I use the E-series Nokia phones on my Wireless LAN. The e series have sip agent
On 8/20/07, asterisk-users-request at lists.digium.com
<asterisk-users-request at lists.digium.com>
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> Today's Topics:
>
> 1. Re: Snom 300 Hints and LIne Buttons (Philipp Kempgen)
> 2. Asterisk 2 Speechphone/Mandi (Steve Turner)
> 3. Rewriting the From and Subject from voicemail for a MMS
> Message to a Cell Phone - like visual voicemail (Steve Turner)
> 4. How many calls can use the same username (bilal ghayyad)
> 5. Re: How many calls can use the same username (Julio Tejera)
> 6. Re: How many calls can use the same username (Philipp Kempgen)
> 7. Nokia cell connected to Asterisk (Jonathan GF)
> 8. Re: Nokia cell connected to Asterisk (Steve Totaro)
> 9. Re: asterisk multiport (Walter Willis)
> 10. Re: Quick DUNDi Poll Questions, For All Asterisk, Users,
> Please Give Feedback (Matthew Brothers)
> 11. Application for Home Delivery Restaurants (Kashif Naeem)
> 12. Re: Nokia cell connected to Asterisk (mitcheloc)
> 13. asterisk1.2.24 or asterisk1.4.10.1 (fateme fatah)
> 14. Re: Change Packetization Time (Dovid B)
> 15. Re: Quick DUNDi Poll Questions, For All Asterisk, Users,
> Please Give Feedback (Tzafrir Cohen)
> 16. Re: 2 asterisk servers, how to connect them together? (Lenz)
> 17. Re: Siemens Gigaset DECT base provisioning (Olivier)
> 18. Re: Faxing through a PAP2 (Olivier)
> 19. Firefly IAX2 configuration (bilal ghayyad)
> 20. Redundancy / Failover (Khaled Chehab)
> 21. Re: Application for Home Delivery Restaurants (Matt Riddell)
> 22. Re: Firefly IAX2 configuration (Gordon Henderson)
> 23. Queues with Dynanic Users (BUG?) (Tim Groeneveld)
> 24. Re: Queues with Dynanic Users (BUG?) (Atis)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Sun, 19 Aug 2007 19:36:46 +0200
> From: Philipp Kempgen <philipp.kempgen at amooma.de>
> Subject: Re: [asterisk-users] Snom 300 Hints and LIne Buttons
> To: Asterisk Users <asterisk-users at lists.digium.com>
> Message-ID: <46C87FAE.80102 at amooma.de>
> Content-Type: text/plain; charset=ISO-8859-15
>
> Russell Brown wrote:
>
> > I've setup hints for a couple of Snom 300's but Asterisk
doesn't send
> > Extension Changed messages to subscribed phones unless the second
'line'
> > button is used (I've tried Snom's version 6 and 7 and two
difference
> > 300s).
> >
> > On the Asterisk Console I don't see any message when picking up a
Snom
> > 300 and dialing the hold music (or making any otehr call).
> >
> > As soon as I put the first call on hold though (by pressing the L2
> > button), Asterisk pops up the message "xtension Changed 116 new
state
> > Hold for Notify User Russell".
> >
> > If I drop the first 'line', there's no message from
Asterisk.
> >
> > When I flip back to the second line Asterisk says "Extension
Changed 116
> > new state Idle for Notify User Russell" - even though it's
patently not!
> >
> > This obviously makes the BLF lamp on my Snom 370 pretty useless as it
> > only lights up when the Snom 300's got two lines going :-(
> >
> > Can anyone point me in the right direction to getting this fixed?
>
> Do your peers in sip.conf have call-limit=<something>, e.g.
> call-limit=10
>
> sip.conf [general] settings:
> allowsubscribe=yes
> subscribecontext=default
> notifyringing=yes
> notifyhold=yes
> limitonpeers=yes
>
> Regards,
> Philipp Kempgen
>
> --
> amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
> Let's use IT to solve problems and not to create new ones.
> Asterisk? -> http://www.das-asterisk-buch.de
> My pick of the month: rfc 2822 3.6.5
>
> Gesch?ftsf?hrer: Stefan Wintermeyer
> Handelsregister: Neuwied B 14998
>
>
>
> ------------------------------
>
> Message: 2
> Date: Sun, 19 Aug 2007 15:39:41 -0400
> From: "Steve Turner" <lists65 at gmail.com>
> Subject: [asterisk-users] Asterisk 2 Speechphone/Mandi
> To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
> <asterisk-users at lists.digium.com>
> Message-ID: <000001c7e298$b01f34d0$9001a8c0 at xpwp01>
> Content-Type: text/plain; charset="us-ascii"
>
> Has anyone that has the Speechphone/Mandi service been able to set up a SIP
> connection directly with their servers?
>
> If so, would you want to share any information on how to do this?
>
>
>
>
>
>
>
> ------------------------------
>
> Message: 3
> Date: Sun, 19 Aug 2007 15:49:13 -0400
> From: "Steve Turner" <lists65 at gmail.com>
> Subject: [asterisk-users] Rewriting the From and Subject from
> voicemail for a MMS Message to a Cell Phone - like visual voicemail
> To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
> <asterisk-users at lists.digium.com>
> Message-ID: <000101c7e29a$057f7ab0$9001a8c0 at xpwp01>
> Content-Type: text/plain; charset="us-ascii"
>
> I would like to send Multimedia Messaging (MMS) email (gateway) to my cell
> phone and have the from and subject be the callerid/calleridnam information
> from the voice mail message.
>
> I know there is a way to call another perl script or program up when an
> email message is written, but I am not a programmer.
>
> I know there could be a perl script or program that could run every minute
> and check the
>
> /var/spool/asterisk/voicemail/default/XXXX/INBOX and read the msgxxxx.txt
> file and get the information and then attach the msgxxx.wav file and email
> it but again I am no programmer. Does anyone know if this has been done or
> is willing to do it?
>
> This would be similar to the iPhone visual voicemail using MMS on cell
> phones. Just a thought.
>
> Any ideas or thoughts?
>
>
>
>
>
>
>
> ------------------------------
>
> Message: 4
> Date: Sun, 19 Aug 2007 14:29:51 -0700 (PDT)
> From: bilal ghayyad <bilmar_gh at yahoo.com>
> Subject: [asterisk-users] How many calls can use the same username
> To: asterisk-users at lists.digium.com
> Message-ID: <479763.20456.qm at web53908.mail.re2.yahoo.com>
> Content-Type: text/plain; charset=iso-8859-1
>
> Hi List;
>
> If I configured one SIP account or one IAX account
> [sipuser1] or [iaxuser1] then how many calls can be
> originate/terminate using the same account [sipuser1]
> or [iaxuser1]?
>
> In other words, can 10 IP Phones (users) do a calls
> via Asterisk using the same account (SIP or IAX2)?
>
> If yes, how can I control the number of calls per
> account?
>
> Regards
> Bilal
>
>
>
>
____________________________________________________________________________________
> Luggage? GPS? Comic books?
> Check out fitting gifts for grads at Yahoo! Search
>
http://search.yahoo.com/search?fr=oni_on_mail&p=graduation+gifts&cs=bz
>
>
>
> ------------------------------
>
> Message: 5
> Date: Sun, 19 Aug 2007 17:04:09 -0600
> From: "Julio Tejera" <jat at unixtrends.com>
> Subject: Re: [asterisk-users] How many calls can use the same username
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID: <000a01c7e2b5$416a9350$6401a8c0 at Guajiro>
> Content-Type: text/plain; format=flowed; charset="iso-8859-1";
> reply-type=original
>
>
>
>
> > Hi List;
> >
> > If I configured one SIP account or one IAX account
> > [sipuser1] or [iaxuser1] then how many calls can be
> > originate/terminate using the same account [sipuser1]
> > or [iaxuser1]?
> >
> > In other words, can 10 IP Phones (users) do a calls
> > via Asterisk using the same account (SIP or IAX2)?
> >
> > If yes, how can I control the number of calls per
> > account?
> >
> > Regards
> > Bilal
> >
>
> Hi
>
> It can be done with "call-limit" into sip.conf
>
> And I'm not pretty sure but in iax.conf it must be
> "incominglimit/outgoinglimit"
>
> jat
>
>
>
> ------------------------------
>
> Message: 6
> Date: Mon, 20 Aug 2007 00:21:37 +0200
> From: Philipp Kempgen <philipp.kempgen at amooma.de>
> Subject: Re: [asterisk-users] How many calls can use the same username
> To: Asterisk Users <asterisk-users at lists.digium.com>
> Message-ID: <46C8C271.9060402 at amooma.de>
> Content-Type: text/plain; charset=ISO-8859-15
>
> bilal ghayyad wrote:
>
> > If I configured one SIP account or one IAX account
> > [sipuser1] or [iaxuser1] then how many calls can be
> > originate/terminate using the same account [sipuser1]
> > or [iaxuser1]?
>
> The number of calls per account is not really limited
> (for SIP at least).
>
> > In other words, can 10 IP Phones (users) do a calls
> > via Asterisk using the same account (SIP or IAX2)?
>
> Unless things have changed: No. (not sure about IAX)
> The number of registrations to an account *is* limited
> (to 1).
>
> > If yes, how can I control the number of calls per
> > account?
>
> sip.conf: call-limit
>
>
> Regards,
> Philipp Kempgen
>
> --
> amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
> Let's use IT to solve problems and not to create new ones.
> Asterisk? -> http://www.das-asterisk-buch.de
> My pick of the month: rfc 2822 3.6.5
>
> Gesch?ftsf?hrer: Stefan Wintermeyer
> Handelsregister: Neuwied B 14998
>
>
>
> ------------------------------
>
> Message: 7
> Date: Mon, 20 Aug 2007 00:26:32 +0200
> From: "Jonathan GF" <jonathan at surestorm.com>
> Subject: [asterisk-users] Nokia cell connected to Asterisk
> To: asterisk-users at lists.digium.com
> Message-ID:
> <a50c1e90708191526l23222f8do790719ac9f90c3c7 at mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> Hi folks,
>
> i've been looking for in many sources but i cannot see clear if the
options
> i'm chasing is feasible with Asterisk. I understand that should be.
>
> I would like to connect a nokia cell to Asterisk but i don't know how
> exactly.
>
> Any ideas, inputs, docs or refs will be welcomed.
>
> Thanks in advance.
>
> Jonathan GF
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> ------------------------------
>
> Message: 8
> Date: Sun, 19 Aug 2007 18:45:45 -0400
> From: "Steve Totaro" <stotaro at totarotechnologies.com>
> Subject: Re: [asterisk-users] Nokia cell connected to Asterisk
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <6B79B93B96CD4540BBDFA78DAA2A4EB00241B0 at
mail.first-notification.local>
> Content-Type: text/plain; charset="iso-8859-1"
>
> If it is bluetooth and you don't mind running Asterisk 1.4 trunk, you
should
> look at chan_mobile.
>
> Thanks,
> Steve Totaro
>
> ________________________________
>
> From: asterisk-users-bounces at lists.digium.com on behalf of Jonathan GF
> Sent: Sun 8/19/2007 6:26 PM
> To: asterisk-users at lists.digium.com
> Subject: [asterisk-users] Nokia cell connected to Asterisk
>
>
> Hi folks,
>
> i've been looking for in many sources but i cannot see clear if the
options
> i'm chasing is feasible with Asterisk. I understand that should be.
>
> I would like to connect a nokia cell to Asterisk but i don't know how
> exactly.
>
> Any ideas, inputs, docs or refs will be welcomed.
>
> Thanks in advance.
>
> Jonathan GF
>
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> ------------------------------
>
> Message: 9
> Date: Sun, 19 Aug 2007 18:27:15 -0500
> From: "Walter Willis" <walterwn at gmail.com>
> Subject: Re: [asterisk-users] asterisk multiport
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <6b0bc7870708191627x3a2c6e6dwb8d80ae2abb38cec at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> thank you.
>
> On 8/17/07, Steven <asterisk at tescogroup.com> wrote:
> >
> > Ahh, I see.
> >
> >
> > Good point.
> >
> > --
> > --
> > Steven
> >
> > http://www.glimasoutheast.org
> >
> >
> >
> > "Steve Totaro" <stotaro at totarotechnologies.com>
wrote in message news:
> > 46C5827E.9020905 at totarotechnologies.com...
> > > Steven wrote:
> > >> I am curious.
> > >>
> > >> Why would one need to do this?
> > >>
> > >> If a phone connect to 5060 from another port number, asterisk
happily
> > works, so why use multiple port on asterisk?
> > >>
> > >
> > > I cannot see the thread history but from the context, I would say
> > > because many ISPs block 5060, 25, and others.
> > >
> > > 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
> > >
> >
> >
> >
> >
> > _______________________________________________
> > --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
> >
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> ------------------------------
>
> Message: 10
> Date: Sun, 19 Aug 2007 21:00:33 -0400
> From: Matthew Brothers <matthew at brothersfamily.net>
> Subject: Re: [asterisk-users] Quick DUNDi Poll Questions, For All
> Asterisk, Users, Please Give Feedback
> To: asterisk-users at lists.digium.com
> Message-ID: <46C8E7B1.1030602 at brothersfamily.net>
> Content-Type: text/plain; charset=UTF-8
>
> > Questions:
> >
> > 1. Is the wiki DUNDi example and the dundi.conf file too difficult to
> > follow for new users?
> >
>
> I wouldn't exactly say that it is too difficult but that the target
> audience for the default examples is not the average person/entity
> that could make use of the power inherent with DUNDi. When an
> average * user/admin wants to use DUNDi they will want to start out
> small and local rather than worry about all of the intricacies of
> the e164 standard. It is much easier, in my opinion, to learn the
> power of DUNDi on a simple level and scale that up to a more
> globally connected platform.
>
> > 2. Does the complexity of the DUNDi setup discourage you from using it
> > or even attempting to configure it?
>
> I don't see this as the case. Most people who use * are comfortable
> with the level of complexity that is present in DUNDi, they just
> don't know where to start.
>
> > 3. If there was a simple tutorial, step by step guide with easy to
> > setup and test examples, would this encourage more users to
> > investigate and use DUNDi?
>
> Absolutely. If you need any help in putting this together or if you
> simply need people to review a tutorial, I would be glad to assist.
>
> > I'm interested in putting together a new-user tutorial about DUNDi
> > configuration and setup. There is a lot of great information, setup
> > guides already but the feedback I get is that the current examples are
> > a bit complicated to follow for new users.
>
> Thank you for being a part of the conference last Friday. Your
> participation is greatly appreciated.
>
>
>
> Matthew Brothers
>
>
>
> ------------------------------
>
> Message: 11
> Date: Mon, 20 Aug 2007 10:03:00 +0500
> From: "Kashif Naeem" <kashif at softhand.com.pk>
> Subject: [asterisk-users] Application for Home Delivery Restaurants
> To: asterisk-users at lists.digium.com
> Message-ID:
> <58fcf7400708192203u7e980b02q6506c83e64833f16 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hello All
>
> We have developed an application for Home Delivery Restaurants using
> Asterisk, Java (JSP/ JSF) and MySQL. Here is listing of its features. If
> someone is interested then we can provide him more details.
>
>
> - POP up window with caller data containing his/her name, address and
> transactions history.
> - In case of new customer, Pop up window with blank form to add
> customer data and order detail.
> - Invoice generation and print functionality of Invoice.
> - Black list a customer if he placed fake order and next time its
> black list status would show based on his CLI.
> - Call recording
> - Sales Analysis
>
>
> Regards,
>
> --
> Kashif Naeem
> Director
> Soft Hand
> www.softhand.com.pk
>
> Cell: +92 (0)345 4226006
> Office: +92 (0)42 5692766
>
> Email: kashif at softhand.com.pk
> MSN: kashif__naeem at hotmail.com
> Gmail: meet.kashif at gmail.com
> Skype: kashif.naeem
>
> 302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan.
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> ------------------------------
>
> Message: 12
> Date: Sun, 19 Aug 2007 22:08:06 -0700
> From: mitcheloc <mitcheloc at gmail.com>
> Subject: Re: [asterisk-users] Nokia cell connected to Asterisk
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <da5bbae90708192208v185cdf73y93281789444ac124 at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Jonathon,
>
> Are you talking about using the built in SIP client on some Nokia
> phones? I'm using an E90 with Asterisk and it works very well. I used
> Google for help and it returned plenty of results.
>
> Cheers,
> Mitchel
>
> On 8/19/07, Steve Totaro <stotaro at totarotechnologies.com> wrote:
> > If it is bluetooth and you don't mind running Asterisk 1.4 trunk,
you
> should look at chan_mobile.
> >
> > Thanks,
> > Steve Totaro
> >
> > ________________________________
> >
> > From: asterisk-users-bounces at lists.digium.com on behalf of Jonathan
GF
> > Sent: Sun 8/19/2007 6:26 PM
> > To: asterisk-users at lists.digium.com
> > Subject: [asterisk-users] Nokia cell connected to Asterisk
> >
> >
> > Hi folks,
> >
> > i've been looking for in many sources but i cannot see clear if
the
> options i'm chasing is feasible with Asterisk. I understand that should
be.
> >
> > I would like to connect a nokia cell to Asterisk but i don't know
how
> exactly.
> >
> > Any ideas, inputs, docs or refs will be welcomed.
> >
> > Thanks in advance.
> >
> > Jonathan GF
> >
> >
> > _______________________________________________
> > --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
> >
> >
>
>
> --
> ________________
> Mitchel Constantin
> Snap - A desktop user interface for Asterisk
> www.snapanumber.com
>
>
>
> ------------------------------
>
> Message: 13
> Date: Sun, 19 Aug 2007 22:27:00 -0700 (PDT)
> From: fateme fatah <faza_404 at yahoo.com>
> Subject: [asterisk-users] asterisk1.2.24 or asterisk1.4.10.1
> To: asterisk-users at lists.digium.com
> Message-ID: <262034.64100.qm at web56602.mail.re3.yahoo.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi:
> You offer me use asterisk1.2.24 or asterisk1.4.10.1.How's it if I want
to
> use astbill?
> Best Regards.
>
>
> ---------------------------------
> Fussy? Opinionated? Impossible to please? Perfect. Join Yahoo!'s user
panel
> and lay it on us.
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> ------------------------------
>
> Message: 14
> Date: Mon, 20 Aug 2007 09:09:50 +0300
> From: "Dovid B" <asteriskusers at dovid.net>
> Subject: Re: [asterisk-users] Change Packetization Time
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID: <004f01c7e2f0$ba875c10$0500a8c0 at DovidLaptop>
> Content-Type: text/plain; format=flowed; charset="iso-8859-1";
> reply-type=original
>
>
> ----- Original Message -----
> From: "Dan Austin" <Dan_Austin at Phoenix.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Sent: Sunday, August 19, 2007 7:58 PM
> Subject: Re: [asterisk-users] Change Packetization Time
>
>
> > Dovid wrote:
> >
> >> Does anyone know if it is possible to change the
> >> packetization time in Asterisk ? I was told by a client
> >> of mine that adjusting this with using G729 can greatly
> >> lower the amount of bandwidth used.
> >
> > Your client is correct. Configurable packetization was added
> > introduced with the release of 1.4.0. For details look at the
> > rtp-packetization.txt file in the doc directory for full details.
> >
> > The short answer is to append :<size> to any codec on your allow
> > directive that you want to change from the default of 20ms.
> > Ex.
> > Allow=g729:40
> >
> > Dan
> >
> >
>
> Dan,
> Can I make this change in 1.2.X ? (maybe in the source ?). I have not moved
> to 1.4.X because of the lack of support. Currently using SpanDSP.
>
>
>
>
>
> ------------------------------
>
> Message: 15
> Date: Mon, 20 Aug 2007 09:50:33 +0300
> From: Tzafrir Cohen <tzafrir.cohen at xorcom.com>
> Subject: Re: [asterisk-users] Quick DUNDi Poll Questions, For All
> Asterisk, Users, Please Give Feedback
> To: asterisk-users at lists.digium.com
> Message-ID: <20070820065033.GD12822 at xorcom.com>
> Content-Type: text/plain; charset=us-ascii
>
> On Sun, Aug 19, 2007 at 09:00:33PM -0400, Matthew Brothers wrote:
> > > Questions:
> > >
> > > 1. Is the wiki DUNDi example and the dundi.conf file too
difficult to
> > > follow for new users?
> > >
> >
> > I wouldn't exactly say that it is too difficult but that the
target
> > audience for the default examples is not the average person/entity
> > that could make use of the power inherent with DUNDi. When an
> > average * user/admin wants to use DUNDi they will want to start out
> > small and local rather than worry about all of the intricacies of
> > the e164 standard. It is much easier, in my opinion, to learn the
> > power of DUNDi on a simple level and scale that up to a more
> > globally connected platform.
>
> I'd say that duni.conf is a reference, and you expect it to be an
> introductory document. A reference should be comprehensive. It is best
> used after you've grasped the basic concepts, and together with a text
> search. Asterisk's "sample" configuration files actually
serve a role
> of a reference.
>
> If you were to look for an introduction-level document in the asterisk
> source, you should have started in the /doc directory.
>
> Sadly the documentation there is close to non-existing at the moment:
> http://www.asterisk.org/doxygen/1.4/AstDUNDi.html
>
> How did I find that page? I went to the doxygen-generated documentation
> for 1.4:
>
> http://www.asterisk.org/doxygen/1.4/
>
> In there, one non-trivial jump to the rest of the interesting
> documentation:
>
> Related Pages
>
> And there I can find some pretty handy documentation. If you have
> anything more to comment on that, I guess the place for that is either
> the (practically dead) asterisk-doc mailing list, or looking at some of
> the work done on the admin guide for 1.6 .
>
> (yeah, I know, patches are welcome, docs talk, whatever)
>
> --
> Tzafrir Cohen
> icq#16849755 jabber:tzafrir at jabber.org
> +972-50-7952406 mailto:tzafrir.cohen at xorcom.com
> http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir
>
>
>
> ------------------------------
>
> Message: 16
> Date: Mon, 20 Aug 2007 09:02:17 +0200
> From: Lenz <lenz-ml at loway.it>
> Subject: Re: [asterisk-users] 2 asterisk servers, how to connect them
> together?
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID: <op.txci53z43wzjep at pc-lenz>
> Content-Type: text/plain; format=flowed; delsp=yes;
> charset=iso-8859-15
>
>
> You may want to start from here: http://astrecipes.net/index.php?n=204
> l.
>
>
> On Sun, 19 Aug 2007 00:46:45 +0200, Ade Vickers
> <javickers at solutionengineers.com> wrote:
>
> > Hi...
> >
> > I have what is, I am sure, a relatively common & straightforward
problem
> > (no, NOT that kind of problem!)... I'm trying to hook two asterisk
> > servers
> > together so I can make a "distributed" PBX.
> >
> > Here's the scenario:
> >
> > [MASTER] is in the office. It has unrestricted access to the internet,
> > and a
> > fixed IP address. It has 3 SIP hardphones configured & working,
plus a
> > couple of softphones which log in/out as necessary. The phones are on
> > extensions 5100-5104, with a special extension 5999 which just plays
> > music.
> >
> > [HOME] is at home. It has internet access only through a Microsoft ISA
> > 2003
> > firewall, and has a dynamic IP address. It has 1 SIP hardphone
> > configured,
> > and working, on extension 5110. I can add a second hardphone to verify
> > that
> > this (new build) server is working OK, but all of the messages
indicate
> > it's
> > fine.
> >
> > What I want to do, obviously, is have ALL of the extensions (5XXX)
> > "pretending" to be on the same PBX. i.e. if I dial 5100 (on
[MASTER])
> > from
> > 5110 (on [HOME]), the call goes through & everyone's happy;
and vice
> > versa,
> > calling 5110 from 5100.
> >
> > I know I need to use IAX to achieve this (as IAX can negotiate its way
> > past
> > the firewall), but I can't find the magic incantations for
IAX.CONF (on
> > either server) to make them talk nicely to each other. They did, very
> > briefly, as the [MASTER] server spotted the IP address of [HOME],
added
> > it
> > to the peer list, & my heart rose; but, now it's dead again.
Rather than
> > post my broken conf files here, can anyone suggest a
nice'n'easy way to
> > get
> > this to work?
> >
> > Many thanks in advance.
> > Ade.
> >
>
> --
> Loway Research - Home of QueueMetrics
> http://queuemetrics.com
>
>
>
> ------------------------------
>
> Message: 17
> Date: Mon, 20 Aug 2007 09:19:33 +0200
> From: Olivier <oza-4h07 at myamail.com>
> Subject: Re: [asterisk-users] Siemens Gigaset DECT base provisioning
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <442fbb120708200019n4807cd08w8465a978edd022dd at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> 2007/8/13, Paul Hayes <paul at provu.co.uk>:
> >
> >
> > It's not currently possible but Siemens are working on new
firmware for
> > at least the S450IP model which will support auto-config using http.
> > I'm not sure when it's due for release though.
>
>
> Thanks for the tip !
>
> Directly asking to Siemens (
> http://gigaset.siemens.com/shc/0,1935,hq_en_0_11729_rArNrNrNrN,00.html)
> before posting to this list, was not very helpful (to say the least).
>
> How should I track this firmware release ?
> Should I just check with
>
http://gigaset.siemens.com/shc/0,1935,hq_en_0_123868_rArNrNrNrN_variation%253A-5_pageType%253Adownloads_imagePos%253A0,00.html#content
> for post V02063 firmware ?
>
> Regards
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> ------------------------------
>
> Message: 18
> Date: Mon, 20 Aug 2007 09:26:16 +0200
> From: Olivier <oza-4h07 at myamail.com>
> Subject: Re: [asterisk-users] Faxing through a PAP2
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <442fbb120708200026h4a4f2f53j18231c8c8e64dbd9 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Did you try T.38 ?
> These PAP2 boxes should be able to benefit from Asterisk T.38 pass through
> capabilities.
> You would then have to install a T.38 termination device, such as Linksys
> 3102 :
>
> PSTN -------- Linksys 3102 ----------- LAN --------- PAP2 ----------- Fax
> machine
>
> Cheers
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> ------------------------------
>
> Message: 19
> Date: Mon, 20 Aug 2007 00:47:45 -0700 (PDT)
> From: bilal ghayyad <bilmar_gh at yahoo.com>
> Subject: [asterisk-users] Firefly IAX2 configuration
> To: asterisk-users at lists.digium.com
> Message-ID: <823112.69927.qm at web53910.mail.re2.yahoo.com>
> Content-Type: text/plain; charset=iso-8859-1
>
> Hi List;
>
> I am using Firefly softphone Version 1.9.9 Build 4521
> and I select IAX protocol and did the configuration in
> Network1 (and I checked the Active checkbox) as
> following:
>
> Server: 192.168.8.4
> username: iax2user1
> password: password
>
> In the Asterisk, I did the following configuration on
> the /etc/asterisk/iax.conf:
>
> [iax2user1]
> type=friend
> context=internal
> username=iax2user1
> secret=password
> host=dynamic
>
> Then I ran the following:
> #/usr/sbin/asterisk -cvvv
> CLI>reload
>
> But always I get a message at the firefly that an
> error occured while trying to connect to the network.
>
> What else I have to do?
>
> By the way: what is the command that I can type it to
> do tracing on the user [iax2user1] or to do traces on
> any registeration attempts from the clients?
>
> Last thing, if I am outside the console (in unix
> mode), is there any command from unix I can type it to
> know if asterisk is running or not?
>
> Regards
> Bilal
>
>
>
>
>
>
>
>
>
>
____________________________________________________________________________________
> Moody friends. Drama queens. Your life? Nope! - their life, your story.
Play
> Sims Stories at Yahoo! Games.
> http://sims.yahoo.com/
>
>
>
> ------------------------------
>
> Message: 20
> Date: Mon, 20 Aug 2007 10:47:39 +0300
> From: "Khaled Chehab" <kchehab at xplorium.com>
> Subject: [asterisk-users] Redundancy / Failover
> To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
> <asterisk-users at lists.digium.com>
> Cc: <asterisk-users-bounces at lists.digium.com>
> Message-ID:
> <mailman.2399.1187604999.10646.asterisk-users at lists.digium.com>
> Content-Type: text/plain; charset="utf-8"
>
> Dears
>
>
>
> Any one succeeded to make Redundancy / Failover with asterisk 1.4.9 on
> centos with kernel 2.6.9-55.EL.
>
> Can you please send me the documentation link on how to or write down how
to
> .
>
>
>
>
>
>
>
> Regards
>
>
>
>
>
>
> *********************************************
> No employee or agent is authorized to conclude any binding agreement on
> behalf of Xplorium with another party by e-mail without express written
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> in this electronic message do not necessarily reflect views of Xplorium or
> its subsidiaries and associates.
>
> This electronic message and its attachments are solely addressed to the
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> If you are not the intended addressee of this electronic message and its
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>
> ------------------------------
>
> Message: 21
> Date: Mon, 20 Aug 2007 20:07:52 +1200
> From: Matt Riddell <matt at venturevoip.com>
> Subject: Re: [asterisk-users] Application for Home Delivery
> Restaurants
> To: kashif at softhand.com.pk, Asterisk Users Mailing List -
> Non-Commercial Discussion <asterisk-users at lists.digium.com>
> Message-ID: <46C94BD8.3030701 at venturevoip.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Kashif Naeem wrote:
> > Hello All
> >
> > We have developed an application for Home Delivery Restaurants using
> > Asterisk, Java (JSP/ JSF) and MySQL. Here is listing of its features.
If
> > someone is interested then we can provide him more details.
> >
> >
> > - POP up window with caller data containing his/her name, address
and
> > transactions history.
> > - In case of new customer, Pop up window with blank form to add
> > customer data and order detail.
> > - Invoice generation and print functionality of Invoice.
> > - Black list a customer if he placed fake order and next time its
> > black list status would show based on his CLI.
> > - Call recording
> > - Sales Analysis
>
> URL?
>
> Licence? I'm assuming free seeing as this was sent to the
> "Non-Commercial Discussion" list.
>
> - --
> Kind Regards,
>
> Matt Riddell
> Director
> _______________________________________________
>
> http://www.venturevoip.com (Great new VoIP end to end solution)
> http://www.venturevoip.com/news.php (Daily Asterisk News - html)
> http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss)
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.7 (MingW32)
> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
>
> iD8DBQFGyUvYDQNt8rg0Kp4RAodVAJ90MjdlubuVD0Em6ekXXkjWi6uy3gCfVGzu
> E4u0QbRRxKTG1AvRL5kgUU8> =iiJk
> -----END PGP SIGNATURE-----
>
>
>
> ------------------------------
>
> Message: 22
> Date: Mon, 20 Aug 2007 09:31:21 +0100 (BST)
> From: Gordon Henderson <gordon+asterisk at drogon.net>
> Subject: Re: [asterisk-users] Firefly IAX2 configuration
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <Pine.LNX.4.64.0708200925540.5361 at lion.drogon.net>
> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed
>
> On Mon, 20 Aug 2007, bilal ghayyad wrote:
>
> > Hi List;
> >
> > I am using Firefly softphone Version 1.9.9 Build 4521
> > and I select IAX protocol and did the configuration in
> > Network1 (and I checked the Active checkbox) as
> > following:
> >
> > Server: 192.168.8.4
> > username: iax2user1
> > password: password
> >
> > In the Asterisk, I did the following configuration on
> > the /etc/asterisk/iax.conf:
> >
> > [iax2user1]
> > type=friend
> > context=internal
> > username=iax2user1
> > secret=password
> > host=dynamic
> >
> > Then I ran the following:
> > #/usr/sbin/asterisk -cvvv
> > CLI>reload
> >
> > But always I get a message at the firefly that an
> > error occured while trying to connect to the network.
> >
> > What else I have to do?
>
> Have you checked your firewall? Is it letting UDP data through to the
> asterisk box on port 4569?
>
> > By the way: what is the command that I can type it to
> > do tracing on the user [iax2user1] or to do traces on
> > any registeration attempts from the clients?
>
> iax2 debug
>
> will generate lots of output for you...
>
> > Last thing, if I am outside the console (in unix mode), is there any
> > command from unix I can type it to know if asterisk is running or not?
>
> ps ax | grep asterisk
>
> is crude, but visual.
>
> Asterisk stores it's PID in /var/run/asterisk.pid, so you could then
read
> that, and check to see if the process with that PID is actually running
> asterisk.
>
> ie. see if /proc/<number> existis, and if-so, see if it's
actually
> asterisk by reading /proc/<number>/cmdline
>
> or just see if you can connect to it with the rasterisk command ...
>
> Gordon
>
>
>
> ------------------------------
>
> Message: 23
> Date: Mon, 20 Aug 2007 19:40:24 +1000
> From: Tim Groeneveld <tim at timg.ws>
> Subject: [asterisk-users] Queues with Dynanic Users (BUG?)
> To: asterisk-users at lists.digium.com
> Message-ID: <200708201940.27145.tim at timg.ws>
> Content-Type: text/plain; charset="us-ascii"
>
> I am running r79979 of Asterisk Trunk, and I am having problems trying to
> use
> app_queue.so.
>
> I want to use the extension 510 to be a line where users can call technical
> support.
>
> Extensions 511 and 512 are used by the operators to dynamically make
> themselves a Queue Member or not.
>
> So, operators call 511, and they should get added to the Queue as a Queue
> member.
>
> When users call 510 then, it actually does ring everyone who has called
511.
>
> The problem is when the operator comes to pick up the call. The operator
> hears
> nothing, and the user still hears the Music on Hold. Not only that, but
> after
> about 5 seconds, the operators call gets dropped.
>
> Is there anything that I am doing wrong?
>
> Thanks,
>
> Tim
>
>
> here are snipits of my config files:
> == extensions.conf => [default]
> exten => 510,1,Answer
> exten => 510,2,Queue(techsupport,t)
>
> exten => 511,2,Set(CALLBACKNUM=${CALLERID(number)})
> exten => 511,3,AddQueueMember(techsupport)
> exten => 511,4,Playback(queue-techsupport-in)
> exten => 511,5,Hangup
>
> == queues.conf => [techsupport]
> music=default
> strategy = ringall
> timeout = 10
> retry = 2
> maxlen = 0
> announce-frequency = 10
> announce-holdtime = yes
>
> == agents.conf => [general]
> ackcall=no
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> ------------------------------
>
> Message: 24
> Date: Mon, 20 Aug 2007 13:16:32 +0300
> From: Atis <atis at BEST.eu.org>
> Subject: Re: [asterisk-users] Queues with Dynanic Users (BUG?)
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <945196e0708200316s45110040ldaa8cac6f03ef13 at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> On 8/20/07, Tim Groeneveld <tim at timg.ws> wrote:
> > I am running r79979 of Asterisk Trunk, and I am having problems trying
to
> use
> > app_queue.so.
> >
> > I want to use the extension 510 to be a line where users can call
> technical
> > support.
> >
> > Extensions 511 and 512 are used by the operators to dynamically make
> > themselves a Queue Member or not.
> >
> > So, operators call 511, and they should get added to the Queue as a
Queue
> > member.
> >
> > When users call 510 then, it actually does ring everyone who has
called
> 511.
> >
> > The problem is when the operator comes to pick up the call. The
operator
> hears
> > nothing, and the user still hears the Music on Hold. Not only that,
but
> after
> > about 5 seconds, the operators call gets dropped.
> >
> > Is there anything that I am doing wrong?
> >
> > Thanks,
> >
> > Tim
> >
> >
> > here are snipits of my config files:
> > == extensions.conf => > [default]
> > exten => 510,1,Answer
> > exten => 510,2,Queue(techsupport,t)
> >
> > exten => 511,2,Set(CALLBACKNUM=${CALLERID(number)})
> > exten => 511,3,AddQueueMember(techsupport)
> > exten => 511,4,Playback(queue-techsupport-in)
> > exten => 511,5,Hangup
> >
> > == queues.conf => > [techsupport]
> > music=default
> > strategy = ringall
> > timeout = 10
> > retry = 2
> > maxlen = 0
> > announce-frequency = 10
> > announce-holdtime = yes
> >
> > == agents.conf => > [general]
> > ackcall=no
>
> Can you also provide output of "show queues" and "show
channels"? Plus
> the logfile of dial to 511.
>
> I'm using QueueAdd after AgentCallbackLogin (trough manager API).
> Maybe you need to use AgentCallbackLogin first?
>
> Regards,
> Atis
>
>
> --
> Atis Lezdins,
> IT Responsible of BEST Riga,
> atis at BEST.eu.org
> ICQ: 142239285
> Skype: atis.lezdins
> Cell Phone: +371 28806004 [Tele2, Latvia]
> Work phone: +1 800 7502835 [Toll free, USA]
> ?BEST? -> www.BEST.eu.org
>
>
>
> ------------------------------
>
> _______________________________________________
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>
> End of asterisk-users Digest, Vol 37, Issue 79
> **********************************************
>
