[Asterisk-Users] skype channel
Wessel de Roode
wessel at sourcelab.nl
Sun May 15 17:15:51 MST 2005
> Message: 10
> Date: Sun, 15 May 2005 21:41:23 +0000
> From: Laurent Lesage <laurent at lesagepono.be>
> Subject: Re: [Asterisk-Users] skype channel
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <4287C203.20903 at lesagepono.be>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Hi *,
>
> I was just going to ask the same question. Does anybody have an
> information about Skype and Asterisk? Any link?
>
> Thanks in advance
I've just added a view day's ago some information on it on the wiki.
As far as I know there is nothing really working 'yet' but I'm sure since
the API is out it' won't take long :-)
http://www.voip-info.org/tiki-index.php?page=bounty%20skype
Wessel de Roode
>
> Laurent
>
>
> Bartek Kania a icrit :
>
> >-----BEGIN PGP SIGNED MESSAGE-----
> >Hash: SHA1
> >
> >I just noticed that the Skype API for linux seems to be available.
> >I've read before a number of posts where people were talking about
> >implementing a chan_skype with the skype API.
> >
> >I wonder if there is any progress in that direction, and if anyone is
> >working on it.
> >
> >/B
> >- --
> >* GPG-Key: http://evil.gnarf.org/mrbk.pgp
> >
> >A: Because we read from top to bottom, left to right.
> >Q: Why should i start my reply below the quoted text?
> >- -- http://www.i-hate-computers.demon.co.uk/
> >
> >-----BEGIN PGP SIGNATURE-----
> >Version: GnuPG v1.2.5 (GNU/Linux)
> >
> >iD8DBQFCgLlVWYjaxM2wIe4RAuSKAJ9VNMIO2h838Y2yXAFDAQaJOjPa3gCfeokZ
> >Ghsrpa8Gp3pHt5/bUinZKUA=
> >=fUgt
> >-----END PGP SIGNATURE-----
> >_______________________________________________
> >Asterisk-Users mailing list
> >Asterisk-Users at lists.digium.com
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> >
>
>
>
> ------------------------------
>
> Message: 11
> Date: Sun, 15 May 2005 17:49:41 -0400
> From: Paul <digium-list at 9ux.com>
> Subject: Re: [Asterisk-Users] knopsterisk
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <4287C3F5.8000607 at 9ux.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> trixter http://www.0xdecafbad.com wrote:
>
> >does anyone have knopsterisk for download, I assume that
> because its GPL
> >the creator of that iso cant restrict spreading it. A
> friend wanted it
> >to play on a box and the only thing I can find with google is the
> >knopsterisk.com site which wants $10 to get a copy and does
> not provide
> >(as far as I can tell) any free distribution access which is
> >his/hers/its/them/they/whatever right (being politically correct is
> >hard).
> >
> >If there is some distribution problem with doing this then I
> would also
> >appreciate hearing why it cant be distroed by 3rd parties.
> >
> >Thanks
> >
> >
> The website says "/*Now with Asterisk Version 1.0!" which makes me
> wonder how many they have sold. Also makes me wonder if the
> knoppix part
> is very up to date.
>
> They don't mention licensing/copyright anywhere. We can
> figure that all
> the software is on the CD is under free licensing but all
> they have to
> do is add a single readme file with a restricted license or copyright
> and you make identical copies of the CD. I would first try contacting
> them and get those details. You also want to know where the source is
> because there might be some modifications they made to
> knoppix packages
> or the packages they added.
>
> I think you would be better off to make a knoppix CD, boot it
> and get *
> installed and running. After that read the following and
> maybe you can
> create something better to share with the world.
>
> http://www.knoppix.net/wiki/Knoppix_Remastering_Howto
>
> */
>
>
> ------------------------------
>
> Message: 12
> Date: Sun, 15 May 2005 15:55:03 -0600
> From: Ira Burton <ira.burton at gmail.com>
> Subject: [Asterisk-Users] 911 Options
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <add0723b050515145542d49c47 at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> I am curious if anybody has pointers on the best way to get the 7
> digit PSAP number for an area. I am thinking about making a '911'
> extension that will dial the PSAP number, wait for the PSAP to answer
> and play a message giving the address of the originating call, and
> replay the the information every three minutes. I am concerned what
> may happen if my children try to dial 911 in an emergency but do not
> yet know our address.
>
> How are other people handling this?
>
>
> ------------------------------
>
> Message: 13
> Date: Sun, 15 May 2005 15:15:15 -0700
> From: "trixter http://www.0xdecafbad.com" <trixter at 0xdecafbad.com>
> Subject: Re: [Asterisk-Users] knopsterisk
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <1116195315.15943.91.camel at rufus.home.tld>
> Content-Type: text/plain; charset="us-ascii"
>
> On Sun, 2005-05-15 at 17:49 -0400, Paul wrote:
> > I think you would be better off to make a knoppix CD, boot
> it and get *
> > installed and running. After that read the following and
> maybe you can
> > create something better to share with the world.
>
> Unfortunately that wont help my friend who wanted to play but not
> reformat his disk. I already have a debian system running
> asterisk, so
> I personally wouldnt get anything (other than having to redo all my
> configs :P
>
> I just didnt know if it was restricted in any way which would prevent
> someone else who had a copy from making it available, and if
> someone on
> here does have a copy if they could make it available, in the
> hopes that
> maybe today my friend could download it.
>
>
> --
> Trixter http://www.0xdecafbad.com
> UK +44 870 340 4605 Germany +49 801 777 555 3402
> US +1 360 207 0479 or +1 516 687 5200
> FreeWorldDialup: 635378
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> ------------------------------
>
> Message: 14
> Date: Sun, 15 May 2005 15:17:53 -0700
> From: "trixter http://www.0xdecafbad.com" <trixter at 0xdecafbad.com>
> Subject: Re: [Asterisk-Users] 911 Options
> To: Ira Burton <ira.burton at gmail.com>, Asterisk Users
> Mailing List -
> Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <1116195473.15979.95.camel at rufus.home.tld>
> Content-Type: text/plain; charset="us-ascii"
>
> On Sun, 2005-05-15 at 15:55 -0600, Ira Burton wrote:
> > I am curious if anybody has pointers on the best way to get the 7
> > digit PSAP number for an area. I am thinking about making a '911'
> > extension that will dial the PSAP number, wait for the PSAP
> to answer
> > and play a message giving the address of the originating call, and
> > replay the the information every three minutes. I am concerned what
> > may happen if my children try to dial 911 in an emergency but do not
> > yet know our address.
> >
>
> You can buy them on CD, however to do E911 you have to have a special
> trunk to the switch that the PSAP is off of, which transmits
> the E parts
> of E911 not just the audio.
>
> Where to buy them I dont know offhand, I do specifically recall seeing
> pages that sold national CDs (how adt, onstar, even other
> PSAPs contact
> a specific PSAP when needed).
>
> I do remember that I was googling psap administrators and other such
> things.
>
>
> --
> Trixter http://www.0xdecafbad.com
> UK +44 870 340 4605 Germany +49 801 777 555 3402
> US +1 360 207 0479 or +1 516 687 5200
> FreeWorldDialup: 635378
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> ------------------------------
>
> Message: 15
> Date: Mon, 16 May 2005 07:20:47 +0900 (JST)
> From: Zen Kato <zenkato at pis.bekkoame.ne.jp>
> Subject: [Asterisk-Users] can't CLI> STOP NOW by zombie MOH
> To: asterisk-users at lists.digium.com
> Message-ID: <20050516.072047.74757831.zenkato at pis.bekkoame.ne.jp>
> Content-Type: Text/Plain; charset=us-ascii
>
> I am using CVS-v1-0-04/20/05-12:31:26. Windows Messenger5.1 works MOH
> fine. After I stop MOH on Windows Messenger, if the hungup
> signal could
> not send to *, the sip channel(e.g.,SIP/52001-08ca) for this
> MOH remains.
> Then the user trys again MOH, a new sip channel starts. And again
> the hugup signal can not send to *,.........
>
> When I 'stop now' from CLI> , * cleanups the remaining sip
> channels as follows;
>
> *CLI> stop now
> Beginning asterisk shutdown....
> -- Stopped music on hold on SIP/52001-9e3b
> == Spawn extension (sip, 6000, 2) exited non-zero on
> 'SIP/52001-9e3b'
> -- Stopped music on hold on SIP/52001-08ca
> -- Stopped music on hold on SIP/52001-63fd
> == Spawn extension (sip, 6000, 2) exited non-zero on
> 'SIP/52001-63fd'
> == Spawn extension (sip, 6000, 2) exited non-zero on
> 'SIP/52001-08ca'
> -- Executing Hangup("SIP/52001-9e3b", "") in new stack
> == Spawn extension (sip, h, 1) exited non-zero on 'SIP/52001-9e3b'
> -- Executing Hangup("SIP/52001-08ca", "") in new stack
> == Spawn extension (sip, h, 1) exited non-zero on 'SIP/52001-08ca'
> -- Executing Hangup("SIP/52001-63fd", "") in new stack
> == Spawn extension (sip, h, 1) exited non-zero on 'SIP/52001-63fd'
> Executing last minute cleanups
> == Destroying any remaining musiconhold processes
>
> The CPU goes to 99% usage, but can't stop mpg123 and asterisk,
> the following 'top' show ;
>
> 5737 root 25 0 3832 2588 2424 R 99.7 0.5 2:54.00 mpg123
> 5397 root 15 0 167m 18m 6212 S 2.7 3.7 0:10.95 X
> 5627 root 15 0 47988 6604 3624 S 1.3 1.3 0:15.64 asterisk
> 4673 root 17 0 8588 6820 1620 S 0.3 1.3 0:02.05 hald
> 5509 zenkato 15 0 24168 9.8m 7068 S 0.3 2.0 0:00.47 metacity
> 5551 zenkato 15 0 58500 17m 10m S 0.3 3.4 0:01.20
> gnome-terminal
> ....(snip)...
>
> So, I have to do 'kill -9 pid-of-mpg123'. * goes to
> segmentation fault.
>
> Is this safe way to stop asterisk?
>
> When UA could not send 'hangup signal' to *, the following
> warnig came out
> console.
>
> -- Executing Answer("SIP/52003-f48d", "") in new stack
> -- Executing MusicOnHold("SIP/52003-f48d", "") in new stack
> -- Started music on hold, class 'default', on SIP/52003-f48d
> -- Stopped music on hold on SIP/52003-f48d
> == Spawn extension (sip, 6000, 2) exited non-zero on
> 'SIP/52003-f48d'
> -- Executing Hangup("SIP/52003-f48d", "") in new stack
> == Spawn extension (sip, h, 1) exited non-zero on 'SIP/52003-f48d'
> -- Registered SIP '52001' at 192.168.0.12 port 9558 expires 120
> -- Saved useragent "RTC/1.3.5369 (Messenger 5.1.0639)"
> for peer 52001
> -- Executing Answer("SIP/52001-9e3b", "") in new stack
> -- Executing MusicOnHold("SIP/52001-9e3b", "") in new stack
> -- Started music on hold, class 'default', on SIP/52001-9e3b
> May 16 05:45:50 WARNING[5627]: chan_sip.c:695 retrans_pkt:
> Maximum retries exceeded on call
> a4ea6066f3e34d5aa5ce56b09ea520bf for seqno 1 (Non-critical
> Response)Warning, flexibel rate not heavily tested!
> Warning, flexibel rate not heavily tested!
> Warning, flexibel rate not heavily tested!
> Warning, flexibel rate not heavily tested!
> Warning, flexibel rate not heavily tested!
> Warning, flexibel rate not heavily tested!
> Warning, flexibel rate not heavily tested!
> May 16 06:01:25 NOTICE[5627]: rtp.c:355 ast_rtcp_read: RTP:
> Received packet with bad UDP checksum
> May 16 06:01:45 NOTICE[5627]: rtp.c:355 ast_rtcp_read: RTP:
> Received packet with bad UDP checksum
> May 16 06:02:05 NOTICE[5627]: rtp.c:355 ast_rtcp_read: RTP:
> Received packet with bad UDP checksum
> ........(snip).....
> May 16 06:12:05 NOTICE[5627]: rtp.c:355 ast_rtcp_read: RTP:
> Received packet with bad UDP checksum
> Warning, flexibel rate not heavily tested!
> May 16 06:12:35 NOTICE[5627]: rtp.c:355 ast_rtcp_read: RTP:
> Received packet with bad UDP checksum
> May 16 06:12:55 NOTICE[5627]: rtp.c:355 ast_rtcp_read: RTP:
> Received packet with bad UDP checksum
> -- Executing Answer("SIP/52001-63fd", "") in new stack
> -- Executing MusicOnHold("SIP/52001-63fd", "") in new stack
> -- Started music on hold, class 'default', on SIP/52001-63fd
> May 16 06:13:05 NOTICE[5627]: rtp.c:355 ast_rtcp_read: RTP:
> Received packet with bad UDP checksum
> May 16 06:13:09 WARNING[5627]: chan_sip.c:695 retrans_pkt:
> Maximum retries exceeded on call
> 0dc3a12940a34f21bd7f816ef45d0f75 for seqno 1 (Non-critical
> Response)May 16 06:13:25 NOTICE[5627]: rtp.c:355
> ast_rtcp_read: RTP: Received packet with bad UDP checksum
> -- Executing Answer("SIP/52001-08ca", "") in new stack
> -- Executing MusicOnHold("SIP/52001-08ca", "") in new stack
> -- Started music on hold, class 'default', on SIP/52001-08ca
> May 16 06:13:39 WARNING[5627]: chan_sip.c:695 retrans_pkt:
> Maximum retries exceeded on call
> 4e503aa4498b45e89632b34034b5e9f1 for seqno 1 (Non-critical
> Response)May 16 06:13:45 NOTICE[5627]: rtp.c:355
> ast_rtcp_read: RTP: Received packet with bad UDP checksum
> May 16 06:13:54 NOTICE[5627]: rtp.c:355 ast_rtcp_read: RTP:
> Received packet with bad UDP checksum
> May 16 06:14:14 NOTICE[5627]: rtp.c:355 ast_rtcp_read: RTP:
> Received packet with bad UDP checksum
> May 16 06:14:14 NOTICE[5627]: rtp.c:355 ast_rtcp_read: RTP:
> Received packet with bad UDP checksum
> May 16 06:14:15 NOTICE[5627]: rtp.c:355 ast_rtcp_read: RTP:
> Received packet with bad UDP checksum
>
>
> Regards,
>
> Zen Kato
>
>
>
>
> ------------------------------
>
> Message: 16
> Date: Sun, 15 May 2005 18:35:27 -0400
> From: "Chris Mason (Lists)" <lists at masonc.com>
> Subject: RE: [Asterisk-Users] Road Warrior phone config
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <asterisk-users at lists.digium.com>
> Message-ID: <20050515223527.D089792C412 at mercury.mason.home>
> Content-Type: text/plain; charset="us-ascii"
>
> > You serious? I typed all that and you were asking about
> > RETRIEVING vm all along? Wow, I must be really dense today.
>
> No, I know it now :-)
>
> > So: don't pass calleridnum to extension 8500. Or configure a
> > different voicemail retrieval exten for roaming users and
> > pass null to voicemailmain.
>
> Yes, that could work.
>
> > Or, even better, scrape off and discard the fourth extension
> > digit when parsing calleridnum and handing to voicemailmain.
>
> I like that better.
>
> Thanks
>
> Chris Mason
> www.anguillaguide.com
> Tel: (305) 704-7249 Fax: (815)301-9759
>
>
>
> ------------------------------
>
> Message: 17
> Date: Sun, 15 May 2005 16:47:33 -0600
> From: Andres Paglayan <andres at paglayan.com>
> Subject: Re: [Asterisk-Users] Road Warrior phone config
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <4287D185.8040904 at paglayan.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> question about this thread,
> would a wi-fi voip phone work for this guy?
> meaning, he takes it to wherever he goes and it gets
> registered wherever
> it as wireless access.
> is that theoretically correct?
>
> >
> >
> >
>
>
>
> ------------------------------
>
> Message: 18
> Date: Mon, 16 May 2005 00:58:25 +0200
> From: "Thierry Wehr" <wehr at japet.com>
> Subject: [Asterisk-Users] Compile problem on last CVS
> To: <asterisk-users at lists.digium.com>
> Message-ID: <20050515225833.E43A574FC6 at mail.lirbox.net>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Good evening
>
> from the CVS of the 2005/05/14 it's impossible to build asterisk* on a
> redhat 7.3
>
> i get this at compile time
>
> chan_sip.c: In function `build_user':
> chan_sip.c:10007: parse error before `struct'
> chan_sip.c:10029: `userflags' undeclared (first use in this function)
> chan_sip.c:10029: (Each undeclared identifier is reported only once
> chan_sip.c:10029: for each function it appears in.)
> chan_sip.c:10029: `mask' undeclared (first use in this function)
> chan_sip.c:10094: warning: type defaults to `int' in
> declaration of `__s'
> chan_sip.c:10094: warning: comparison of distinct pointer
> types lacks a cast
> chan_sip.c: In function `build_peer':
> chan_sip.c:10176: parse error before `struct'
> chan_sip.c:10221: `peerflags' undeclared (first use in this function)
> chan_sip.c:10221: `mask' undeclared (first use in this function)
> chan_sip.c:10391: warning: type defaults to `int' in
> declaration of `__s'
> chan_sip.c:10391: warning: comparison of distinct pointer
> types lacks a cast
> make[1]: *** [chan_sip.o] Erreur 1
> make[1]: Quitte le ripertoire
> `/usr/src/asterisk-cvs/asterisk/channels'
> make: *** [subdirs] Erreur 1
>
> may be someone have a clue to fix it
>
> best rehards
> Thierry
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> ------------------------------
>
> Message: 19
> Date: Sun, 15 May 2005 18:03:50 -0500
> From: "Tim Connolly" <tim at timsnet.com>
> Subject: RE: [Asterisk-Users] Compile problem on last CVS
> To: <wehr at japet.com>, "'Asterisk Users Mailing List - Non-Commercial
> Discussion'" <asterisk-users at lists.digium.com>
> Message-ID: <20050515230402.91E622FF8CE at lists.digium.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Maybe try a version of redhat that was released in the past 5 years?
> Seriously, why do you require RH7.3 over Fedora or even RH 9?
>
>
>
> _____
>
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
> Thierry Wehr
> Sent: Sunday, May 15, 2005 5:58 PM
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] Compile problem on last CVS
>
>
>
> Good evening
>
>
>
> from the CVS of the 2005/05/14 it's impossible to build asterisk* on a
> redhat 7.3
>
>
>
> i get this at compile time
>
>
>
> chan_sip.c: In function `build_user':
> chan_sip.c:10007: parse error before `struct'
> chan_sip.c:10029: `userflags' undeclared (first use in this function)
> chan_sip.c:10029: (Each undeclared identifier is reported only once
> chan_sip.c:10029: for each function it appears in.)
> chan_sip.c:10029: `mask' undeclared (first use in this function)
> chan_sip.c:10094: warning: type defaults to `int' in
> declaration of `__s'
> chan_sip.c:10094: warning: comparison of distinct pointer
> types lacks a cast
> chan_sip.c: In function `build_peer':
> chan_sip.c:10176: parse error before `struct'
> chan_sip.c:10221: `peerflags' undeclared (first use in this function)
> chan_sip.c:10221: `mask' undeclared (first use in this function)
> chan_sip.c:10391: warning: type defaults to `int' in
> declaration of `__s'
> chan_sip.c:10391: warning: comparison of distinct pointer
> types lacks a cast
> make[1]: *** [chan_sip.o] Erreur 1
> make[1]: Quitte le ripertoire
> `/usr/src/asterisk-cvs/asterisk/channels'
> make: *** [subdirs] Erreur 1
>
>
>
> may be someone have a clue to fix it
>
>
>
> best rehards
>
> Thierry
>
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> ------------------------------
>
> Message: 20
> Date: Sun, 15 May 2005 19:29:58 -0400
> From: Paul <digium-list at 9ux.com>
> Subject: Re: [Asterisk-Users] Road Warrior phone config
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <4287DB76.7060706 at 9ux.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Andres Paglayan wrote:
>
> >question about this thread,
> >would a wi-fi voip phone work for this guy?
> >meaning, he takes it to wherever he goes and it gets
> registered wherever
> >it as wireless access.
> >is that theoretically correct?
> >
> >
> I like that approach. Those toys will be getting more affordable. One
> concern I would have is battery life. I think a wisip phone
> that can be
> recharged/powered via standard usb cable would be nice.
>
>
>
> ------------------------------
>
> Message: 21
> Date: Mon, 16 May 2005 00:33:02 +0100
> From: Tony Hoyle <tmh at nodomain.org>
> Subject: Re: [Asterisk-Users] knopsterisk
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <4287DC2E.4080007 at nodomain.org>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> trixter http://www.0xdecafbad.com wrote:
> > does anyone have knopsterisk for download, I assume that
> because its GPL
> > the creator of that iso cant restrict spreading it. A
> friend wanted it
> > to play on a box and the only thing I can find with google is the
> > knopsterisk.com site which wants $10 to get a copy and does
> not provide
> > (as far as I can tell) any free distribution access which is
> > his/hers/its/them/they/whatever right (being politically correct is
> > hard).
>
> The GPL does allow the creator to charge a redistribution
> charge... it
> doesn't require free distribution (which would be a bit harsh
> for some
> projects - bandwidth isn't free, and CDs/Burners certainly aren't).
>
> The CD may contain items of proprietary software - until
> recently SuSE
> was like this, and Redhat RHEL still is. In that case you can
> redistribute most of the contents of the CD but not the CD
> itself (ie.
> not a working copy, especially if the proprietary part is the
> installer). In that case your option is to strip the GPL parts and
> build a new distribution, wait for someone to do the same, or
> pay for a
> copy.
>
> If it doesn't contain proprietary parts, and all its contents
> are freely
> licensed, just find someone who's paid for a copy and dupicate
> it/download it from them.
>
> Tony
>
>
> ------------------------------
>
> Message: 22
> Date: Sun, 15 May 2005 18:33:37 -0500
> From: "Tim Connolly" <tim at timsnet.com>
> Subject: RE: [Asterisk-Users] Road Warrior phone config
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <asterisk-users at lists.digium.com>
> Message-ID: <20050515233338.1DDDB2FE4CD at lists.digium.com>
> Content-Type: text/plain; charset="us-ascii"
>
> Or have a small solar panel on the back of the phone. Stick
> it on the dash
> of your car, assuming it doesn't burst into flames from heat;
> it should be
> fully charged in an hour or two.
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Paul
> Sent: Sunday, May 15, 2005 6:30 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Road Warrior phone config
>
> Andres Paglayan wrote:
>
> >question about this thread,
> >would a wi-fi voip phone work for this guy?
> >meaning, he takes it to wherever he goes and it gets
> registered wherever
> >it as wireless access.
> >is that theoretically correct?
> >
> >
> I like that approach. Those toys will be getting more affordable. One
> concern I would have is battery life. I think a wisip phone
> that can be
> recharged/powered via standard usb cable would be nice.
>
> _______________________________________________
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>
>
> ------------------------------
>
> Message: 23
> Date: Sun, 15 May 2005 18:53:19 -0500
> From: "Tim Connolly" <tim at timsnet.com>
> Subject: [Asterisk-Users] FXO/FXS suggestions:
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <asterisk-users at lists.digium.com>
> Message-ID: <20050515235319.C08F02FFAED at lists.digium.com>
> Content-Type: text/plain; charset="us-ascii"
>
> I'm looking for a zaptel type device with one (or
> more) FXO and
> one (or more) FXS port. Basically this guy would sit in-line
> of your phone
> line (PCI card). Any suggestions? TDM400 would be overkill.
>
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>
> End of Asterisk-Users Digest, Vol 10, Issue 117
> ***********************************************
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