have included dtmfmode=inband and it works fine when calling other phones though not with Voicemail. Voicemail doesn't regonise the password. I am using SJPhone, and works fine for me. Is there a way to not send a password when logging into Voicemail as a temp measure. Try something like like this, it will not ask for your password: exten = your extension,1,Ringing exten = your extension,2,Wait(2) exten = your extension,3,VoicemailMain,s ; is the mail box number Also, check out this url: http://www.automated.it/guidetoasterisk.htm Regards, Girish _ Post Classifieds on MSN classifieds. http://www.sulekha.com/msnclassifieds Buy and Sell on MSN Classifieds. ___ 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] Iconnect behind NAT
Anyone got an example of sip and extensions confs for Iconnect outgoing calls behind NAT.
Re: [Asterisk-Users] DTMF Issues with SJPHONE
What ver of SJPHONE? SJPhone Evaluation Version, release Jul 31, 2003, Build: 1.10.187c Girish _ All the news that matters. All the gossip from home. http://www.msn.co.in/NRI/ Specially for NRIs! ___ 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] retrieve_sip_conf_from_mysql.pl data format
On Fri, 2004-02-27 at 21:39, Chad Sawyer wrote: In the contrib/scripts directory I have been trying to figure out the format of the entries in the MySQL table. -CUT- There are 3 different approaches to storing users in a database. The first is dynamic - the user details are read directly from the database. This is used for SIP IAX friends also for Voicemail: http://voip-info.org/tiki-index.php?page=Asterisk+sip+mysql+peers http://voip-info.org/tiki-index.php?page=Asterisk+voicemail+database However the number of options supported by this 'MySQL friends' system is currently very limited. The other possibility is to store all the details in the database when changes are made, write out new versions of the conf files. This is the approach taken by res_config: http://voip-info.org/wiki-Asterisk+res_config also by the contrib scripts, such as: retrieve_sip_conf_from_mysql.pl Obviously, the disadvantage of such systems is that Asterisk needs to be reloaded to see these changes, which can be disruptive to calls in progress, or may never get the chance to happen 'when convenient' on a busy system. Olle's chan_sip2 introduces a 3rd possibility: Using templates autocreate peers for the majority of user options storing just the passwords in the MYSQL database. For me, the ideal would be to hack the code to extend the functionality of MySQL friends...however I'm not a C programmer. I am currently starting work on the 2nd option since I want to expose as many options within a web-based GUI as possible. Any suggestions on how to minimise impact on the running system during reload are welcome :) Best Wishes, Fran. ___ 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] RE: simple H323 question
Dear All,
Thanks, but I am not using chan_oh323, I am using chan_h323. The major
reason why I am not using chan_oh323 is because of a bug that Michael is not
yet able to resolve. Every call that goes out via this h323 channel will be
considered connected and picked up (false answer supervision) immediately
after the call setup, even though the call is busy or ring no answer. So is
there anyway to find out codec negotiated for each h323 call via chan_h323
channel?
Thanks, all
Tommy
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michiel Betel
Sent: Saturday, February 28, 2004 2:45 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] RE: simple H323 question
Ron McMillan wrote:
One way to do it is to use a sniffer, such as ethereal, to capture the
traffic. You should see it in capability exchange, but also easily see in
RTP packets. There might be better ways. But if you're interested in
pursuing it this way and not sure how to do, please follow up with another
question...
Ron
On Fri, 27 Feb 2004, T. Chan wrote:
Hi, all
I wonder when passing calls through asterisk with H323, is there anyway to
find out what codec the calls are using, anyone can help please, thanks
alot
!
TC
TC, When using chan_oh323 the codec used is stored in the variable
${OH323_CHANCODEC}
Michiel
___
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
---
Incoming mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004
---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004
___
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] Anybody managed to call a phone through sipgate.de
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
David Hajek wrote:
| Is there english version of their sipgate.de website?
no ... I just tried the google translater - it did not work (for me) I
think the translation programs don't work with php pages...
Birk
|
| -D
|
|
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Birk Bremer
|Sent: Friday, February 27, 2004 7:06 PM
|To: [EMAIL PROTECTED]
|Subject: Re: [Asterisk-Users] Anybody managed to call a phone
|through sipgate.de
|
| Hi David,
|
| no the number after the slash is necessary (and yes this is
| my number) Without that slash/number I'm not able to get a
| call anymore.
|
| But thanks
|
| Birk
|
|
|
|
| David J Carter wrote:
| | Hi,
| |
| | I would be tempted to get rid of the slash and number on
| the register
| line,
| | unless your asterisk extension is 02115800.
| |
| | dave
| |
| | -Original Message-
| | From: [EMAIL PROTECTED]
| | [mailto:[EMAIL PROTECTED] Behalf Of
| Birk Bremer
| | Sent: 27 February 2004 16:47
| | To: [EMAIL PROTECTED]
| | Subject: [Asterisk-Users] Anybody managed to call a phone through
| | sipgate.de
| |
| |
| | Hello everybody,
| |
| | has anybody managed to call a (old fashioned) phone using
| Sipgate.de
| | and asterisk? (yes I have money on my account :-) )
| |
| |
| | The configuration I got from the sipgate.de people is at
| the botton of
| | the mail
| |
| |
| | Here is mine:
| |
| | sip.conf:
| |
| | register = 800:[EMAIL PROTECTED]/02115800
| |
| | [sipgate]
| | type=friend
| | username=800
| | secret=SECRET
| | host=sipgate.de
| | fromuser=800
| | fromdomain=sipgate.net
| | nat=no
| | ;dtmfband=3Dinband
| | context=sipin
| | canreinvite=no
| |
| |
| | extension.conf:
| | exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
| |
| | To be called on my sipgate number - no problem
| |
| | If I want to call somebody I get the following error:
| |
| | When I call a number directly out of the softphone:
| | Executing Dial([EMAIL PROTECTED]/2,
| SIP/[EMAIL PROTECTED]|30|tr)
| | in new stack
| | ~-- Called [EMAIL PROTECTED]
| | ~-- Got SIP response 403 Forbidden back from 217.10.79.9
| | ~ == No one is available to answer at this time
| | ~-- Hungup '[EMAIL PROTECTED]/2
| |
| |
| |
| | when I use the webinterface at sipgate.de I get a ring at my
| | softphone, when I pick the call I get the message (in the appearing
| | box) Teilnehmer nicht gefunden - User/Number not found
| |
| | sometimes (while tried different config. I also got (at *
| console) to
| | many hops...
| |
| |
| | Has anybody managed this - can you please send me your
| configuration
| | (sip, extensions) or can anybody help
| |
| | Thanks in advance
| |
| | Birk Bremer
| |
| |
| |
| |
| |
| | The configuration the sipgate people suggest:
| |
| | ~ register = 800:[EMAIL PROTECTED]/800
| | ^ can't be correct
| | |
| | |
| | |
| | | [sipgate]
| | |
| | | type=friend
| | |
| | | username=800
| | |
| | | secret=sipgatepasswort
| | |
| | | host=sipgate.de
| | |
| | | fromuser=800
| | |
| | | fromdomain=sipgate.net
| | |
| | | nat=yes
| | |
| | | ;dtmfband=inband
| | |
| | | context=incomingsipgate
| | |
| | | canreinvite=no
| | |
| | |
| | |
| | | Aus der extensions.conf :
| | |
| | |
| | |
| | | [incomingsipgate]
| | |
| | | exten = h,1,Hangup
| | |
| | | exten = 800,1,Dial(SIP/internestelefon,20,tr)
| | |
| | |
| | |
| | | [sipgate]
| | |
| | | exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
| | |
| | | exten = _9.,2,Playback(invalid)
| | |
| | | exten = _9.,3,Hangup
|
| ___
| 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 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
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux)
Comment: Using GnuPG with Debian - http://enigmail.mozdev.org
iD8DBQFAQHWS7QhrwFQeHVsRAmYZAJ9qRsavtw1tO4s+A8wJ8BjA9dv7rQCffVSz
WtJYNw+f1EKu5y/sfE5fVlA=
=Wkng
-END PGP SIGNATURE-
RE: [Asterisk-Users] Anybody managed to call a phone through sipgate.de
Hi,
Are you behind a NAT/Firewall?
dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Birk Bremer
Sent: 28 February 2004 11:04
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Anybody managed to call a phone through
sipgate.de
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
David Hajek wrote:
| Is there english version of their sipgate.de website?
no ... I just tried the google translater - it did not work (for me) I
think the translation programs don't work with php pages...
Birk
|
| -D
|
|
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Birk Bremer
|Sent: Friday, February 27, 2004 7:06 PM
|To: [EMAIL PROTECTED]
|Subject: Re: [Asterisk-Users] Anybody managed to call a phone
|through sipgate.de
|
| Hi David,
|
| no the number after the slash is necessary (and yes this is
| my number) Without that slash/number I'm not able to get a
| call anymore.
|
| But thanks
|
| Birk
|
|
|
|
| David J Carter wrote:
| | Hi,
| |
| | I would be tempted to get rid of the slash and number on
| the register
| line,
| | unless your asterisk extension is 02115800.
| |
| | dave
| |
| | -Original Message-
| | From: [EMAIL PROTECTED]
| | [mailto:[EMAIL PROTECTED] Behalf Of
| Birk Bremer
| | Sent: 27 February 2004 16:47
| | To: [EMAIL PROTECTED]
| | Subject: [Asterisk-Users] Anybody managed to call a phone through
| | sipgate.de
| |
| |
| | Hello everybody,
| |
| | has anybody managed to call a (old fashioned) phone using
| Sipgate.de
| | and asterisk? (yes I have money on my account :-) )
| |
| |
| | The configuration I got from the sipgate.de people is at
| the botton of
| | the mail
| |
| |
| | Here is mine:
| |
| | sip.conf:
| |
| | register = 800:[EMAIL PROTECTED]/02115800
| |
| | [sipgate]
| | type=friend
| | username=800
| | secret=SECRET
| | host=sipgate.de
| | fromuser=800
| | fromdomain=sipgate.net
| | nat=no
| | ;dtmfband=3Dinband
| | context=sipin
| | canreinvite=no
| |
| |
| | extension.conf:
| | exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
| |
| | To be called on my sipgate number - no problem
| |
| | If I want to call somebody I get the following error:
| |
| | When I call a number directly out of the softphone:
| | Executing Dial([EMAIL PROTECTED]/2,
| SIP/[EMAIL PROTECTED]|30|tr)
| | in new stack
| | ~-- Called [EMAIL PROTECTED]
| | ~-- Got SIP response 403 Forbidden back from 217.10.79.9
| | ~ == No one is available to answer at this time
| | ~-- Hungup '[EMAIL PROTECTED]/2
| |
| |
| |
| | when I use the webinterface at sipgate.de I get a ring at my
| | softphone, when I pick the call I get the message (in the appearing
| | box) Teilnehmer nicht gefunden - User/Number not found
| |
| | sometimes (while tried different config. I also got (at *
| console) to
| | many hops...
| |
| |
| | Has anybody managed this - can you please send me your
| configuration
| | (sip, extensions) or can anybody help
| |
| | Thanks in advance
| |
| | Birk Bremer
| |
| |
| |
| |
| |
| | The configuration the sipgate people suggest:
| |
| | ~ register = 800:[EMAIL PROTECTED]/800
| | ^ can't be correct
| | |
| | |
| | |
| | | [sipgate]
| | |
| | | type=friend
| | |
| | | username=800
| | |
| | | secret=sipgatepasswort
| | |
| | | host=sipgate.de
| | |
| | | fromuser=800
| | |
| | | fromdomain=sipgate.net
| | |
| | | nat=yes
| | |
| | | ;dtmfband=inband
| | |
| | | context=incomingsipgate
| | |
| | | canreinvite=no
| | |
| | |
| | |
| | | Aus der extensions.conf :
| | |
| | |
| | |
| | | [incomingsipgate]
| | |
| | | exten = h,1,Hangup
| | |
| | | exten = 800,1,Dial(SIP/internestelefon,20,tr)
| | |
| | |
| | |
| | | [sipgate]
| | |
| | | exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
| | |
| | | exten = _9.,2,Playback(invalid)
| | |
| | | exten = _9.,3,Hangup
|
| ___
| 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 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
Re: [Asterisk-Users] Anybody managed to call a phone through sipgate.de
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
The Server I use is somewhere in the Internet with a real ip. Myself and
others connect to the server via vpn in order to go through various
firewalls. Since I can get calls but only can't place calls (via
sipgate.de) I don't think it is a firewall matter...
Birk
David J Carter wrote:
| Hi,
|
| Are you behind a NAT/Firewall?
|
| dave
|
| -Original Message-
| From: [EMAIL PROTECTED]
| [mailto:[EMAIL PROTECTED] Behalf Of Birk Bremer
| Sent: 28 February 2004 11:04
| To: [EMAIL PROTECTED]
| Subject: Re: [Asterisk-Users] Anybody managed to call a phone through
| sipgate.de
|
|
| David Hajek wrote:
| | Is there english version of their sipgate.de website?
|
|
| no ... I just tried the google translater - it did not work (for me) I
| think the translation programs don't work with php pages...
|
| Birk
|
|
| |
| | -D
| |
| |
| |-Original Message-
| |From: [EMAIL PROTECTED]
| |[mailto:[EMAIL PROTECTED] On Behalf Of
| |Birk Bremer
| |Sent: Friday, February 27, 2004 7:06 PM
| |To: [EMAIL PROTECTED]
| |Subject: Re: [Asterisk-Users] Anybody managed to call a phone
| |through sipgate.de
| |
| | Hi David,
| |
| | no the number after the slash is necessary (and yes this is
| | my number) Without that slash/number I'm not able to get a
| | call anymore.
| |
| | But thanks
| |
| | Birk
| |
| |
| |
| |
| | David J Carter wrote:
| | | Hi,
| | |
| | | I would be tempted to get rid of the slash and number on
| | the register
| | line,
| | | unless your asterisk extension is 02115800.
| | |
| | | dave
| | |
| | | -Original Message-
| | | From: [EMAIL PROTECTED]
| | | [mailto:[EMAIL PROTECTED] Behalf Of
| | Birk Bremer
| | | Sent: 27 February 2004 16:47
| | | To: [EMAIL PROTECTED]
| | | Subject: [Asterisk-Users] Anybody managed to call a phone through
| | | sipgate.de
| | |
| | |
| | | Hello everybody,
| | |
| | | has anybody managed to call a (old fashioned) phone using
| | Sipgate.de
| | | and asterisk? (yes I have money on my account :-) )
| | |
| | |
| | | The configuration I got from the sipgate.de people is at
| | the botton of
| | | the mail
| | |
| | |
| | | Here is mine:
| | |
| | | sip.conf:
| | |
| | | register = 800:[EMAIL PROTECTED]/02115800
| | |
| | | [sipgate]
| | | type=friend
| | | username=800
| | | secret=SECRET
| | | host=sipgate.de
| | | fromuser=800
| | | fromdomain=sipgate.net
| | | nat=no
| | | ;dtmfband=3Dinband
| | | context=sipin
| | | canreinvite=no
| | |
| | |
| | | extension.conf:
| | | exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
| | |
| | | To be called on my sipgate number - no problem
| | |
| | | If I want to call somebody I get the following error:
| | |
| | | When I call a number directly out of the softphone:
| | | Executing Dial([EMAIL PROTECTED]/2,
| | SIP/[EMAIL PROTECTED]|30|tr)
| | | in new stack
| | | ~-- Called [EMAIL PROTECTED]
| | | ~-- Got SIP response 403 Forbidden back from 217.10.79.9
| | | ~ == No one is available to answer at this time
| | | ~-- Hungup '[EMAIL PROTECTED]/2
| | |
| | |
| | |
| | | when I use the webinterface at sipgate.de I get a ring at my
| | | softphone, when I pick the call I get the message (in the appearing
| | | box) Teilnehmer nicht gefunden - User/Number not found
| | |
| | | sometimes (while tried different config. I also got (at *
| | console) to
| | | many hops...
| | |
| | |
| | | Has anybody managed this - can you please send me your
| | configuration
| | | (sip, extensions) or can anybody help
| | |
| | | Thanks in advance
| | |
| | | Birk Bremer
| | |
|
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux)
Comment: Using GnuPG with Debian - http://enigmail.mozdev.org
iD8DBQFAQHwy7QhrwFQeHVsRAgHIAKCcm9fr2CoIVAaTLGLkoUaGF6uZdwCfRaMd
n54rHyhWAMcQSCKXZNTbEfk=
=Mzc2
-END PGP SIGNATURE-
___
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] Anybody managed to call a phone through sipgate.de
Hi again,
What is your sipgate number, I have just setup my asterisk to call a sipgate
numbar and it rings.
If you want to call me, then try my IAXTEL # 1 700 818 8820
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Birk Bremer
Sent: 28 February 2004 11:04
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Anybody managed to call a phone through
sipgate.de
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
David Hajek wrote:
| Is there english version of their sipgate.de website?
no ... I just tried the google translater - it did not work (for me) I
think the translation programs don't work with php pages...
Birk
|
| -D
|
|
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Birk Bremer
|Sent: Friday, February 27, 2004 7:06 PM
|To: [EMAIL PROTECTED]
|Subject: Re: [Asterisk-Users] Anybody managed to call a phone
|through sipgate.de
|
| Hi David,
|
| no the number after the slash is necessary (and yes this is
| my number) Without that slash/number I'm not able to get a
| call anymore.
|
| But thanks
|
| Birk
|
|
|
|
| David J Carter wrote:
| | Hi,
| |
| | I would be tempted to get rid of the slash and number on
| the register
| line,
| | unless your asterisk extension is 02115800.
| |
| | dave
| |
| | -Original Message-
| | From: [EMAIL PROTECTED]
| | [mailto:[EMAIL PROTECTED] Behalf Of
| Birk Bremer
| | Sent: 27 February 2004 16:47
| | To: [EMAIL PROTECTED]
| | Subject: [Asterisk-Users] Anybody managed to call a phone through
| | sipgate.de
| |
| |
| | Hello everybody,
| |
| | has anybody managed to call a (old fashioned) phone using
| Sipgate.de
| | and asterisk? (yes I have money on my account :-) )
| |
| |
| | The configuration I got from the sipgate.de people is at
| the botton of
| | the mail
| |
| |
| | Here is mine:
| |
| | sip.conf:
| |
| | register = 800:[EMAIL PROTECTED]/02115800
| |
| | [sipgate]
| | type=friend
| | username=800
| | secret=SECRET
| | host=sipgate.de
| | fromuser=800
| | fromdomain=sipgate.net
| | nat=no
| | ;dtmfband=3Dinband
| | context=sipin
| | canreinvite=no
| |
| |
| | extension.conf:
| | exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
| |
| | To be called on my sipgate number - no problem
| |
| | If I want to call somebody I get the following error:
| |
| | When I call a number directly out of the softphone:
| | Executing Dial([EMAIL PROTECTED]/2,
| SIP/[EMAIL PROTECTED]|30|tr)
| | in new stack
| | ~-- Called [EMAIL PROTECTED]
| | ~-- Got SIP response 403 Forbidden back from 217.10.79.9
| | ~ == No one is available to answer at this time
| | ~-- Hungup '[EMAIL PROTECTED]/2
| |
| |
| |
| | when I use the webinterface at sipgate.de I get a ring at my
| | softphone, when I pick the call I get the message (in the appearing
| | box) Teilnehmer nicht gefunden - User/Number not found
| |
| | sometimes (while tried different config. I also got (at *
| console) to
| | many hops...
| |
| |
| | Has anybody managed this - can you please send me your
| configuration
| | (sip, extensions) or can anybody help
| |
| | Thanks in advance
| |
| | Birk Bremer
| |
| |
| |
| |
| |
| | The configuration the sipgate people suggest:
| |
| | ~ register = 800:[EMAIL PROTECTED]/800
| | ^ can't be correct
| | |
| | |
| | |
| | | [sipgate]
| | |
| | | type=friend
| | |
| | | username=800
| | |
| | | secret=sipgatepasswort
| | |
| | | host=sipgate.de
| | |
| | | fromuser=800
| | |
| | | fromdomain=sipgate.net
| | |
| | | nat=yes
| | |
| | | ;dtmfband=inband
| | |
| | | context=incomingsipgate
| | |
| | | canreinvite=no
| | |
| | |
| | |
| | | Aus der extensions.conf :
| | |
| | |
| | |
| | | [incomingsipgate]
| | |
| | | exten = h,1,Hangup
| | |
| | | exten = 800,1,Dial(SIP/internestelefon,20,tr)
| | |
| | |
| | |
| | | [sipgate]
| | |
| | | exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
| | |
| | | exten = _9.,2,Playback(invalid)
| | |
| | | exten = _9.,3,Hangup
|
| ___
| 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 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]
|
RE: [Asterisk-Users] Anybody managed to call a phone through sipgate.de
Birk,
Even using VPN to get to the server you will still have I assume a private
IP address on the VPN side. This will pass through a NAT/Firewall to the
outside world. This may or may not be on the server you connect to, but I
would bet you still pass through a NAT/Firewall.
I assume your connection is something like: -
Softphone Asterisk VPN to Server -- Server ---
Firewall/NAT/Router - Internet
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Birk Bremer
Sent: 28 February 2004 11:32
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Anybody managed to call a phone through
sipgate.de
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
The Server I use is somewhere in the Internet with a real ip. Myself and
others connect to the server via vpn in order to go through various
firewalls. Since I can get calls but only can't place calls (via
sipgate.de) I don't think it is a firewall matter...
Birk
David J Carter wrote:
| Hi,
|
| Are you behind a NAT/Firewall?
|
| dave
|
| -Original Message-
| From: [EMAIL PROTECTED]
| [mailto:[EMAIL PROTECTED] Behalf Of Birk Bremer
| Sent: 28 February 2004 11:04
| To: [EMAIL PROTECTED]
| Subject: Re: [Asterisk-Users] Anybody managed to call a phone through
| sipgate.de
|
|
| David Hajek wrote:
| | Is there english version of their sipgate.de website?
|
|
| no ... I just tried the google translater - it did not work (for me) I
| think the translation programs don't work with php pages...
|
| Birk
|
|
| |
| | -D
| |
| |
| |-Original Message-
| |From: [EMAIL PROTECTED]
| |[mailto:[EMAIL PROTECTED] On Behalf Of
| |Birk Bremer
| |Sent: Friday, February 27, 2004 7:06 PM
| |To: [EMAIL PROTECTED]
| |Subject: Re: [Asterisk-Users] Anybody managed to call a phone
| |through sipgate.de
| |
| | Hi David,
| |
| | no the number after the slash is necessary (and yes this is
| | my number) Without that slash/number I'm not able to get a
| | call anymore.
| |
| | But thanks
| |
| | Birk
| |
| |
| |
| |
| | David J Carter wrote:
| | | Hi,
| | |
| | | I would be tempted to get rid of the slash and number on
| | the register
| | line,
| | | unless your asterisk extension is 02115800.
| | |
| | | dave
| | |
| | | -Original Message-
| | | From: [EMAIL PROTECTED]
| | | [mailto:[EMAIL PROTECTED] Behalf Of
| | Birk Bremer
| | | Sent: 27 February 2004 16:47
| | | To: [EMAIL PROTECTED]
| | | Subject: [Asterisk-Users] Anybody managed to call a phone through
| | | sipgate.de
| | |
| | |
| | | Hello everybody,
| | |
| | | has anybody managed to call a (old fashioned) phone using
| | Sipgate.de
| | | and asterisk? (yes I have money on my account :-) )
| | |
| | |
| | | The configuration I got from the sipgate.de people is at
| | the botton of
| | | the mail
| | |
| | |
| | | Here is mine:
| | |
| | | sip.conf:
| | |
| | | register = 800:[EMAIL PROTECTED]/02115800
| | |
| | | [sipgate]
| | | type=friend
| | | username=800
| | | secret=SECRET
| | | host=sipgate.de
| | | fromuser=800
| | | fromdomain=sipgate.net
| | | nat=no
| | | ;dtmfband=3Dinband
| | | context=sipin
| | | canreinvite=no
| | |
| | |
| | | extension.conf:
| | | exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
| | |
| | | To be called on my sipgate number - no problem
| | |
| | | If I want to call somebody I get the following error:
| | |
| | | When I call a number directly out of the softphone:
| | | Executing Dial([EMAIL PROTECTED]/2,
| | SIP/[EMAIL PROTECTED]|30|tr)
| | | in new stack
| | | ~-- Called [EMAIL PROTECTED]
| | | ~-- Got SIP response 403 Forbidden back from 217.10.79.9
| | | ~ == No one is available to answer at this time
| | | ~-- Hungup '[EMAIL PROTECTED]/2
| | |
| | |
| | |
| | | when I use the webinterface at sipgate.de I get a ring at my
| | | softphone, when I pick the call I get the message (in the appearing
| | | box) Teilnehmer nicht gefunden - User/Number not found
| | |
| | | sometimes (while tried different config. I also got (at *
| | console) to
| | | many hops...
| | |
| | |
| | | Has anybody managed this - can you please send me your
| | configuration
| | | (sip, extensions) or can anybody help
| | |
| | | Thanks in advance
| | |
| | | Birk Bremer
| | |
|
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux)
Comment: Using GnuPG with Debian - http://enigmail.mozdev.org
iD8DBQFAQHwy7QhrwFQeHVsRAgHIAKCcm9fr2CoIVAaTLGLkoUaGF6uZdwCfRaMd
n54rHyhWAMcQSCKXZNTbEfk=
=Mzc2
-END PGP SIGNATURE-
___
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:
[Asterisk-Users] Cisco 7960 sip v6.2 is out
FWIW... Version 6.2 of Cisco's sip code for the 7940/7960 was posted on Cisco's download site Feb 17th. The v6.2 release notes suggest the following caveats were addressed: SIPPhone: CANCEL messages not formatted properly after 180 received Branch ID is not compliant to RFC3261 SIP IP Phone resets rtp session when it receives sip re-invite Open v6.2 caveats include: SIPPhone: DND config causes weird NTP behavior Media takes 0.4 sec to be set up The v6.1 coded reportedly addressed 18 caveats from v6.0 code. ___ 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] Asterisk PABX switch
I wonder if the next is possible with *: PABX | E1 | PABX E1- Asterisk E1 PABX | \ E1 \ | IP PABX \ Cisco 827V Analogue PBX If possible, how much power the CPU must have? Much appreciate any help. Nikolay Koev
[Asterisk-Users] OTish: Firefly Crashing with *
- Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, February 29, 2004 1:44 AM Subject: OTish: Firefly Carshing with * Firefly seems to be crashing when I dial from the console (i.e. Dial [EMAIL PROTECTED]) but works fine from telephone... Also, it cuts my ringing sound off after about .5 seconds. Any ideas? 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] OTish: Firefly Crashing with *
Hi, -Original Message- Firefly seems to be crashing when I dial from the console (i.e. Dial [EMAIL PROTECTED]) but works fine from telephone... Also, it cuts my ringing sound off after about .5 seconds. What version are you using ? There was a small bug in Firefly, fixed last week: From Adam Hart on 20-feb-2004: I've released a beta version of the new firefly, to address the crashing issue with incoming calls from Asterisk. (the problem was I assumed the caller id would be populated). Also, firefly will now reject calls if there's no common codecs. I'd recommended anyone using firefly with asterisk should get it http://www.virbiage.com/firefly/download/firefly-dev.exe Also, I'm trying to get Firefly running under wine. I have no experience with Wine so any help or pointers to what needs fixed would be helpful. Steps for getting firefly running 1) Install the above firefly version (1.5 or later) 2) Remove extensions.dll 3) Run firefly.exe -internalblend (this will indicate firefly to use it's own blend func instead of window's blend func) 4) Let me know your results If anyone else experiences any other issues with firefly asterisk, let me know. SIP version out soonish. -Adam ___ 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] Iconnect behind NAT
carl wrote: Anyone got an example of sip and extensions confs for Iconnect outgoing calls behind NAT. Here you go: [Scene starts out with you on the phone with IConnect technical support.] You: I know that Asterisk isn't one of your supported platforms. I'm not asking you to support my 'device'... I'm asking you to support your 'service'... Why is it that I can't have multiple outbound calls at a time? Why doesn't inbound caller-ID work right when someone is calling from a Nextel phone? Why do calls I make show up with no caller-ID? I need them to show caller-ID or the people I'm calling won't answer the phone. Why do I have to wait several (10-15) seconds between calls to prevent getting congestion tone from IConnect? Iconnect: We do not support Asterisk. You: Cancel my account. I'm going to find a REAL provider. [curtain closes - both on the scene and on IConnect.] Seriously, you're much better off finding a provider that will support IAX interconnect as well as address the problems in our scene. I'll be happy to get you set up with IAX peering. Drop me an email if you're interested. John ___ 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] Asterisk PABX switch
Nikolay Koev wrote: I wonder if the next is possible with *: PABX | E1 | PABX E1- Asterisk E1PABX | \ E1\ | IP PABX \ Cisco 827V Analogue PBX Yes, this is possible to do, assuming your other IP PBX supports on of the VoIP protocols * does. You'll also need a TE405P or a TE410P for the E1 interface. If possible, how much power the CPU must have? Since you'll be doing encoding and decoding on a bunch of channels, you'll want a farly beefy setup. Nick ___ 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] Galaxy Voice - Good or Bad?
I am look into Galaxy Voice's service as they provide numbers in my area. Has anyone on the list had any experience using this service with Asterisk? I am interested to know if it can be made with work with Asterisk. How is the quality of sound? Are there limits with the number of concurrent calls? How is the reliability of the service? Thanks, Robert ___ 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] sip:user@domain.tld
If I want users to be able to call each other (or others to be able to call users on our Asterisk system) using their email address ([EMAIL PROTECTED]) what would have to be done? I am guessing the folowing.. In sip.conf the phone definition would have to be.. [user.name] secret.. blah.. In extentions.conf I would probably have to have a line like.. exten = user.name,1,Dial(SIP/user.name) exten = user.name,2,blah.. I would have to allow anon access to a default context which will be able to contact the extensions.. Is this possible? And finally I would need a DNS server that supproted SRV records.. Have I left anything out? Is the ability to query DNS for a SRV record something that may not be supported by IP phones or does it just work? Thanks.. ___ 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] How to compile bri-stuff.0.0.2.rc12
I am new to * and I am trying to set up a test box with a ISDN card with the cologne chip set (twin towers on the isdn chip) . I have downloaded bri-stuff.0.0.2.rc12 from www.junghanns.net site. I would like to test with a driver that supports echo cancellation in software. I am running the ./inshall.sh script and the download etc commenced fine. I first tried with a redHat kernel, but were told that that was a no-no. I then downloaded a 2.4.24 vanilla kernel from www.kernel.org The question is : what configurations need to be made to the kernel to get the zaphfc stuff compile. I get a lot off errors and warnings and it might be due to a wrongly configured kernel ? Is there anything else you could share on how to get this stuff up and running ? Regards MH _ MSN Hotmail http://www.hotmail.com Med markedets beste SPAM-filter. Gratis! ___ 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] sip:user@domain.tld
That's all you need. At least, that's kinda how I have mine set up and it works fine to dial-by-email. WipeOut wrote: If I want users to be able to call each other (or others to be able to call users on our Asterisk system) using their email address ([EMAIL PROTECTED]) what would have to be done? I am guessing the folowing.. In sip.conf the phone definition would have to be.. [user.name] secret.. blah.. In extentions.conf I would probably have to have a line like.. exten = user.name,1,Dial(SIP/user.name) exten = user.name,2,blah.. I would have to allow anon access to a default context which will be able to contact the extensions.. Is this possible? And finally I would need a DNS server that supproted SRV records.. Have I left anything out? Is the ability to query DNS for a SRV record something that may not be supported by IP phones or does it just work? Thanks.. ___ 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
Re: [Asterisk-Users] retrieve_sip_conf_from_mysql.pl data format
On Fri, 2004-02-27 at 21:39, Chad Sawyer wrote: In the contrib/scripts directory I have been trying to figure out the format of the entries in the MySQL table. It isn't at all obvious is it? I've now worked out what it does have written this up on the Wiki, along with my previous post about database integration in general: http://voip-info.org/tiki-index.php?page=Asterisk+sip+conf+from+mysql http://voip-info.org/wiki-Asterisk+configuration+from+database Now back to the task of getting a workable UI for my specific situation's needs ;) F ___ 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] G729 troubles
I am a new asterisk user. I have had a box up and running for a couple of months and been very happy with it. Last night I came up with a question that I have not been able to find an answer too. I purchased 5 licenses for the G729 codec from digium. My source is current from CVS as of late last night. Here are messages I'm getting from Asterisk. Can anybody help me? [codec_g729b.so] = (Annex B (floating point) G.729/PCM16 Codec Translator) Feb 28 08:47:49 WARNING[-1084458064]: asterisk.c:260 listener: Select retured er ror: Interrupted system call Feb 28 08:47:49 WARNING[-1084458064]: asterisk.c:260 listener: Select retured er ror: Interrupted system call == Detected 5 licensed G.729 transcoders Feb 28 08:47:49 WARNING[-1084456832]: translate.c:219 calc_cost: Translator 'g72 9tolinb' does not produce sample frames. == Registered translator 'g729tolinb' from format G729A to SLINR, cost 9 == Registered translator 'lintog729b' from format SLINR to G729A, cost 26 Thanks so much, Darren Wiebe [EMAIL PROTECTED] ___ 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] Can You Specify Codec Per Extension?
Matt wrote: I looked at my sample config's and I cannot find an example of an extension where you specify the codec differently for each extension. Can someone show me a sample extension? You're looking in the wrong place. (I should have been more specific.) You specify the codecs when you set up the device in asterisk. You want to specify the codecs in your sip.conf and h323.conf where you set up each device that will be talking to asterisk. - Andrew Thompson http://aktzero.com/ ___ 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] G729 troubles
I forgot to mention what I have been trying to fix it. I'm running it from the console asterisk -vvvcng but this does not help. I've searched the mailing lists and found a lot of messages with people having the same problem. I'll try calling digium Monday if I cannot resolve it today and see if they can help me. Darren Wiebe [EMAIL PROTECTED] Darren Wiebe wrote: I am a new asterisk user. I have had a box up and running for a couple of months and been very happy with it. Last night I came up with a question that I have not been able to find an answer too. I purchased 5 licenses for the G729 codec from digium. My source is current from CVS as of late last night. Here are messages I'm getting from Asterisk. Can anybody help me? [codec_g729b.so] = (Annex B (floating point) G.729/PCM16 Codec Translator) Feb 28 08:47:49 WARNING[-1084458064]: asterisk.c:260 listener: Select retured er ror: Interrupted system call Feb 28 08:47:49 WARNING[-1084458064]: asterisk.c:260 listener: Select retured er ror: Interrupted system call == Detected 5 licensed G.729 transcoders Feb 28 08:47:49 WARNING[-1084456832]: translate.c:219 calc_cost: Translator 'g72 9tolinb' does not produce sample frames. == Registered translator 'g729tolinb' from format G729A to SLINR, cost 9 == Registered translator 'lintog729b' from format SLINR to G729A, cost 26 Thanks so much, Darren Wiebe [EMAIL PROTECTED] ___ 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
Re: [Asterisk-Users] wisip firmware, updates, features??
NOt sure if there is an official download site, but I just recieved a copy of the updated firmware from pulver. I can send it to you if you like. I have emailed back asking for instructions on how to load. -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Quoting Miguel Cavazos [EMAIL PROTECTED]: hi guys finally i got my wisip this week and im very happy with it. It works but i was wondering anyone know where can i find new firmware, updates or a wish list? I cross emails with jeff pulver about having a small http browser for auth on starbucks hotspots mcdonalds or prodigy movil(mexico). Even to check some text things via web maybe email??? He seems not to be so intrested so ill try emailing the manufacture. However if someone has a useful url or can tell me where to find this information please send me an email. Miguel ___ 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 Visit Winfield Public Schools at http://usd465.com - This mail sent through IMP: http://horde.org/imp/ ___ 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] wisip firmware, updates, features??
Hi Johnathan, I wouldn't mind a copy of the firmware if you could send it. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jonathan Moore Sent: 28 February 2004 19:24 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] wisip firmware, updates, features?? NOt sure if there is an official download site, but I just recieved a copy of the updated firmware from pulver. I can send it to you if you like. I have emailed back asking for instructions on how to load. -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Quoting Miguel Cavazos [EMAIL PROTECTED]: hi guys finally i got my wisip this week and im very happy with it. It works but i was wondering anyone know where can i find new firmware, updates or a wish list? I cross emails with jeff pulver about having a small http browser for auth on starbucks hotspots mcdonalds or prodigy movil(mexico). Even to check some text things via web maybe email??? He seems not to be so intrested so ill try emailing the manufacture. However if someone has a useful url or can tell me where to find this information please send me an email. Miguel ___ 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 Visit Winfield Public Schools at http://usd465.com - This mail sent through IMP: http://horde.org/imp/ ___ 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
Re: [Asterisk-Users] HT 286 Any information about will be great !!!
The original message isn't copied because its HTML encoded. Basicly the author wanted to know if he could use a HT-286 (Grandstream) to bypass the PBX, generate busy, answer calls, etc. The Grandstream HT-286, and the Sipura SPA-2000 are both ATA FXS based devices. ATA == Analog Telephone Adapter FXS == Foreign eXchange Station (if memory is correct). FXS devices are designed to have telephones plugged into them. Basicly things that need a ringer voltage/signal get plugged into a FXS device. You DONT plug a FSX device into the telco PSTN line. I've taken FSX/ATA devices and plugged them into PBX's. Take the PSTN lines and plugged those into FXO cards that are inside a Asterisk box. Using the Asterisk box as front end processor of calls. If the FXO lines have DID you can then do interesting routing tricks. In one application we took multiple lines feeding multiple gas stations and pulled them back via IP to a single location in the city. By collapsing the remote line counts it became cost affective to get a PRI for all the voice traffic. Hope this helps. John Brown Chagres Technologies, Inc (Americas) Chagres Technologies, B.V. (EMEA) ___ 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] E911 support
From a New Mexico perspective, When you order a PRI from a CLEC they typically will dump your CLID info and replace it with the main number on the span. You can request that they not do this and that they pass the CLID thru. We have been working with some of the local PSAP's here, CLEC's and the ILEC. In about 2 weeks we are going to do some live 911 trials using asterisk. The goal is for us to set the CallerID to a test number, then dial 911 and see which PSAP the call goes to, and if the CLID shows up. If it does, then some issues are resolved for us. There is also a special circuit you can purchase from the ILEC that is basicly a 911 T1 interface into the selective router the ILEC maintains here. Its a little pricey, but still worth it. So we could also switch calls inside Asterisk to the 911 truck directly. In either case, the County PSAP and the state 911 director have been very very helpful and most willing to be involved. NENA has also been very helpfull as has Intrado (once you sign NDA documents). My recommendation is that if you are looking at 911 issues, go buy your local PSAP manager a cup of coffee and listen to how things work in your area. The country doesnt work the same in all areas. Help educate them on how this VoIP stuff works, most are dealing with two full plates and trying to get Cell 911 stuff up to Phase 1, or in some cases Phase 2 standards. john brown chagres technologies, inc (americas) chagres technologies, B.V. (emea) On Thu, Feb 26, 2004 at 04:31:41PM -0500, John Fraizer wrote: Steve Dolloff wrote: I have the following in my sip.conf entries: callerid=Anonymous 8885551212 This still passes the number for 911, but flags the call as private. I believe this will meet your requirements. Stephen OK. I was under the impression that the PSAP got their information based on ALI/ANI and not from CLID. Are you telling me that they're looking at CLID? Also, at least in the testing I've done, the text portion of the CLID string is ignored by the telco. They only look at the number and generate the text based on what is in their database. IE; If I tell my asterisk server to set my callerID to test my home number and call someplace, What I get on the CLID display of the phone I dial is John Fraizer and my home number. Since Powell has stated that we must provide E911 services, I am wondering what precisely is going to have to be done to do so with Asterisk. Routing the call to the PSAP when someone dials 911 is the easy part. Sending all of the information they want/need (much more than just CLID and something that is regulated) is an alltogether different story. John ___ 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] zaphfc bri with overlap sending/receiving
Hi all,
I am currently testing Klaus-Peter Junghanns' zaphfc bri driver
0.0.2rc12 with two HFC ISDN cards in PtP setup - one connected to telco,
one to the legacy pbx - and try to dial from a pbx extension out to the
pstn through astersik.
This works perfectly as long as I dial on hook and pick up after dialing
the complete number.
Using the isdn phone (and any analog pbx extension which cannot prepare
dialing on hook) the way people are used to (first pick up, then dial)
results in dialing only the first few digits out to the Zap channel
connected to pstn and call setup to fail.
Obviously this is a problem with overlap sending/receiving in the zap
channels. Unfortunately we have a variable length numbering plan in
germany (local numbers can be anything between 4 and 9 digits long), so
putting more X in the regex doesn't seem to be an option.
Ideas how to get this work are greatly appreciated and very welcome. :)
Thank you and regards,
Jan Baumann
My current config:
extensions.conf:
; outbound dialing local calls
; try Enum, then PSTN
[local-pstn]
exten = _0[1-9]XX.,1,EnumLookup(49821${EXTEN:1})
exten = _0[1-9]XX.,2,SetCallerID(49821xx)
exten = _0[1-9]XX.,3,Dial(${ENUM},30)
exten = _0[1-9]XX.,4,Goto(102) ; Failure on SIP, fallback to PSTN
exten = _0[1-9]XX.,52,Congestion
exten = _0[1-9]XX.,102,SetCallerID(xx)
exten = _0[1-9]XX.,103,Dial(Zap/g1/${EXTEN:1},,tr)
exten = _0[1-9]XX.,104,Congestion
zapata.conf:
switchtype = euroisdn
; to/from ISDN PtP
signalling = bri_cpe
pridialplan=unknown
echocancel=no
immediate=no
group = 1
context=pstn-in
channel = 1-2
; to/from the PBX
signalling = bri_net
pridialplan=unknown
echocancel=no
immediate=no
group = 2
context=intern
channel = 4-5
zaptel.conf:
# PSTN DTAG
span=1,1,3,ccs,ami
bchan=1-2
dchan=3
# PtP to PBX
span=2,0,3,ccs,ami
bchan=4-5
dchan=6
___
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] DTMF Issues with SJPHONE
Same as mine. Strange! I'll keep trying. Cheers. - Original Message - From: Girish Gopinath [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, February 28, 2004 9:53 PM Subject: Re: [Asterisk-Users] DTMF Issues with SJPHONE What ver of SJPHONE? SJPhone Evaluation Version, release Jul 31, 2003, Build: 1.10.187c Girish _ All the news that matters. All the gossip from home. http://www.msn.co.in/NRI/ Specially for NRIs! ___ 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 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 ___ 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] New to T-1/Channel Bank hardware -- help?
I'm considering a small office setup with at least 12 extensions. Seems (as has been stated in previous threads) that for the FXS ports, a T100P and a channel bank could be the most cost-effective way to do this. I've got * set up w/ one X100P and one TDM400P, and have been very happy with it. I have zero experience with T-1 channel bank hardware, and would really like some beginner questions asked before dropping the dime... Anybody willing to help me out, please send email directly. I'll summarize for the list. Thanks, Rob -- Rob Fugina, Systems Guy [EMAIL PROTECTED] -- http://www.geekthing.com My firewall filters MS Office attachments. IF numcooks .maxcooks THEN;SET V broth = 'spoiled';END ___ 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] Iconnect behind NAT
I signed up with nufone. Their customer service is a little bit slow but they seem to be pretty decent. I'd recommend checking them out. www.nufone.net Darren Wiebe [EMAIL PROTECTED] Carl wrote: Ha ha I get the picture :-) I've tried Voicepulse but can't manage to get through with them either. Emailed their customer support a week ago and heard nothing since. They get the destination numbers as I can see it on their cdr records. Any other providers offering IAX interconnects? - Original Message - From: John Fraizer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Sunday, February 29, 2004 2:43 AM Subject: Re: [Asterisk-Users] Iconnect behind NAT carl wrote: Anyone got an example of sip and extensions confs for Iconnect outgoing calls behind NAT. Here you go: [Scene starts out with you on the phone with IConnect technical support.] You: I know that Asterisk isn't one of your supported platforms. I'm not asking you to support my 'device'... I'm asking you to support your 'service'... Why is it that I can't have multiple outbound calls at a time? Why doesn't inbound caller-ID work right when someone is calling from a Nextel phone? Why do calls I make show up with no caller-ID? I need them to show caller-ID or the people I'm calling won't answer the phone. Why do I have to wait several (10-15) seconds between calls to prevent getting congestion tone from IConnect? Iconnect: We do not support Asterisk. You: Cancel my account. I'm going to find a REAL provider. [curtain closes - both on the scene and on IConnect.] Seriously, you're much better off finding a provider that will support IAX interconnect as well as address the problems in our scene. I'll be happy to get you set up with IAX peering. Drop me an email if you're interested. John ___ 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 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 ___ 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] Help needed setting up H323 gateway.
Hi, Can someone offer some assistance in setting up Asterisk as a gateway to connect to a third party gatekeeper. I have looked at the h323.conf.sample file but not sure of the following: Do I need to create a new h323.conf file? Where should this file reside i.e., h323 directory? Do you need to add info to extensions file to point to context in the h323.conf file? How do u send an account number with the call so that the third party gatekeeper can verify? Your help will be much appreciated! Carl. ---Outgoing mail is certified Virus Free.Checked by AVG anti-virus system (http://www.grisoft.com).Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004
Re: [Asterisk-Users] Iconnect behind NAT
I'll give them a whirl. Cheers C. - Original Message - From: Darren Wiebe [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, February 29, 2004 11:00 AM Subject: Re: [Asterisk-Users] Iconnect behind NAT I signed up with nufone. Their customer service is a little bit slow but they seem to be pretty decent. I'd recommend checking them out. www.nufone.net Darren Wiebe [EMAIL PROTECTED] Carl wrote: Ha ha I get the picture :-) I've tried Voicepulse but can't manage to get through with them either. Emailed their customer support a week ago and heard nothing since. They get the destination numbers as I can see it on their cdr records. Any other providers offering IAX interconnects? - Original Message - From: John Fraizer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Sunday, February 29, 2004 2:43 AM Subject: Re: [Asterisk-Users] Iconnect behind NAT carl wrote: Anyone got an example of sip and extensions confs for Iconnect outgoing calls behind NAT. Here you go: [Scene starts out with you on the phone with IConnect technical support.] You: I know that Asterisk isn't one of your supported platforms. I'm not asking you to support my 'device'... I'm asking you to support your 'service'... Why is it that I can't have multiple outbound calls at a time? Why doesn't inbound caller-ID work right when someone is calling from a Nextel phone? Why do calls I make show up with no caller-ID? I need them to show caller-ID or the people I'm calling won't answer the phone. Why do I have to wait several (10-15) seconds between calls to prevent getting congestion tone from IConnect? Iconnect: We do not support Asterisk. You: Cancel my account. I'm going to find a REAL provider. [curtain closes - both on the scene and on IConnect.] Seriously, you're much better off finding a provider that will support IAX interconnect as well as address the problems in our scene. I'll be happy to get you set up with IAX peering. Drop me an email if you're interested. John ___ 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 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 ___ 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 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 ___ 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] Iconnect behind NAT
Carl wrote: I'll give them a whirl. Cheers C. Carl, are you not getting my emails? John ___ 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] Iconnect behind NAT
Carl wrote: I'll give them a whirl. Cheers C. If you email me a username/PW combo, I'll get you an account set up and email you the particulars for this side (or telephone you if you include a number) as soon as I get home from dinner. John ___ 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] iaxComm updates at sourceforge
There are new iaxComm binaries for Windows, Linux and Mac OSX posted at http://iaxclient.sourceforge.net/iaxcomm/index.html These binaries also have the recent library change that allows client to client connections to be handed off correctly. Recent changes include speakerphone mode, blind transfer, music on hold and custom ringtones based upon callerid. ___ 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] zaphfc bri with overlap sending/receiving
how is your outgoing dialplan?
tried into specifing something like
exten = _XXX.,1,Dial(blah/${EXTEN})
note the point : this rule will match
at least 4 digits, but also 5,6,7...N
matteo
Il sab, 2004-02-28 alle 22:34, Jan Baumann ha scritto:
Hi all,
I am currently testing Klaus-Peter Junghanns' zaphfc bri driver
0.0.2rc12 with two HFC ISDN cards in PtP setup - one connected to telco,
one to the legacy pbx - and try to dial from a pbx extension out to the
pstn through astersik.
This works perfectly as long as I dial on hook and pick up after dialing
the complete number.
Using the isdn phone (and any analog pbx extension which cannot prepare
dialing on hook) the way people are used to (first pick up, then dial)
results in dialing only the first few digits out to the Zap channel
connected to pstn and call setup to fail.
Obviously this is a problem with overlap sending/receiving in the zap
channels. Unfortunately we have a variable length numbering plan in
germany (local numbers can be anything between 4 and 9 digits long), so
putting more X in the regex doesn't seem to be an option.
Ideas how to get this work are greatly appreciated and very welcome. :)
Thank you and regards,
Jan Baumann
My current config:
extensions.conf:
; outbound dialing local calls
; try Enum, then PSTN
[local-pstn]
exten = _0[1-9]XX.,1,EnumLookup(49821${EXTEN:1})
exten = _0[1-9]XX.,2,SetCallerID(49821xx)
exten = _0[1-9]XX.,3,Dial(${ENUM},30)
exten = _0[1-9]XX.,4,Goto(102) ; Failure on SIP, fallback to PSTN
exten = _0[1-9]XX.,52,Congestion
exten = _0[1-9]XX.,102,SetCallerID(xx)
exten = _0[1-9]XX.,103,Dial(Zap/g1/${EXTEN:1},,tr)
exten = _0[1-9]XX.,104,Congestion
zapata.conf:
switchtype = euroisdn
; to/from ISDN PtP
signalling = bri_cpe
pridialplan=unknown
echocancel=no
immediate=no
group = 1
context=pstn-in
channel = 1-2
; to/from the PBX
signalling = bri_net
pridialplan=unknown
echocancel=no
immediate=no
group = 2
context=intern
channel = 4-5
zaptel.conf:
# PSTN DTAG
span=1,1,3,ccs,ami
bchan=1-2
dchan=3
# PtP to PBX
span=2,0,3,ccs,ami
bchan=4-5
dchan=6
___
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
--
Brancaleoni Matteo [EMAIL PROTECTED]
Espia - Emmegi Srl
___
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] zaphfc bri: crackling sound
Hello all, trying the zaphfc driver from Klaus-Peter Junghanns with Cologne-Chip PCI card I experience a clearly audible crackling sound during calls through the Zap BRI channel to PSTN. Calling the same destination from the same SIP extension via sipgate.de the sound is perfect. What I hear sounds like massive pattern slipping on the BRI. The channel used is the primary clock source. I have 'echocancel=no' (also tried 'yes') in zapata.conf und this in zaptel.conf: # PSTN DTAG span=1,1,3,ccs,ami bchan=1-2 dchan=3 Help is greatly appreciated because faxing is terribly impossible right now. Thank you and king regards, Jan Baumann ___ 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] PCphoneline FXO to FXS box??
pcphoneline.com sells a little box with two RJ-11 jacks that is supposed to convert an FXS port into an FXO port. According to their blurb, when a call comes in it basically conferences the two lines together. Is anyone out there using this box with Asterisk? Any problems? What happens to callerid when you get an incoming call? I'm thinking about using one of these things with the Grandstream ATA-286 for a spot where I may not have a PC available to put a Digium FXO card into. (Don't have Ethernet where the PSTN jack is, so the easiest thing to do is WiFi it. Seems a shame to dedicate a whole PC to just a single FXO port ...) -T.i.A., Jim ___ 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] Hotel wake-up
Anybody know how to implement a hotel wake-up call feature with *? -- Bill Michaelson - COS, Incorporated - Software Development - [EMAIL PROTECTED] Thanks for putting up with my spam filter! ___ 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] Hotel wake-up
On Sat, Feb 28, 2004 at 08:39:26PM -0500, Bill Michaelson wrote: Anybody know how to implement a hotel wake-up call feature with *? I just wrote an AGI for it. I literally just got it working the day before yesterday, so it's not really 'pretty' yet. I also don't have all of the voice prompts I need, so it's a little rough there, too. I don't have time to go into more detail at the moment, but send me a message directly if you're interested... Rob -- Rob Fugina, Systems Guy [EMAIL PROTECTED] -- http://www.geekthing.com My firewall filters MS Office attachments. Psychoceramics: The study of crackpots. ___ 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] Hotel wake-up
Bill, tell us about your system! How many rooms, what kind of extension set in the rooms, number of outside lines, front desk capabilities, how you bill back tel charges to the room, etc. Have you worked-out the ratio of guests to outside lines? IVR? Do you use the directory function for guests? Wow, what a market this could be! Bill Michaelson wrote: Anybody know how to implement a hotel wake-up call feature with *? -- Michael Welter Introspect Consulting, Inc. Denver, Colorado +1 303 674 2575 [EMAIL PROTECTED] www.introspect.com ___ 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] billing question
Arretni VoIP Tech wrote: hello, Is it normal that * starts its billing when voicemail starts to prompt? can I do something like it will only start to bill if the caller left a message? right now, im seeing that unanswered calls that are forwared to voicemail are considered billable as well as calls to voicemailmain. thanks, This should help some... bebop*CLI show application NoCDR -= Info about application 'NoCDR' =- [Synopsis]: Make sure asterisk doesn't save CDR for a certain call [Description]: NoCDR(): makes sure there won't be any CDR written for a certain call ___ 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] A working number at enum.fierymoon.com?
I'm trying to play around with ENUM, and John Todd helped me last night on the IRC channel in terms of finding this site and other docs to get me going. But now I wonder: how can I test it? I started by trying to randomly try every possible number in the US, but soon tired of that approach. . . . Does anyone know of a number that ought to return a dialplan hit from that server? Thx. B. ___ 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] A working number at enum.fierymoon.com?
Brian Capouch wrote: I'm trying to play around with ENUM, and John Todd helped me last night on the IRC channel in terms of finding this site and other docs to get me going. But now I wonder: how can I test it? I started by trying to randomly try every possible number in the US, but soon tired of that approach. My that must have been fun, at least you didn't have to attempt a real call there to test, or the Men In Black might be converging on your place as I type. Does anyone know of a number that ought to return a dialplan hit from that server? Sort of shooting from the hip here, but you could try limiting it to area code 700, the iaxtel numbers. It might be pointless, but perhaps someone is announcing an alternate means of dialing themselves there. - Andrew Thompson http://aktzero.com/ ___ 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] If one extension is busy...
Title: If one extension is busy... One of my users has two extensions, both of which ring simultaneously when a call comes in for her. This works fine. If her primary extension is busy, then she is on the phone and there is no reason to ring the secondary extension. In this case, the call should go directly to voicemail, but it in fact rings the alternate extension before going to voicemail. This gives the caller the perception that she is available, but did not answer. How can I program it to go directly to voicemail if her primary line is busy? I could start out by testing to see if the line is busy using GotoIf, if I knew how to test for busy. Alternately, I could ring her primary line for 1 second and go to voicemail if it is busy. Then I could ring both lines together for a longer period of time. However, Im afraid it might not work if she picked up just as the one second was up. Can anyone suggest a solution? Thanks. Jim
Re: [Asterisk-Users] Hotel wake-up
On Sat, 2004-02-28 at 19:39, Bill Michaelson wrote: Anybody know how to implement a hotel wake-up call feature with *? You could modify my callback script. It would require some pretty significant changes, but it's a good place to start. You can find it, and other scripts, (some good, some bad) at http://www.fnords.org/~eric/asterisk/ -- Eric Wieling [EMAIL PROTECTED] BTEL Consulting ___ 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] Hotel wake-up
Bill Michaelson wrote: Anybody know how to implement a hotel wake-up call feature with *? It seems like it could be accomplished with an AGI and a script that wrote call files. Have the AGI prompt for the wakeup time (or have a web interface for a front-desk person do it) and write a file to a directory indicating when the wakeup call should occur. Then, have a Perl script that goes through those files and generates a call file in /var/spool/asterisk/outgoing at the right time. Call files make retries simple as well, allowing you to space them and choose how many you want. If you wanted to get fancy, you could use a database (perhaps with triggers?), voice recognition, or mp3s for the user to wake up to. PlugIf this sounds too complicated, email me off list; I could write this very inexpensively for you./Plug Nick ___ 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] Load average ...
Hi Floks, I am just starting with * and while playing with the demo configuation I notice that the CPU utilization is 98-100% no matter if I am leavin a message or listening to the various voice prompts. Is this normal? The system is a P4 1.6GHz / 512MB running redhat 7.3 and kernel 2.4.22 from Fedora COre 1. Regards, -- Andrew McRory - President/CTO Linux Systems Engineers, Inc. PO BOX 3791 Tallahassee, FL 32315 (850)224-5737 (850)294-7567 ___ 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] Hotel wake-up
I would be interested in the AGI Script. As for the voice prompts, I am having Allison record some stuff for me on Monday, including prompts for such a wake up system, that I plan to donate back to the Asterisk community. This is what I have for Allison: Wake up call! This is your requested wake up call! To request a wake-up call, press 1. To confirm a wake-up call, press 2. To cancel a pending wake-up call, press 3. Enter the two digit hour of the wake up call. Enter the two digit minute of the wake up call. Press 1 for A.M. or press 2 for P.M. You have requested a wake-up call for You do not have a scheduled wake-up call. Your wake up call has been canceled. Hours must be between zero one and one two. Minutes must be between zero zero and five nine. Will these prompts be compatible with your script? Robert Rob Fugina wrote: On Sat, Feb 28, 2004 at 08:39:26PM -0500, Bill Michaelson wrote: Anybody know how to implement a hotel wake-up call feature with *? I just wrote an AGI for it. I literally just got it working the day before yesterday, so it's not really 'pretty' yet. I also don't have all of the voice prompts I need, so it's a little rough there, too. I don't have time to go into more detail at the moment, but send me a message directly if you're interested... Rob
[Asterisk-Users] Asterisk on Feebsd , pls. HELP !
Hello, Pls. help ! I have server on Freebsd 5.2 and don't may install asterisk , following errors: ( gmake clean ; gmake install ) - iasing ruleshash/ndbm.c: In function `dbm_store':hash/ndbm.c:185: warning: dereferencing type-punned pointer will break strict-aliasing ruleshash/ndbm.c:185: warning: dereferencing type-punned pointer will break strict-aliasing rulesgcc -Wall -c -D__DBINTERFACE_PRIVATE -O2 -I. -Iinclude -Ibtree -o bt_close.o btree/bt_close.cIn file included from btree/btree.h:44, from btree/bt_close.c:50:include/mpool.h:53: error: syntax error before "CIRCLEQ_ENTRY"include/mpool.h:64: error: syntax error before "CIRCLEQ_HEAD"gmake[1]: *** [bt_close.o] Error 1gmake[1]: Leaving directory `/usr/src/asterisk/db1-ast'gmake: *** [db1-ast/libdb1.a] Error 2su-2.05b#--- have any idee? Thanks, Regards, Serge.
Re: [Asterisk-Users] Hotel wake-up
All the digits should already be recorded so you could easily skip that part and play back any digit from the AGI 1-9 that it was assigned. On Sun, 2004-02-29 at 00:03, Robert Lawrence wrote: I would be interested in the AGI Script. As for the voice prompts, I am having Allison record some stuff for me on Monday, including prompts for such a wake up system, that I plan to donate back to the Asterisk community. This is what I have for Allison: Wake up call! This is your requested wake up call! To request a wake-up call, press 1. To confirm a wake-up call, press 2. To cancel a pending wake-up call, press 3. Enter the two digit hour of the wake up call. Enter the two digit minute of the wake up call. Press 1 for A.M. or press 2 for P.M. You have requested a wake-up call for You do not have a scheduled wake-up call. Your wake up call has been canceled. Hours must be between zero one and one two. Minutes must be between zero zero and five nine. Will these prompts be compatible with your script? Robert Rob Fugina wrote: On Sat, Feb 28, 2004 at 08:39:26PM -0500, Bill Michaelson wrote: Anybody know how to implement a hotel wake-up call feature with *? I just wrote an AGI for it. I literally just got it working the day before yesterday, so it's not really 'pretty' yet. I also don't have all of the voice prompts I need, so it's a little rough there, too. I don't have time to go into more detail at the moment, but send me a message directly if you're interested... Rob ___ 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] OTish: Firefly Crashing with *
You know what would be nice? If Firefly could have a Network to use assigned to a contact. I.E. I use 800 to check my voicemail at work and call work extensions etc so I have to have IAX as my internal calls...but this means I can't contact people on the firefly network... Kind regards, Matt Riddell ___ 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] OTish: Firefly Crashing with *
if u add #'s to your contact list w/ @networknameinyourclient they are connected thru that network such as firefly or others On Sun, 2004-02-29 at 15:05, asdasd wrote: You know what would be nice? If Firefly could have a Network to use assigned to a contact. I.E. I use 800 to check my voicemail at work and call work extensions etc so I have to have IAX as my internal calls...but this means I can't contact people on the firefly network... Kind regards, Matt Riddell ___ 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
Re: [Asterisk-Users] OTish: Firefly Crashing with *
sweet, cheers - Original Message - From: William Suffill [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, February 29, 2004 8:44 PM Subject: Re: [Asterisk-Users] OTish: Firefly Crashing with * if u add #'s to your contact list w/ @networknameinyourclient they are connected thru that network such as firefly or others On Sun, 2004-02-29 at 15:05, asdasd wrote: You know what would be nice? If Firefly could have a Network to use assigned to a contact. I.E. I use 800 to check my voicemail at work and call work extensions etc so I have to have IAX as my internal calls...but this means I can't contact people on the firefly network... Kind regards, Matt Riddell ___ 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 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
