[asterisk-users] Asterisk Manager Interface, reliably monitor NewCall for an extension
Hi Everyone, I am writing an open source application that brings desktops widgets to OS X (http://sourceforge.net/projects/astrxtools4osx/), for which I am trying to get my head around the Asterisk Manager Interface. I had been using the Event: NewCallerid to detect a new call which my Asterisk server doesn't seem to send to the socket anymore, because of which I have reverted to using Event: Newexten. Which is the most efficient way of monitoring if a new phone call is coming my way? Also my application will only monitor a single extension, should I filter the requests on the client side or can a manager interface user be restricted to a single extensions events. Thanks for your time. -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --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] stop log/debug messages into /var/log/messages
Hello good ppl, Any way of stopping asterisk writing into syslogs or any other file, if I set verbose 6 on the console? All I want is the verbose output only on the console, nowhere else. My logger.conf says : console= notice,error ;messages = notice,warning,error Thanks in advance. - Benjamin Jacob. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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] stop log/debug messages into /var/log/messages
Btw, even the syslog line in logger.conf is commented : ; syslog.local0 = notice,warning,error Benjamin Jacob wrote: Hello good ppl, Any way of stopping asterisk writing into syslogs or any other file, if I set verbose 6 on the console? All I want is the verbose output only on the console, nowhere else. My logger.conf says : console= notice,error ;messages = notice,warning,error Thanks in advance. - Benjamin Jacob. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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 EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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] Manager Originate without phone off hook?
Quoth Moises Silva... May be I am missing something, but, manager command DBPut should do the trick of putting the DB value. And, since you are already using the manager interface, you are using PHP or PERL to connect to the Database, why not wait for the DBPut command response and from the script execute wget?? Yes I'm using DBPut but the GUI (in tcl/tk FWIW) is running on a different network to the phones so the http request has to come from the Asterisk box and not the one running the GUI. I guess I'm going to have to write an API and call that with Originate but I just wondered if anyone had a better (read easier!) suggestion. -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ --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] Asterisk Manager Interface, reliably monitor NewCall for an extension
On 9/4/07, Devraj Mukherjee [EMAIL PROTECTED] wrote: Hi Everyone, I am writing an open source application that brings desktops widgets to OS X (http://sourceforge.net/projects/astrxtools4osx/), for which I am trying to get my head around the Asterisk Manager Interface. I had been using the Event: NewCallerid to detect a new call which my Asterisk server doesn't seem to send to the socket anymore, because of which I have reverted to using Event: Newexten. Which is the most efficient way of monitoring if a new phone call is coming my way? Also my application will only monitor a single extension, should I filter the requests on the client side or can a manager interface user be restricted to a single extensions events. I don't know about manager, but i've done the same using PHP script that executes from dialplan before dial + ActiveMQ (message queue) + custom app. I just didn't wanted to do filtering with manager, and so on.. Additionally, from my experience, creating a bunch of manager connections isn't quite good for asterisk stability.. Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.org ___ --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] stop log/debug messages into /var/log/messages
What logs are coming out to /var/log/messages? Adrian Marsh -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin Jacob Sent: 04 September 2007 07:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] stop log/debug messages into /var/log/messages Btw, even the syslog line in logger.conf is commented : ; syslog.local0 = notice,warning,error Benjamin Jacob wrote: Hello good ppl, Any way of stopping asterisk writing into syslogs or any other file, if I set verbose 6 on the console? All I want is the verbose output only on the console, nowhere else. My logger.conf says : console= notice,error ;messages = notice,warning,error Thanks in advance. - Benjamin Jacob. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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 EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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] Cisco 7960 or 7960G
On Sun, Sep 02, 2007 at 03:47:45PM +0100, Chris Bagnall wrote: There's both a 7960 and a 7960G (and a 7961 to confuse matters further). The 7960 is the earlier version. The easiest way to identify it from a picture is to look at the messages/services/etc. buttons. On the 7960 the words messages and services are written on them. On the G, there's an envelope and a globe on the buttons themselves, and the words messages and services are provided on a surround sticker (one assumes to make internationalization easier). ...although I don't think Cisco ever produced any other languages for the 7960G anyway, but 7960 and 7960G are pretty much identical. 7961 is a completely different phone with totally different software, although it has a better screen and much better audio quality than the 7960. 7960 was end-of-life a while ago by Cisco. Not sure about the 7960G though. If you run them in SIP Only mode, they are quite limited when it comes to actual functionality when compared to what other phones are offering. 7961, although a better bit of hardware, does not offer much noticable improvement for SIP. The functionality is about exactly the same, but with more possibilities for integration via XML than the 7960. 7961 does support standard 802.3af PoE and not Cisco's legacy proprietary PoE system which they introduced before 802.3af. You need a Cisco switch or a switch that supports legacy PoE (Foundry FES for example) to make the 7960s power on, but 7961 works with standard 802.3af PoE kit. Contact me off-list if you want my list of specific limitations of the 7960/SIP, as there are many. Rob -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 ___ --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] stop log/debug messages into /var/log/messages
On Tue, Sep 04, 2007 at 10:43:15AM +0100, Adrian Marsh wrote: What logs are coming out to /var/log/messages? Ask asterisk logger show channels -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] stop log/debug messages into /var/log/messages
Exactly the same lines as on the console. Adrian Marsh wrote: What logs are coming out to /var/log/messages? Adrian Marsh -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin Jacob Sent: 04 September 2007 07:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] stop log/debug messages into /var/log/messages Btw, even the syslog line in logger.conf is commented : ; syslog.local0 = notice,warning,error Benjamin Jacob wrote: Hello good ppl, Any way of stopping asterisk writing into syslogs or any other file, if I set verbose 6 on the console? All I want is the verbose output only on the console, nowhere else. My logger.conf says : console= notice,error ;messages = notice,warning,error Thanks in advance. - Benjamin Jacob. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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 EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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 EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] NAT-troubles with RTP
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Because it seems my mail from 30th august didn't make it to the list i send it again. If the mail _did_ get to the list and i didn't see it please excuse the duplicate post Below is the mail from the 30th: I have a setup like this: An asterisk-server with SIP-phones on the outside of a NAT. For example: asterisk with local IP-address (on the network interface) 1.2.3.4 and the extern IP-address should be 2.3.4.5 (yes, i'm aware these are no correct IP-addresses. It's only for describing my setup). SIP communication works perfekt. The phones are able to register on asterisk and to place calls on other SIP-phones. But when it comes to speaking, nothing works. A little bit of tcpdumping told me that the phones are trying to send the RTP traffic to 1.2.3.4 instead of 2.3.4.5 because asterisk sends 1.2.3.4 as contact-address in the SIP-header. I thought i enabled everything necessary but something must be still missing. Here is my sip.conf (with changed IP-addresses of course): [general] port = 5060 bindaddr = 0.0.0.0 context = others ;nat specific stuff below externip = 2.3.4.5 canreinvite = no nat = yes qualify = yes ;qos stuff tos_sip = cs3 tos_audio = ef ;;; ;template for voip-testing [testphones](!) context = voiptest type = friend host = dynamic nat = yes disallow = all ;disable all codecs except the ones defined below allow = ulaw allow = g723.1 allow = g729 ;;; [2000](testphones) secret = foobar [2001](testphones) secret = foobar [2002](testphones) secret = foobar Any hints on what i did wrong or forgot? Ah, and asterisk is version 1.2 on a debian etch with kernel 2.6.18 regards Florian - -- Florian Arthofer Technik Web- und Mailservices/Administrator Web- and Mailservices lagis Internet Serviceprovider GmbH Wiener Straße 151, 4021 Linz, Austria Phone +43(0)732/3400-5636 Fax +43(0)732/3400-5644 E-Mail [EMAIL PROTECTED] URL http://www.lagis.at FN 270805 w des Landesgerichtes Linz -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG3TPXNAbU6R7INwcRAlRTAJ9RyzmD8FOL2cLBC9Fp1LSE/BUgmQCaA2xt YUci7H9wuFGgDPBirAR3FHc= =xmtz -END PGP SIGNATURE- ___ --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] Asterisk Manager Interface, reliably monitor NewCall for an extension
Hi Atis, Is your code open source, or are you willing to share your PHP code snippets with me? And thanks for the information on Asterisk's stability. Do you think there is an issue in the implementation or just network/traffic issues? Thanks for your time. On 9/4/07, Atis [EMAIL PROTECTED] wrote: On 9/4/07, Devraj Mukherjee [EMAIL PROTECTED] wrote: Hi Everyone, I am writing an open source application that brings desktops widgets to OS X (http://sourceforge.net/projects/astrxtools4osx/), for which I am trying to get my head around the Asterisk Manager Interface. I had been using the Event: NewCallerid to detect a new call which my Asterisk server doesn't seem to send to the socket anymore, because of which I have reverted to using Event: Newexten. Which is the most efficient way of monitoring if a new phone call is coming my way? Also my application will only monitor a single extension, should I filter the requests on the client side or can a manager interface user be restricted to a single extensions events. I don't know about manager, but i've done the same using PHP script that executes from dialplan before dial + ActiveMQ (message queue) + custom app. I just didn't wanted to do filtering with manager, and so on.. Additionally, from my experience, creating a bunch of manager connections isn't quite good for asterisk stability.. Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.org ___ --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 -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --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] stop log/debug messages into /var/log/messages
Here it is : SIP01*CLI logger show channels Channel Type StatusConfiguration --- --- Console Enabled- Notice Error Tzafrir Cohen wrote: On Tue, Sep 04, 2007 at 10:43:15AM +0100, Adrian Marsh wrote: What logs are coming out to /var/log/messages? Ask asterisk logger show channels EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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] How can i send my sip channel 3 to mailbox 2? Please Help!
On Sun, Sep 02, 2007 at 04:03:51PM +0200, Jonathan GF wrote:
Hi folks,
i'm trying to configure my extensions.conf as small as posible and for
that reason i'm using macros. The problem is that maybe I have a
misunderstood the concept for the directive mailbox in sip.conf.
What mailbox= seems to do in sip.conf is set the message waiting indicator
(MWI) light on or off when there are messages waiting in a particular
mailbox for that extension using a SIP message to the phone to update it.
It does not control anything else such as who can access a particular
mailbox etc. just which extensions get notifications of voicemail. It is not
in voicemail.conf
I suppose because asterisk can have different channel types other than SIP,
it needs configuring for the different notification methods depending on
devices. (i.e, voicemail app doesn't want to be getting involved in how to
set and unset MWI for all sorts of different channel types.)
What i'm trying is to have ONLY 2 voicemail boxes and depending which
extensions i'm dialing send the caller to one or the other, but not
send based on the called id/name, but to that mailbox i want (mailbox
1 or mailbox 2, just this).
The error i'm getting is:
WARNING[2102]: app_voicemail.c:2461 leave_voicemail: No entry in
voicemail config file for '3'
The error is correct... i don't have a voicemail box named/numbered
3 but this is the behavior i want to control. How can i send my sip
channel 3 to mailbox 2?
Essentially, you need to pass the mailbox you want to access to the
voicemail app, which is the thing in ARG1 in your macro.
So, Voicemail(u2) would play the unavailable message for mailbox 2 instead
of what you are currently passing it, which appears to be ${EXTEN}, the
dialled extension.
What you can do is check to see if a voicemail mailbox exists for a
particular extension before you try it, and if no mailbox exists (i.e, you
have not configured it in voicemail.conf) then you can do something else.)
Something like this will check to see if a mailbox exists before trying it,
if not then default to mailbox 2:
exten = s,1,MailboxExists(${ARG1},j)
exten = s,2,Voicemail(u2)
exten = s,3,Hangup
exten = s,102,Voicemail(u${ARG1})
exten = s,103,Hangup
Note, you can also check the variable ${VMBOXEXISTSSTATUS} for one of
SUCCESS or FAILED if you don't like the old style priority jumping,
which can get a bit awkward if you have to renumber things, this is the
'newer' way to do it, something like:-
exten = s,1,MailboxExists(${ARG1},j)
exten = s,2,Goto(s-${VMBOXEXISTSSTATUS},1}
exten = s-FAILED,Voicemail(u2)
exten = s-FAILED,Hangup
exten = s-SUCCESS,Voicemail(u${ARG1})
exten = s-SUCCESS,Hangup
Of course, how you work out when somebody accesses your voicemail to listen
to messages depends on how you are authenticating them into voicemail in the
first place. You might just prompt for the mailbox number and/or PIN, or you
can drop them straight into the right mailbox using a similar technique.
If it gets more exotic than your two mailboxes, then you could use astdb
entries to work out which mailbox is associated with a particular extension,
which is more elaborate but might be worth doing for ease of configuration.
(In that you are not hardcoding stuff into extensions.conf for every
extension)
astdb is asterisk's builtin database, which is really handy for this kind of
thing (Unless you have millions of mailboxes which is an entirely different
database proposition!)
$ asterisk -r
asterisk*CLI database put 3 mailbox 2
asterisk*CLI database show 3
/3/mailbox : 2
(That is to say, for the extension 3, we want mailbox 2)
Then, to see that db variable in where you need it in the dialplan,
would look like this:-
${DB(${EXTEN}/mailbox)} (where ${EXTEN} is 3, this would return 2)
or ${DB(${ARG1}/mailbox)} in the case of your macro.
This will look in the astdb for that mailbox variable you set up and
use that instead of hardcoding it into the dialplan.
Suppose your voicemail access extension is 444 and you want a passwordless
login from the extension based on what you have set in the astdb for that
extension, based on caller ID of the incoming extension:-
; passwordless login
exten = 444,1,VoiceMailMain(${DB(${CALLERID(num)}/mailbox)}|s)
exten = 444,n,Hangup
(Yes, I know, it's a bit fugly bracket hell, but it's worth it!)
You could combine this of course with MailboxExists to drop them into some
default mailbox, or prompt for a mailbox number, or if there is a mailbox
for that extension and no translation is required. (i.e, Do a MailboxExists
and then decide if the ${DB lookup is needed.)
This is just an example of DB lookups, you could do a similar thing for
determining which mailbox to drop callers in to as well as for mailbox
access.
Then for all future requirements, you just add that to your astdb as you want
them and it will take care of it for you. No tweaking of
Re: [asterisk-users] off-hook warning tone
On Sun, Sep 02, 2007 at 05:15:41PM -0500, Anthony Messina wrote: is asterisk capable of generating the off-hook warning tone for the us? 1400+2060+2450+2600/100,0/100 i have placed it into indications.conf, but all i get is one high-pitched screech instead of alternating tones. I am thinking this might be handset specific thing, as unless you dial something the call is not going to be placed to asterisk yet, unless you can somehow first Answer() the call after some timeout (i.e, if the handset has a hotline extn config to dial after N seconds of no digits being dialed - some handsets support that functionality) Rob -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 pgpR7r3AeW3j9.pgp Description: PGP signature ___ --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] off-hook warning tone
On Tuesday 04 September 2007 06:54:55 am Robert Lister wrote: On Sun, Sep 02, 2007 at 05:15:41PM -0500, Anthony Messina wrote: is asterisk capable of generating the off-hook warning tone for the us? 1400+2060+2450+2600/100,0/100 i have placed it into indications.conf, but all i get is one high-pitched screech instead of alternating tones. I am thinking this might be handset specific thing, as unless you dial something the call is not going to be placed to asterisk yet, unless you can somehow first Answer() the call after some timeout (i.e, if the handset has a hotline extn config to dial after N seconds of no digits being dialed - some handsets support that functionality) Rob well i'm looking for the feature that the telco provides where, if you've left the phone off-hook for 60 seconds or so without input, it gives you the loud put the damn phone back on the hook noise. it works if i set absolute timeout to 60 and use the congestion tone, but i was hoping to use the actual off-hook warning tone. it seems as if the tone itself is not generated properly within asterisk. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ --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] How can i send my sip channel 3 to mailbox 2? Please Help!
Hey Robert,
you can't imagine how much i appreciate your post, which is most a
tutorial than a post :)
Really, many thanks for your thoughts. Take for sure i will try to
implement the options you showed me here in asap.
Thank you again!
Best regards,
Jonathan GF
On 9/4/07, Robert Lister [EMAIL PROTECTED] wrote:
On Sun, Sep 02, 2007 at 04:03:51PM +0200, Jonathan GF wrote:
Hi folks,
i'm trying to configure my extensions.conf as small as posible and for
that reason i'm using macros. The problem is that maybe I have a
misunderstood the concept for the directive mailbox in sip.conf.
What mailbox= seems to do in sip.conf is set the message waiting indicator
(MWI) light on or off when there are messages waiting in a particular
mailbox for that extension using a SIP message to the phone to update it.
It does not control anything else such as who can access a particular
mailbox etc. just which extensions get notifications of voicemail. It is not
in voicemail.conf
I suppose because asterisk can have different channel types other than SIP,
it needs configuring for the different notification methods depending on
devices. (i.e, voicemail app doesn't want to be getting involved in how to
set and unset MWI for all sorts of different channel types.)
What i'm trying is to have ONLY 2 voicemail boxes and depending which
extensions i'm dialing send the caller to one or the other, but not
send based on the called id/name, but to that mailbox i want (mailbox
1 or mailbox 2, just this).
The error i'm getting is:
WARNING[2102]: app_voicemail.c:2461 leave_voicemail: No entry in
voicemail config file for '3'
The error is correct... i don't have a voicemail box named/numbered
3 but this is the behavior i want to control. How can i send my sip
channel 3 to mailbox 2?
Essentially, you need to pass the mailbox you want to access to the
voicemail app, which is the thing in ARG1 in your macro.
So, Voicemail(u2) would play the unavailable message for mailbox 2 instead
of what you are currently passing it, which appears to be ${EXTEN}, the
dialled extension.
What you can do is check to see if a voicemail mailbox exists for a
particular extension before you try it, and if no mailbox exists (i.e, you
have not configured it in voicemail.conf) then you can do something else.)
Something like this will check to see if a mailbox exists before trying it,
if not then default to mailbox 2:
exten = s,1,MailboxExists(${ARG1},j)
exten = s,2,Voicemail(u2)
exten = s,3,Hangup
exten = s,102,Voicemail(u${ARG1})
exten = s,103,Hangup
Note, you can also check the variable ${VMBOXEXISTSSTATUS} for one of
SUCCESS or FAILED if you don't like the old style priority jumping,
which can get a bit awkward if you have to renumber things, this is the
'newer' way to do it, something like:-
exten = s,1,MailboxExists(${ARG1},j)
exten = s,2,Goto(s-${VMBOXEXISTSSTATUS},1}
exten = s-FAILED,Voicemail(u2)
exten = s-FAILED,Hangup
exten = s-SUCCESS,Voicemail(u${ARG1})
exten = s-SUCCESS,Hangup
Of course, how you work out when somebody accesses your voicemail to listen
to messages depends on how you are authenticating them into voicemail in the
first place. You might just prompt for the mailbox number and/or PIN, or you
can drop them straight into the right mailbox using a similar technique.
If it gets more exotic than your two mailboxes, then you could use astdb
entries to work out which mailbox is associated with a particular extension,
which is more elaborate but might be worth doing for ease of configuration.
(In that you are not hardcoding stuff into extensions.conf for every
extension)
astdb is asterisk's builtin database, which is really handy for this kind of
thing (Unless you have millions of mailboxes which is an entirely different
database proposition!)
$ asterisk -r
asterisk*CLI database put 3 mailbox 2
asterisk*CLI database show 3
/3/mailbox : 2
(That is to say, for the extension 3, we want mailbox 2)
Then, to see that db variable in where you need it in the dialplan,
would look like this:-
${DB(${EXTEN}/mailbox)} (where ${EXTEN} is 3, this would return 2)
or ${DB(${ARG1}/mailbox)} in the case of your macro.
This will look in the astdb for that mailbox variable you set up and
use that instead of hardcoding it into the dialplan.
Suppose your voicemail access extension is 444 and you want a passwordless
login from the extension based on what you have set in the astdb for that
extension, based on caller ID of the incoming extension:-
; passwordless login
exten = 444,1,VoiceMailMain(${DB(${CALLERID(num)}/mailbox)}|s)
exten = 444,n,Hangup
(Yes, I know, it's a bit fugly bracket hell, but it's worth it!)
You could combine this of course with MailboxExists to drop them into some
default mailbox, or prompt for a mailbox number, or if there is a mailbox
for that extension and no translation is required.
[asterisk-users] SIPBroker vs SIPgate
All, I've been experimenting with shortcodes for SIPgate etc. Passing calls to SIPbroker seems a good way to go, but the message I've had back from SIPgate is we don't support SIPBroker... So whats the easiest way to support SIP SIP network calling? At the moment, I've setup some local shortcodes (eg dial **777. to goto sipgate.co.uk) based on what Gradwell have publically posted, but I can't even get SIPgate to work with this either !! (Can't pass these directly to Gradwell as their SIP trunks don't support it..) A. ___ --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] stop log/debug messages into /var/log/messages
When you access the A*k console, is this via a tty connection (ssh/telnet), or actually on the physical console of the server? I don't think it's A*k that's directly logging to the console - the config doesn't show that... I'm guessing, that you're accessing A*k via the local terminal, and that your syslog config for the server is configured to log this to messsages Maybe.. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin Jacob Sent: 04 September 2007 12:51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] stop log/debug messages into /var/log/messages Here it is : SIP01*CLI logger show channels Channel Type StatusConfiguration --- --- Console Enabled- Notice Error Tzafrir Cohen wrote: On Tue, Sep 04, 2007 at 10:43:15AM +0100, Adrian Marsh wrote: What logs are coming out to /var/log/messages? Ask asterisk logger show channels EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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] SIPBroker vs SIPgate
Adrian Marsh wrote: All, I've been experimenting with shortcodes for SIPgate etc. Passing calls to SIPbroker seems a good way to go, but the message I've had back from SIPgate is we don't support SIPBroker... So whats the easiest way to support SIP SIP network calling? At the moment, I've setup some local shortcodes (eg dial **777. to goto sipgate.co.uk) based on what Gradwell have publically posted, but I can't even get SIPgate to work with this either !! (Can't pass these directly to Gradwell as their SIP trunks don't support it..) A. ___ --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 SIP - SIP calling across networks really only works if the receiving network allows incoming calls from non-local networks. SIPgate does not, so unless you're registered on the SIPgate network, calling another SIPgate user from your SIPgate number, it won't accept the call. N. ___ --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] stop log/debug messages into /var/log/messages
On Tue, Sep 04, 2007 at 12:17:02PM +0530, Benjamin Jacob wrote: Hello good ppl, Any way of stopping asterisk writing into syslogs or any other file, if I set verbose 6 on the console? All I want is the verbose output only on the console, nowhere else. My logger.conf says : console= notice,error ;messages = notice,warning,error A ';' means that this is a comment. That line is ignored. The sample logger.conf is well-commented: http://svn.digium.com/svn/asterisk/branches/1.4/configs/logger.conf.sample After editing it, run: logger reload from the asterisk CLI To see the current configuration: logger show channels The full line in the sample file (once you remove the comment) is probably what you want. Naturally you can call it in any name you want. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] E1 to Ethernet Bridge
On 9/3/07, Arinze Izukanne [EMAIL PROTECTED] wrote: Can you show me a sample fo config? The link schematic should look like this: E1 == TDMoE==E1. Refer to the section Sample configs for setting up TDMoE between 2 servers without TDM hardware, using ztdummy on this page: http://www.voip-info.org/wiki/index.php?page=Asterisk+TDMoE The examples are for T1, so you'll have to change the number of channels on each span and change the channel numbers used for bchan= and dchan=, but if you're familiar with E1 deployment already this should be simple. -- j. ___ --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] SIPBroker vs SIPgate
Yeah, I can see that now after testing it all - but this is what raised my question.. What IS the best mechanism for all the VoIP servers/networks to interact ? Setting up individual agreements for each network is so 1980's, and in this modern world there must be a better solution.. A. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of SIP Sent: 04 September 2007 15:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIPBroker vs SIPgate Adrian Marsh wrote: All, I've been experimenting with shortcodes for SIPgate etc. Passing calls to SIPbroker seems a good way to go, but the message I've had back from SIPgate is we don't support SIPBroker... So whats the easiest way to support SIP SIP network calling? At the moment, I've setup some local shortcodes (eg dial **777. to goto sipgate.co.uk) based on what Gradwell have publically posted, but I can't even get SIPgate to work with this either !! (Can't pass these directly to Gradwell as their SIP trunks don't support it..) A. ___ --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 SIP - SIP calling across networks really only works if the receiving network allows incoming calls from non-local networks. SIPgate does not, so unless you're registered on the SIPgate network, calling another SIPgate user from your SIPgate number, it won't accept the call. N. ___ --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] Asterisk Died message
Hello All, Anyone knows what does this error message means and where to check for the cause and why it happened? Asterisk on hyperion exited on signal 11. Might want to take a peek. But when I check Asterisk, its running fine... Cheers, Nitesh ___ --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] Manager Originate without phone off hook?
To answer my own question I found a way to acheive what I wanted so
here's my solution for the record (might help someone else if they
search the archives).
In the Dialplan setup the following entries:
[snom_setdndon]
exten = _.,1,NoOp(Dummy Routine Called for ${EXTEN})
exten = _.,n,TrySystem(wget -qb -O /dev/null -o /dev/null
http://admin:[EMAIL PROTECTED]/dummy.htm?settings=savednd_mode=on)
exten = _.,n,Hangup
[snom_setdndoff]
exten = _.,1,NoOp(Dummy Routine Called for ${EXTEN})
exten = _.,n,TrySystem(wget -qb -O /dev/null -o /dev/null
http://admin:[EMAIL PROTECTED]/dummy.htm?settings=savednd_mode=off)
exten = _.,n,Hangup
and then from the manager interface one can do:
Action: Originate
Channel: Local/[EMAIL PROTECTED]
Application: NoOp
Data: Setting DND
A bit convoluted but it works for me.
--
Regards,
Russell
| Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com |
| Peterborough, England | WWW Play: http://www.ruffle.me.uk |
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[asterisk-users] Cisco 79xx XML Apps (was: Re: Cisco Directory Format)
Do you know where to find clear developers' guides (with some examples) for developing apps that run *on* Cisco 79xx phones (especially the 7970)? Examples that can run against Asterisk (not CallManager) with SIP firmware (not SCCP), and/or LDAP directories (or other open servers) would be best. On Sat, 2007-09-01 at 12:00 -0500, [EMAIL PROTECTED] wrote: Date: Sat, 1 Sep 2007 12:14:49 -0400 From: Time Bandit [EMAIL PROTECTED] Subject: Re: [asterisk-users] Cisco Directory Format To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 A little off topic (sorry..:) ) but anyone know what format Cisco phones use for their contact dirctories. I want to set up my contact lists on the phone, and cannot seem to get any info on it. I am working with a 7970 on Asterisk 1.4.8. 7940 and 7960 use this format of XML file (probably the same on 7970) CiscoIPPhoneDirectory TitleEmployee directory/Title PromptOpen Source Rock/Prompt DirectoryEntry NameEmployee A/Name Telephone7001/Telephone /DirectoryEntry DirectoryEntry NameEmployee B/Name Telephone7002/Telephone /DirectoryEntry /CiscoIPPhoneDirectory Check also Open 79XX XML Directory : http://web.csma.biz/apps/xml_xmldir.php hope that help -- (C) Matthew Rubenstein ___ --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] Asterisk Died message
Signal 11 is a segmentation fault, if you are not running unsupported patches on Asterisk you should compile without Asterisk optimizations and open a bug attaching the debugging backtrace. Read This: http://www.voip-info.org/wiki-Asterisk+debugging Regards, On 9/4/07, Nitesh Divecha [EMAIL PROTECTED] wrote: Hello All, Anyone knows what does this error message means and where to check for the cause and why it happened? Asterisk on hyperion exited on signal 11. Might want to take a peek. But when I check Asterisk, its running fine... Cheers, Nitesh ___ --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 -- Within C++, there is a much smaller and cleaner language struggling to get out. ___ --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] Asterisk w/MS SQL Server 2005
I've got an Asterisk switch that is going to run an IVR menu with a database interface that will be doing lookups based on the user entered data and then reading back strings with the appropriate data integrated into the text. I have found quite a bit of data on using MySQL as a database with Asterisk, but I haven't found much about using MS SQL Server with Asterisk... We have a SQL Server 2005 database server that has all the data that is needed for the IVR interface, and it would be great if we could interface directly with it using Asterisk. Does anyone have any suggestions on even attepting this? Another option might be to setup Asterisk to interface with MySQL and then work out the details of exchanging data between MySQL and SQL Server... Any and all help is greatly appreciated!! -Larry ___ --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] Wireless VOIP Keysets? Recommendations?
I haven't come across any wireless devices that support IAX2, but we have successfully used the Linksys WIP300, Linksys WIP330, Nokia N80, Nokia E61i, and Nokia N95 with asterisk. If you just need wireless and not mobility, the Linksys WBP54G also works well to interconnect Ethernet based VoIP phones. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of William Stillwell (Ki4swy) Sent: Sunday, September 02, 2007 5:45 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Wireless VOIP Keysets? Recommendations? Any Recommendations on a Good Wireless Voip Keyset that works well with Asterisk? I would prefer one that is IAX2 as it works better behind a Nat'd Firewall.. But I am reaching out to you guys as you all would know what would work the best :-) Sent via the WebMail system at kotbh.net ___ --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] Asterisk w/MS SQL Server 2005
You can use Asterisk's AGI and PHP/Perl or whatever else. You'll need to install connecting software, such as FreeTDS, to connect to SQL. Then you can either pass arguments to your script or use environment variables to set Asterisk variables. Here's a good place to start: http://www.voip-info.org/wiki-Asterisk+AGI Larry Costigan wrote: I've got an Asterisk switch that is going to run an IVR menu with a database interface that will be doing lookups based on the user entered data and then reading back strings with the appropriate data integrated into the text. I have found quite a bit of data on using MySQL as a database with Asterisk, but I haven't found much about using MS SQL Server with Asterisk... We have a SQL Server 2005 database server that has all the data that is needed for the IVR interface, and it would be great if we could interface directly with it using Asterisk. Does anyone have any suggestions on even attepting this? Another option might be to setup Asterisk to interface with MySQL and then work out the details of exchanging data between MySQL and SQL Server... Any and all help is greatly appreciated!! -Larry ___ --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] SIPBroker vs SIPgate
Seriously, from our experience, SIPBroker IS the best way to interact with all the open networks. For any closed networks, you might create special rules for interaction, but that would rely on setting up a deal with the respective destination network to actually ALLOW your calls. There are some pay per play networks that do peering automagically (such as XConnect), but it's a cost per connected call (granted, a tiny one, but still a cost), and it won't guarantee you any better connectivity to a closed network than, say, SIPBroker. N. Adrian Marsh wrote: Yeah, I can see that now after testing it all - but this is what raised my question.. What IS the best mechanism for all the VoIP servers/networks to interact ? Setting up individual agreements for each network is so 1980's, and in this modern world there must be a better solution.. A. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of SIP Sent: 04 September 2007 15:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIPBroker vs SIPgate Adrian Marsh wrote: All, I've been experimenting with shortcodes for SIPgate etc. Passing calls to SIPbroker seems a good way to go, but the message I've had back from SIPgate is we don't support SIPBroker... So whats the easiest way to support SIP SIP network calling? At the moment, I've setup some local shortcodes (eg dial **777. to goto sipgate.co.uk) based on what Gradwell have publically posted, but I can't even get SIPgate to work with this either !! (Can't pass these directly to Gradwell as their SIP trunks don't support it..) A. ___ --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 SIP - SIP calling across networks really only works if the receiving network allows incoming calls from non-local networks. SIPgate does not, so unless you're registered on the SIPgate network, calling another SIPgate user from your SIPgate number, it won't accept the call. N. ___ --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 ___ --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] Asterisk w/MS SQL Server 2005
On 9/4/07, Larry Costigan [EMAIL PROTECTED] wrote: I've got an Asterisk switch that is going to run an IVR menu with a database interface that will be doing lookups based on the user entered data and then reading back strings with the appropriate data integrated into the text. I have found quite a bit of data on using MySQL as a database with Asterisk, but I haven't found much about using MS SQL Server with Asterisk... We have a SQL Server 2005 database server that has all the data that is needed for the IVR interface, and it would be great if we could interface directly with it using Asterisk. Does anyone have any suggestions on even attepting this? Another option might be to setup Asterisk to interface with MySQL and then work out the details of exchanging data between MySQL and SQL Server... Any and all help is greatly appreciated!! You'll want to take a look at func_odbc which should probably give you a pretty good conduit to get the data you need to work through the IVR flow. -- Bird's The Word Technologies, Inc. http://www.btwtech.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
Re: [asterisk-users] off-hook warning tone
On Tue, Sep 04, 2007 at 07:41:53AM -0500, Anthony Messina wrote: well i'm looking for the feature that the telco provides where, if you've left the phone off-hook for 60 seconds or so without input, it gives you the loud put the damn phone back on the hook noise. it works if i set absolute timeout to 60 and use the congestion tone, but i was hoping to use the actual off-hook warning tone. it seems as if the tone itself is not generated properly within asterisk. Curious as I have not had problems with generating the tones. It's worth checking that in sip.conf the language= option is set to the same section you are editing in indications.conf In the dialplan, what I think should happen is that when you do: Congestion() You send a congestion message back to the phone using SIP (rather than in-band audio) so the handset is probably generating the Congestion tone, not asterisk as it is not yet in the media path. If you did it inband audio:- Answer() Playtones(congestion) This would play the tone from indications.conf - have an experiment with this by setting up a little extension and dialling it. As far as I can tell, AbsoluteTimeout() is just a global timeout for the duration of a call, so if you set it to AbsoluteTimeout(30) then the call (any call) will be hung up after 30 seconds. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AbsoluteTimeout Some handsets allow you to customise the tones played - depends on the handset. And some handsets have a hotline feature to dial a given extension after no digits have been dial for N seconds. (So you could get the handset to dial a special extension which then answers the channel and plays the noise you want!) I could be wrong of course. Never wanted to do this as our phones just seem to go back on-hook regardless after some dial timeout has elapsed. Rob -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 pgplaQjJlUl1F.pgp Description: PGP signature ___ --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] Udev issue on zaptel install
While attempting to install zaptel I received the following output in response to make install: ... Install -d /etc/udev/rules.d Build_tools /genudevrules /etc/udev/rules.d/zaptel.rules Build_tools /genudevrules :line 1: udevinfo : command not found Make: *** [devices] error 1 And the install aborted. Debian kernel 2.6.17.8-686 Zaptel version 1.4.4 Any ideas? Thanks in advance! Craig smime.p7s Description: S/MIME cryptographic signature ___ --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] off-hook warning tone
The correct term for this tone is howler. I'm surprised it is not in indications.conf Robert Lister wrote: On Tue, Sep 04, 2007 at 07:41:53AM -0500, Anthony Messina wrote: well i'm looking for the feature that the telco provides where, if you've left the phone off-hook for 60 seconds or so without input, it gives you the loud put the damn phone back on the hook noise. it works if i set absolute timeout to 60 and use the congestion tone, but i was hoping to use the actual off-hook warning tone. it seems as if the tone itself is not generated properly within asterisk. Curious as I have not had problems with generating the tones. It's worth checking that in sip.conf the language= option is set to the same section you are editing in indications.conf In the dialplan, what I think should happen is that when you do: Congestion() You send a congestion message back to the phone using SIP (rather than in-band audio) so the handset is probably generating the Congestion tone, not asterisk as it is not yet in the media path. If you did it inband audio:- Answer() Playtones(congestion) This would play the tone from indications.conf - have an experiment with this by setting up a little extension and dialling it. As far as I can tell, AbsoluteTimeout() is just a global timeout for the duration of a call, so if you set it to AbsoluteTimeout(30) then the call (any call) will be hung up after 30 seconds. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AbsoluteTimeout Some handsets allow you to customise the tones played - depends on the handset. And some handsets have a hotline feature to dial a given extension after no digits have been dial for N seconds. (So you could get the handset to dial a special extension which then answers the channel and plays the noise you want!) I could be wrong of course. Never wanted to do this as our phones just seem to go back on-hook regardless after some dial timeout has elapsed. ___ --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] off-hook warning tone
On Tuesday 04 September 2007 03:24:59 pm Robert Lister wrote: On Tue, Sep 04, 2007 at 07:41:53AM -0500, Anthony Messina wrote: well i'm looking for the feature that the telco provides where, if you've left the phone off-hook for 60 seconds or so without input, it gives you the loud put the damn phone back on the hook noise. it works if i set absolute timeout to 60 and use the congestion tone, but i was hoping to use the actual off-hook warning tone. it seems as if the tone itself is not generated properly within asterisk. Curious as I have not had problems with generating the tones. It's worth checking that in sip.conf the language= option is set to the same section you are editing in indications.conf In the dialplan, what I think should happen is that when you do: Congestion() You send a congestion message back to the phone using SIP (rather than in-band audio) so the handset is probably generating the Congestion tone, not asterisk as it is not yet in the media path. If you did it inband audio:- Answer() Playtones(congestion) This would play the tone from indications.conf - have an experiment with this by setting up a little extension and dialling it. As far as I can tell, AbsoluteTimeout() is just a global timeout for the duration of a call, so if you set it to AbsoluteTimeout(30) then the call (any call) will be hung up after 30 seconds. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AbsoluteTimeout Some handsets allow you to customise the tones played - depends on the handset. And some handsets have a hotline feature to dial a given extension after no digits have been dial for N seconds. (So you could get the handset to dial a special extension which then answers the channel and plays the noise you want!) I could be wrong of course. Never wanted to do this as our phones just seem to go back on-hook regardless after some dial timeout has elapsed. Rob thank you for your help. if you define 1400+2060+2450+2600/100,0/100 as a tone in indications.conf, then set up a test extension to play that tone, are you able to hear the same tone generated by the attached mp3 file? i have not been able to replicate that tone (that i got from http://www.3amsystems.com/wireline/tone-search.htm?start=20kCountry=184format=Zaptel ) i can play all the other default tones in indications.conf properly. a quote from indications.conf: ; The frequency component may be a mixture of two ; frequencies (f1+f2) or a frequency modulated by another frequency (f1*f2). my only guess as to why the tone i'm trying to create won't work is because the first part is a mixture of 4 tones (1400+2060+2450+2600) and maybe asterisk won't generate that, though i'm not sure. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E 17-22.mp3 Description: audio/mp3 signature.asc Description: This is a digitally signed message part. ___ --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] off-hook warning tone
On Tue, Sep 04, 2007 at 03:44:46PM -0500, Eric ManxPower Wieling wrote: The correct term for this tone is howler. I'm surprised it is not in indications.conf There are two sorts. The 'howler' is usually a single frequency tone -- 1500ish Hz, I think -- that gets progressively louder over time. The other one, called a Screaming Mimi where I grew up, is a four-frequency tone interrupted at 120ipm (I think, might be 240... turns out it's 300). Both are *much* louder than anything else that ever goes over a copper POTS circuit; loud enough that they'll actually cause crosstalk in some cabling. http://en.wikipedia.org/wiki/Off_hook_tone http://en.wikipedia.org/wiki/Permanent_signal Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ --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] Overhead paging over IP...
I have a customer that has two buildings that are connected with a fiber link. We have a single Asterisk server to cover both buildings. Now the customer went and bought an overhead paging system for the remote building and they want to integrate it with Asterisk. Is there a device that can connect over IP or an ATA that has an audio output port? The buildings are about 500 meters apart so we cannot run a cable from one building to the other just for audio. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --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] off-hook warning tone
Anthony, Robert here is mentioning a SIP phone, but I didn't see you specify what kind of phone you have. Is this accurate, or is it a Zaptel FXS port? Moj Robert Lister wrote: On Tue, Sep 04, 2007 at 07:41:53AM -0500, Anthony Messina wrote: well i'm looking for the feature that the telco provides where, if you've left the phone off-hook for 60 seconds or so without input, it gives you the loud put the damn phone back on the hook noise. it works if i set absolute timeout to 60 and use the congestion tone, but i was hoping to use the actual off-hook warning tone. it seems as if the tone itself is not generated properly within asterisk. Curious as I have not had problems with generating the tones. It's worth checking that in sip.conf the language= option is set to the same section you are editing in indications.conf In the dialplan, what I think should happen is that when you do: Congestion() You send a congestion message back to the phone using SIP (rather than in-band audio) so the handset is probably generating the Congestion tone, not asterisk as it is not yet in the media path. If you did it inband audio:- Answer() Playtones(congestion) This would play the tone from indications.conf - have an experiment with this by setting up a little extension and dialling it. As far as I can tell, AbsoluteTimeout() is just a global timeout for the duration of a call, so if you set it to AbsoluteTimeout(30) then the call (any call) will be hung up after 30 seconds. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AbsoluteTimeout Some handsets allow you to customise the tones played - depends on the handset. And some handsets have a hotline feature to dial a given extension after no digits have been dial for N seconds. (So you could get the handset to dial a special extension which then answers the channel and plays the noise you want!) I could be wrong of course. Never wanted to do this as our phones just seem to go back on-hook regardless after some dial timeout has elapsed. Rob ___ --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] Escape characer for Digit Timeout
For me as well, if I append a #, asterisk tries to match it to an extension and I get congestion indication when it fails to do that. This would be nice though! (TDM card too) Moj Giuffredi wrote: If I press # I get “incorrect number” as asterisk passes to the telco the numbers and the character. Any way to exclude it? Thanx a lot. Date: Sat, 1 Sep 2007 23:36:35 -0400 From: C F [EMAIL PROTECTED] The # (pound or hash) should do that. On 9/1/07, Giuffredi [EMAIL PROTECTED] wrote: I need to set an escape character to stop digits timeout to let phone calls start immediately. Something like '*'. I saw that in many SIP phones it is possible to set timeout and escape character and the most have a send button. But in an analog phone connected to a Digium TDM? ___ --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] Overhead paging over IP...
Carlos Chavez wrote: I have a customer that has two buildings that are connected with a fiber link. We have a single Asterisk server to cover both buildings. Now the customer went and bought an overhead paging system for the remote building and they want to integrate it with Asterisk. Is there a device that can connect over IP or an ATA that has an audio output port? The buildings are about 500 meters apart so we cannot run a cable from one building to the other just for audio. I think there are IP speakers out there but if you already have a Bogen or something you may want to go a different route. Polycom phones have a paging feature which is basically just auto answer on speakerphone. I suppose it would be quite easy to wire up the headphone jack on the Polycom to the Bogen (or whatever). A Polycom 301 should work just fine. If I were you, I would try that (but I have two or three 301s just laying around...) Thanks, Steve Totaro ___ --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] Overhead paging over IP...
Something like this should work http://www.voipsupply.com/product_info.php?products_id=3517osCsid=85766c1b3901219249d6fdea9bc7b7c0 On 9/4/07, Carlos Chavez [EMAIL PROTECTED] wrote: I have a customer that has two buildings that are connected with a fiber link. We have a single Asterisk server to cover both buildings. Now the customer went and bought an overhead paging system for the remote building and they want to integrate it with Asterisk. Is there a device that can connect over IP or an ATA that has an audio output port? The buildings are about 500 meters apart so we cannot run a cable from one building to the other just for audio. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ --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 -- /* Andrew Latham LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] [EMAIL PROTECTED] */ ___ --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] Udev issue on zaptel install
On Tue, Sep 04, 2007 at 01:39:55PM -0700, Markham, Craig (FRTC Contractor) wrote: While attempting to install zaptel I received the following output in response to make install: ... Install -d /etc/udev/rules.d Build_tools /genudevrules /etc/udev/rules.d/zaptel.rules Build_tools /genudevrules :line 1: udevinfo : command not found Make: *** [devices] error 1 And the install aborted. Debian kernel 2.6.17.8-686 Zaptel version 1.4.4 Any ideas? Thanks in advance! Craig What Linux distribution is this? Is udev actually in use? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Overhead paging over IP...
You can use an inexpensive PC with a sound card. Install Asterisk on it and set an extension that calls /dev/dsp. This will send audio out the speaker port on the sound card. Setup a trunk between this unit and your primary Asterisk server and you should be in business. Bryan M. Johns Shelton | Johns Technology Group Office: 678.248.2637 FindMe: 678.229.1809 http://www.sheltonjohns.com On Sep 4, 2007, at 7:07 PM, Carlos Chavez wrote: I have a customer that has two buildings that are connected with a fiber link. We have a single Asterisk server to cover both buildings. Now the customer went and bought an overhead paging system for the remote building and they want to integrate it with Asterisk. Is there a device that can connect over IP or an ATA that has an audio output port? The buildings are about 500 meters apart so we cannot run a cable from one building to the other just for audio. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ --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] Overhead paging over IP...
Most of the overhead paging systems I have worked with had an CO input, which worked with the FXO port on an ATA. A couple brands had multiple source options, it is worth checking, I had problems with poor audio quality using the sound card with asterisk. On 9/4/07, Carlos Chavez [EMAIL PROTECTED] wrote: I have a customer that has two buildings that are connected with a fiber link. We have a single Asterisk server to cover both buildings. Now the customer went and bought an overhead paging system for the remote building and they want to integrate it with Asterisk. Is there a device that can connect over IP or an ATA that has an audio output port? The buildings are about 500 meters apart so we cannot run a cable from one building to the other just for audio. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ --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 -- Bruce Reeves Nortex Networks ___ --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] Tone detection while Dialing
Hi, I want to detect a tone while Dial() through pri. When a secial tone(eg, #), I want to send the call to another extenison. Regards. Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos more. http://mobile.yahoo.com/go?refer=1GNXIC ___ --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] Overhead paging over IP...
Bruce Reeves wrote: multiple source options, it is worth checking, I had problems with poor audio quality using the sound card with asterisk. I did as well using the built-in sound on the motherboard that I was using, switched to a Sound Blaster Live Value card and that problem went away. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] How to make call from asterisk?
Hi, I'm new to asterisk, in order to enable X-lite to make a call, what should i do before making a call? Current stage, 1. i have create a few accounts in sip.conf. 2. Registration are successful. Pls advice me how to continue then... Thanks _ Personalize your Live.com homepage with the news, weather, and photos you care about. http://www.live.com/getstarted.aspx?icid=T001MSN30A0701___ --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] about ChanSpy,thank you!!!
asterisk-usershello,everyone! I was setting up ChanSpy in an Asterisk dialplan today and it just wasn't working. Here is the snippet: extensions.conf: [test] exten=3001, 1, Set(__TRANSFER_CONTEXT=tranfer) exten=3001, 2, Dial(SIP/3001,10,tr) exten=2002, 1, Dial(SIP/2002,10,t) exten=2002, 2, Hangup() exten=2001, 1, Dial(SIP/2001,10,t) exten=2001, 2, Hangup() exten=1234, 1, ChanSpy(SIP,b) [tranfer] exten= 2002, 1,Dial(SIP/2002,10,tr) first,i use 2001 dialed 3001,connected! then use 3001 transfer to 2002,connected!and then i use 3001 dialed 1234 to spy 2001-2002,Asterisk tell me i was spying 2001,afeter it ,i would hear nothing at all. if i used 2001 dialed 2002,connected! then i used 3001 dialed 1234 to spy 2001-2002,it was success!!! why? thanks you weizhao_hou [EMAIL PROTECTED] 2007-09-05 ___ --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] CID debugging help needed
I have a TDM800P with 5 POTS lines hooked into my university's PBX. CID information is being picked up properly from phones inside the PBX, but not from phones outside. I've tried mucking with rxgain with no result, likewise cidstart and cidsignalling. The CID number shows up properly if a telephone with a CID display is plugged into one of the POTS line, whether the call is from inside the university's PBX or outside. I'm suspecting some sort of timing difference between CID information sent for calls inside the PBX, versus calls originating outside. Could anyone shoot me some links with detailed pointers on debugging such a situation? I've explored the wiki, but did not find the incarnation needed to see exactly what chan_zap is doing. Thanks, Jeff -- Jeff Bachtel ([EMAIL PROTECTED],TAMU) http://www.cepheid.org/~jeff The sciences, each straining in [finger [EMAIL PROTECTED] for PGP key] its own direction, have hitherto harmed us little; - HPL, TCoC ___ --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] off-hook warning tone
On Tuesday 04 September 2007 06:31:08 pm Mojo with Horan Company, LLC wrote: Anthony, Robert here is mentioning a SIP phone, but I didn't see you specify what kind of phone you have. Is this accurate, or is it a Zaptel FXS port? a zap fxs port is correct. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ --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] How to make call from asterisk?
Helps us help you further, what do you intend to do? - Dial using a normal telephone line - Dial using a VoIP provider? What hardware do you have, etc On 9/5/07, neoh kumyee [EMAIL PROTECTED] wrote: Hi, I'm new to asterisk, in order to enable X-lite to make a call, what should i do before making a call? Current stage, 1. i have create a few accounts in sip.conf. 2. Registration are successful. Pls advice me how to continue then... Thanks Call and stay connected with your friends and family for free. Seen and be heard with high-definition video calls on Windows Live Messenger. Try it! ___ --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 -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --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] How to make call from asterisk?
To make call to X-lite or any sip phone , 1. create extension in sip.conf for soft phone. 2. register sip phone with that exentension which is in sip.conf -give your asterisk server ip in softphone and also username and password. 3. wait for some time 4. if all are good then your phone has been registered. ___ --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] E1 Line Tapping
Hi all, My name is Ricardo and unfortunately I'm just crawling in this telecomm/asterisk world. So, after reading all day long i still don't understand a few things. :D I'm trying to develop a call recorder for a costumer. He has a small call center ( 10 agents ) and want to record all calls. Since he already has everything (ACD only) working perfectly in the PBX and don't want me to touch it, I need do develop a less intrusive as possible system. I was thinking to do a line tapping in his E1 branch before it reaches the PBX and record it using Asterisk, then develop a small web interface to recover the recordings. In my research about E1 line tapping I found this product from Sangoma ( http://www.sangoma.com/datasheets/tapping ) but could not understand exactly how it really works. Does anybody already used it? Is it possible to use it with Asterisk? tia, Ricardo Gemignani ___ --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] Cepstral's Allison is having troublespeaking clearly
How are you playing the voice? Do you use something like app_swift or app_cepstral? Just fixed app_swift for my own installation by changing the framesize constant definition from 160*4 to 20, after googling for a similar issue. Works like a charm now. It only broke recently, i.e. not with the first 1.4.x releases, but maybe only a couple of months ago. On 9/3/07, Todd Reese [EMAIL PROTECTED] wrote: OK, I just reset the RTP packets to .020 as you have suggested. I can tell a little difference but the problem is still there. TIA, Todd -- I am Dyslexic of Borg. Fusistance is retile. Your ass will be laminated! ___ --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] How to make call from asterisk?
Hi, Thanks for your reply.. I am intend to dial using a VOIP provider.(developed by us) Software: x-Lite (SIP softphone) Registration of account number is fine, but for the case when i dial a number, it prompt out a message that the number not found. From my understanding, asterisk can be SIP server? or we need to implement a SIP server to integrate with Asterisk in order to provide full picture of VOIP system? Thanks. Date: Wed, 5 Sep 2007 13:30:21 +1000 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] How to make call from asterisk? Helps us help you further, what do you intend to do? - Dial using a normal telephone line - Dial using a VoIP provider? What hardware do you have, etc On 9/5/07, neoh kumyee [EMAIL PROTECTED] wrote: Hi, I'm new to asterisk, in order to enable X-lite to make a call, what should i do before making a call? Current stage, 1. i have create a few accounts in sip.conf. 2. Registration are successful. Pls advice me how to continue then... Thanks Call and stay connected with your friends and family for free. Seen and be heard with high-definition video calls on Windows Live Messenger. Try it! ___ --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 -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --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 _ Stay connected with your friends and discover new ones on Windows Live Spaces! http://spaces.live.com?mkt=en-my___ --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
