[Asterisk-Users] Re: Asterisk-Users Digest, Vol 1, Issue 5082

Francisco Perez-Landaeta fplandae at hotmail.com
Thu Sep 9 09:34:50 MST 2004


Anyone using the recently MAC OS X ? Version of asterisk ?
Thanks,

Francisco Perez-Landaeta

> From: asterisk-users-request at lists.digium.com
> Reply-To: asterisk-users at lists.digium.com
> Date: Fri, 27 Aug 2004 13:08:24 -0500 (CDT)
> To: asterisk-users at lists.digium.com
> Subject: Asterisk-Users Digest, Vol 1, Issue 5082
> 
> Send Asterisk-Users mailing list submissions to
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> 
> Today's Topics:
> 
>  1. Re: how to fetch a call? (Tony Mountifield) (Sudhir Kumar)
>  2. Asterisk Assistants Custom Icon (Sunrise Ltd)
>  3. RE: Overhead Paging (Jay Milk)
>  4. libr2 (Vikram Rangnekar)
>  5. Speech Recognition and Asterisk (Mike Meyer)
>  6. Re: FXOs (Marcelo Mercio Dandrea)
>  7. RE: sip change? (Jerry Roy)
>  8. RE: sip change? (Chad Brown)
>  9. RE: Overhead Paging (Rich Adamson)
> 10. Re: sip change? (Doug Shubert)
> 11. RE: Faxing with SPANDSP or any other mean ? Is    itpossible ?
>     Am I dreaming ? (Jean-Fran?ois Rousseau)
> 12. RE: Faxing with SPANDSP or any other mean ? Is    itpossible ?
>     Am I dreaming ? (Jean-Fran?ois Rousseau)
> 13. RE: Faxing with SPANDSP or any other mean ? Is    itpossible ?
>     Am I dreaming ? (Jean-Fran?ois Rousseau)
> 
> 
> ----------------------------------------------------------------------
> 
> Message: 1
> Date: 27 Aug 2004 13:07:10 -0400
> From: Sudhir Kumar <sudhir1 at adelphia.net>
> Subject: [Asterisk-Users] Re: how to fetch a call? (Tony Mountifield)
> To: asterisk-users at lists.digium.com
> Cc: tony at softins.clara.co.uk, roger at planinternet.de
> Message-ID: <1093626430.2021.182.camel at homedell>
> Content-Type: text/plain
> 
> Remote Call Pick up feature is very much implemented in asterisk. You
> can pick up a call for another extension by dialing  *8#
> 
> To be able to do that, you need to have the extensions in the same
> pickup group, configurable through sip.conf and zapata.conf.
> 
> -- sudhir
> 
>> ------------------------------
>> 
>> Message: 14
>> Date: Fri, 27 Aug 2004 14:17:26 +0000 (UTC)
>> From: tony at softins.clara.co.uk (Tony Mountifield)
>> Subject: [Asterisk-Users] Re: how to fetch a call?
>> To: asterisk-users at lists.digium.com
>> Message-ID: <cgnfpm$56k$1 at softins.clara.co.uk>
>> 
>> In article <412F4122.6070401 at planinternet.de>,
>> Roger Schreiter <roger at planinternet.de> wrote:
>>> Hi,
>>> 
>>> there is a feature, which I would like to use with asterisk,
>>> and I assume it exists.
>>> Unfortunately I don't know how to say it in english.
>>> In german it's "einen Ruf heranholen".
>>> 
>>> It means:
>>> The phone set of my collegue is ringing, and I'm hearing
>>> the ringing.
>>> I know, that my collegue is not at his desk, and now
>>> I want to answer the call at my phone (instead of
>>> running to my collegue's desc to answer at his phone).
>> 
>> I don't know whether it is implemented or not in Asterisk, but the
>> feature is known in English as "call pickup".
>> 
>> mfg,
>> Tony
> 
> 
> 
> ------------------------------
> 
> Message: 2
> Date: Sat, 28 Aug 2004 02:14:45 +0900 (JST)
> From: Sunrise Ltd <stsltdtyo at yahoo.co.jp>
> Subject: [Asterisk-Users] Asterisk Assistants Custom Icon
> To: astusr <asterisk-users at lists.digium.com>
> Cc: chriss at watertech.com
> Message-ID: <20040827171445.8897.qmail at web2604.mail.mci.yahoo.co.jp>
> Content-Type: text/plain; charset=iso-2022-jp
> 
> I think I need to clarify what I meant by custom icon for
> the Asterisk Assistants in my earlier posting.
> 
> On the Mac an Assistant is what the Windoze world calls a
> Wizard and there is a generic icon for it - the front of a
> dinner suit with bow tie, the one you can see on the Wiki.
> 
> However, many of Apple's own assistants have a little mark
> in the lower right corner of the generic icon which
> further hints at what the respective assistant is for.
> 
> An example for that is the Airport Assistant which has a
> little Airport base station in the lower right corner ...
> 
> http://www.sunrise-tel.com/screenshots/AirportAssistantIcon.png
> 
> (Aiport is what Apple calls their WiFi gear)
> 
> What I had in mind for the Asterisk Assistants is an icon
> just like the one at the above link, but with an Asterisk
> replacing the Airport base station in the lower right
> corner.
> 
> This would fit in and still project Asterisk's "brand"
> into the Mac world.
> 
> 
> So, please don't get too fancy with this, it's more of a
> cut and paste kind of job which I had in mind. For those
> who are interested in making more fancy icons, we'll have
> other tools outside of the Assistant series later on ;-)
> 
> thanks
> rgds
> benjk
> 
> 
> --
> Sunrise Telephone Systems Ltd
> 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan
> 
> __________________________________________________
> GANBARE! NIPPON!
> Yahoo! JAPAN JOC OFFICIAL INTERNET PORTAL SITE
> http://mail.ganbare-nippon.yahoo.co.jp/
> 
> 
> 
> ------------------------------
> 
> Message: 3
> Date: Fri, 27 Aug 2004 12:27:22 -0500
> From: "Jay Milk" <jay at skimmilk.net>
> Subject: RE: [Asterisk-Users] Overhead Paging
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <asterisk-users at lists.digium.com>
> Message-ID: <134401c48c5b$1e09ae40$c8fea8c0 at gbox.us>
> Content-Type: text/plain;    charset="us-ascii"
> 
> Yeah, like what?  I'm still looking for a reasonably-priced
> paging/monitoring system I can use with Asterisk.  PA systems with
> distributed speakers/amplifiers can be had for around $50/speaker.
> IP-based solutions come in at several hundred dollars per station.
> Considering that disparity, you'll continue to see a lot distributed
> amplification systems going in.
> 
>> -----Original Message-----
>> From: Rich Adamson [mailto:radamson at routers.com]
>> Sent: Friday, August 27, 2004 11:53 AM
>> Subject: Re: [Asterisk-Users] Overhead Paging
>> 
>> 
>> I would seriously doubt whether folks see much of that
>> anymore. Lots of other ways to address boat-loads of speakers
>> and long cable runs with current technology.
>> 
>> Rich
> 
> 
> 
> ------------------------------
> 
> Message: 4
> Date: Fri, 27 Aug 2004 19:22:29 +0200
> From: Vikram Rangnekar <vicky at freebsdcluster.net>
> Subject: [Asterisk-Users] libr2
> To: asterisk-users at lists.digium.com
> Message-ID: <20040827172229.GA40041 at freebsdcluster.org>
> Content-Type: text/plain; charset=us-ascii
> 
> I just came across libr2 anyone using it in its current state. Specifically
> someone from India or around India using it. Also does it work with the
> digium e1 cards or only the Dialogic cards.
> 
> http://digium-cvs.netmonks.ca/viewcvs.cgi/libr2/
> 
> -- 
> regards
> Vikram (http://www.vicramresearch.com)
> 
> 
> ------------------------------
> 
> Message: 5
> Date: Fri, 27 Aug 2004 12:26:51 -0500
> From: Mike Meyer <mjmeyer at gendesign.com>
> Subject: [Asterisk-Users] Speech Recognition and Asterisk
> To: Asterisk Users Group <asterisk-users at lists.digium.com>
> Message-ID: <1093627611.2871.391.camel at newbox.gendesign>
> Content-Type: text/plain
> 
> All;
> 
> Since I have interest in providing the capability for callers to speak
> the department, person or number they wish to call, as well as other IVR
> scenarios, I have been reviewing much of this lists email archives and
> searching the web for open source voice recognition that will work with
> the Asterisk PBX.
> 
> What I am trying to determine, is what will it take to get it working on
> Asterisk? How much effort and cost?
> 
> So far I have uncovered references to the following:
> 
>       1) VoiceXML standards, and forums
>       2) OpenVXI - which supports VoiceXML, simulated speech,
>       telephony
>       3) PublicVoiceXML
>       4) Sphinx - a Carnegie Mellon University Speech recognition
>       project funded by DARPA
>       
>> From what I can tell, I feel I am uncovering the tip of the ice berg and
> this may not be trivial. But it seems that the Voice recognition
> application, once developed, would have to be linked via an AGI to the
> asterisk dial plan.
> 
> Has anyone gotten Voice recognition working with Asterisk? Last I saw, a
> few were attempting to apply Sphinx back in the December and April time
> frame. Any shared successes, progress or direction on Sphinx or any
> other VR app would be appreciated before I start down this road.
> 
> Thanks,
> Mike Meyer
> 
> 
> 
> ------------------------------
> 
> Message: 6
> Date: Fri, 27 Aug 2004 14:42:42 -0300
> From: "Marcelo Mercio Dandrea" <marcdan at terra.com.br>
> Subject: Re: [Asterisk-Users] FXOs
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID: <023e01c48c5d$4161eb80$7a0aa8c0 at HIKARU>
> Content-Type: text/plain;    charset="iso-8859-1"
> 
> Hello,
> 
>       I?ve been doing tests with a Softphone and a TDM400 with two FXO
> modules. At first, I had echo and specially gain problems with it, but after
> enabling Mark2/Aggressive echo cancelation, the echo problem vanished. The
> gain problem seems to be due to (I suppose) some noise induced by my sound
> card. I?ve monitored it using ztmonitor, and the TX gain was fixed on the
> top. On the PSTN side, this resulted in "periods of silence" from time to
> time. When I changed my soundcard, the problem vanished.
> 
> Marcelo
> 
> ----- Original Message -----
> From: <mgraves at mstvp.com>
> To: <asterisk-users at lists.digium.com>
> Sent: Friday, August 27, 2004 12:41 PM
> Subject: [Asterisk-Users] FXOs
> 
> 
>> Hi All,
>> 
>> I'd really like to see a show of hands with regard to people's
>> experience with FXO interfaces. I own a few X100p cards and have had
>> nothing but problems with them.
>> 
>> I also took part in Sipura's beta program, for the SPA-3000. While it
>> can be an improvement over the X100p, it presently has echo problems
>> that make it unusable. Sipura has not acknowledged the problem ( at
>> least to me) although several in the user community make refernce to
>> new firmware that might address the issue, real soon now.
>> 
>> I see a lot of activity recently on-list about the TDM-400. Of course,
>> mentions on-list are more than likely the result of people having
>> problems. We don't hear about people who have no issues with a product.
>> 
>> So, the nature of my inquiry is to explore how many people out here have
>> good/great experiences with the various small FXO adapters? While the
>> TDM-400 is my next possible purchase I'd also like to hear about
>> devices from Welltech, Clipcomm, Micronet, Multitech, Immixtel, etc.
>> With so many products being offered I would hope that we have some
>> collective experience with each one.
>> 
>> Thanks,
>> Michael
>> 
>> 
>> 
>> Michael Graves
>> Sr Product Specialist
>> Pixel Power Inc
>> mgraves at pixelpower.com
>> o(713)861-4005
>> o(800)905-6412
>> f(713)864-8668
>> c(713)201-1262
>> 
>> 
>> 
>> _______________________________________________
>> 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: 7
> Date: Fri, 27 Aug 2004 10:41:55 -0700
> From: Jerry Roy <JRoy at GoRemote.com>
> Subject: RE: [Asterisk-Users] sip change?
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <C0E23A9B1942314FAEC0134C189CF79BC5B8B9 at elm.ent.gric.com>
> Content-Type: text/plain;    charset="iso-8859-1"
> 
> Hi All,
> 
> Looking for a recommendation. I was hoping to purchase a * "KIT" for a
> small office. I have 4 lines and 4 extensions need phones so I need 4
> phones. What phones would many of you recommend? Can you refer me to any
> companies that have built a kit I can plugin and configure?
> 
> Thanks,
> 
> Jerry Roy
> RemoteHand, Inc.
> 562-305-9545
> 
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Rich
> Adamson
> Sent: Friday, August 27, 2004 7:15 AM
> To: Asterisk-a-users-list
> Subject: [Asterisk-Users] sip change?
> 
> Just upgrade from July 12th cvs to last night CVS-HEAD-08/27/04-00:00:09
> 
> 
> When call comes in and is sent to a Cisco 7960, I see:
> 
>   -- Executing Dial("SIP/3008-9a9b", "SIP/3000|15") in new stack
>   -- Called 3000
> Aug 27 08:13:25 WARNING[1092070192]: chan_sip.c:676 retrans_pkt: Maximum
> retries
> exceeded on call 033f41c2187409b13ca364502ea9434e at 206.222.193.101 for
> seqno 102
> (Critical Request)
> == No one is available to answer at this time
>   -- Executing VoiceMail2("SIP/3008-9a9b", "u3000") in new stack
>   -- Playing 'voicemail/default/3000/greet' (language 'en')
>   -- Playing 'vm-isunavail' (language 'en')
> 
> but the phone doesn't ring. The 7960 is registered and can place
> outbound calls. Same with multiple 7960's.
> 
> Did I miss a mandatory config change, or is sip broken?
> 
> Rich
> 
> _______________________________________________
> 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: 8
> Date: Fri, 27 Aug 2004 10:51:05 -0700
> From: "Chad Brown" <chad.brown at identitymine.com>
> Subject: RE: [Asterisk-Users] sip change?
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <93D6FFFFE7DE7142962E7D4A331538E8034048 at IMEXBE01.identitymine.com>
> Content-Type: text/plain;    charset="us-ascii"
> 
> Jerry,
> 
> If your talking sip phones...
> 
> I am using the Cisco 7960 phones and love them. The quality and
> stability against Asterisk have been excellent.
> 
> Chad Brown - IdentityMine
> 
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jerry Roy
> Sent: Friday, August 27, 2004 10:42 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] sip change?
> 
> Hi All,
> 
> Looking for a recommendation. I was hoping to purchase a * "KIT" for a
> small office. I have 4 lines and 4 extensions need phones so I need 4
> phones. What phones would many of you recommend? Can you refer me to any
> companies that have built a kit I can plugin and configure?
> 
> Thanks,
> 
> Jerry Roy
> RemoteHand, Inc.
> 562-305-9545
> 
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Rich
> Adamson
> Sent: Friday, August 27, 2004 7:15 AM
> To: Asterisk-a-users-list
> Subject: [Asterisk-Users] sip change?
> 
> Just upgrade from July 12th cvs to last night CVS-HEAD-08/27/04-00:00:09
> 
> 
> When call comes in and is sent to a Cisco 7960, I see:
> 
>   -- Executing Dial("SIP/3008-9a9b", "SIP/3000|15") in new stack
>   -- Called 3000
> Aug 27 08:13:25 WARNING[1092070192]: chan_sip.c:676 retrans_pkt: Maximum
> retries
> exceeded on call 033f41c2187409b13ca364502ea9434e at 206.222.193.101 for
> seqno 102
> (Critical Request)
> == No one is available to answer at this time
>   -- Executing VoiceMail2("SIP/3008-9a9b", "u3000") in new stack
>   -- Playing 'voicemail/default/3000/greet' (language 'en')
>   -- Playing 'vm-isunavail' (language 'en')
> 
> but the phone doesn't ring. The 7960 is registered and can place
> outbound calls. Same with multiple 7960's.
> 
> Did I miss a mandatory config change, or is sip broken?
> 
> Rich
> 
> _______________________________________________
> 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
> _______________________________________________
> 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: 9
> Date: Fri, 27 Aug 2004 12:45:12 -0600
> From: Rich Adamson <radamson at routers.com>
> Subject: RE: [Asterisk-Users] Overhead Paging
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <Chameleon.1093629021.adar0 at vegas>
> Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1
> 
> My comment was in reference to the old single large amplifier and high
> voltage runs with room volume control, etc, as compared to much less
> expensive amplifier modules with small numbers of overhead speakers
> per module.
> 
> ------------------------
>> Yeah, like what?  I'm still looking for a reasonably-priced
>> paging/monitoring system I can use with Asterisk.  PA systems with
>> distributed speakers/amplifiers can be had for around $50/speaker.
>> IP-based solutions come in at several hundred dollars per station.
>> Considering that disparity, you'll continue to see a lot distributed
>> amplification systems going in.
>> 
>>> -----Original Message-----
>>> I would seriously doubt whether folks see much of that
>>> anymore. Lots of other ways to address boat-loads of speakers
>>> and long cable runs with current technology.
>>> 
>>> Rich
> 
> 
> 
> 
> ------------------------------
> 
> Message: 10
> Date: Fri, 27 Aug 2004 13:57:35 -0400
> From: Doug Shubert <doug at accessgate.net>
> Subject: Re: [Asterisk-Users] sip change?
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <412F760F.60800 at accessgate.net>
> Content-Type: text/plain; charset="us-ascii"
> 
> has anyone tested the vst1000 SIP phone from pcphoneline ?
> http://www.pcphoneline.com/
> 
> Doug
> 
> Jerry Roy wrote:
> 
>> Hi All,
>> 
>> Looking for a recommendation. I was hoping to purchase a * "KIT" for a
>> small office. I have 4 lines and 4 extensions need phones so I need 4
>> phones. What phones would many of you recommend? Can you refer me to any
>> companies that have built a kit I can plugin and configure?
>> 
>> Thanks,
>> 
>> Jerry Roy
>> RemoteHand, Inc.
>> 562-305-9545
>> 
>> -----Original Message-----
>> From: asterisk-users-bounces at lists.digium.com
>> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Rich
>> Adamson
>> Sent: Friday, August 27, 2004 7:15 AM
>> To: Asterisk-a-users-list
>> Subject: [Asterisk-Users] sip change?
>> 
>> Just upgrade from July 12th cvs to last night CVS-HEAD-08/27/04-00:00:09
>> 
>> 
>> When call comes in and is sent to a Cisco 7960, I see:
>> 
>>    -- Executing Dial("SIP/3008-9a9b", "SIP/3000|15") in new stack
>>    -- Called 3000
>> Aug 27 08:13:25 WARNING[1092070192]: chan_sip.c:676 retrans_pkt: Maximum
>> retries
>> exceeded on call 033f41c2187409b13ca364502ea9434e at 206.222.193.101 for
>> seqno 102
>> (Critical Request)
>>  == No one is available to answer at this time
>>    -- Executing VoiceMail2("SIP/3008-9a9b", "u3000") in new stack
>>    -- Playing 'voicemail/default/3000/greet' (language 'en')
>>    -- Playing 'vm-isunavail' (language 'en')
>> 
>> but the phone doesn't ring. The 7960 is registered and can place
>> outbound calls. Same with multiple 7960's.
>> 
>> Did I miss a mandatory config change, or is sip broken?
>> 
>> Rich
>> 
>> _______________________________________________
>> 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
>> _______________________________________________
>> 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
>> 
>> 
>>  
>> 
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> ------------------------------
> 
> Message: 11
> Date: Fri, 27 Aug 2004 14:03:03 -0400
> From: Jean-Fran?ois Rousseau <jrousseau at sys-tech.net>
> Subject: RE: [Asterisk-Users] Faxing with SPANDSP or any other mean ?
> Is    itpossible ? Am I dreaming ?
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <asterisk-users at lists.digium.com>
> Message-ID: <20040827180331.2740B2FE60B at lists.digium.com>
> Content-Type: text/plain;    charset="iso-8859-1"
> 
> Hi,
> 
> I've read the FAQ and tried to find a timing source. So far, I've compiled
> ztdummy and loaded it sucessfully. But it still not working. All I have is
> the beginning of the fax.
> 
> I've tried (HP FAX) --> PSTN --> X100P --> Asterisk -- > SPANDSP
> 
> And (HP FAX) --> IAXy --> Asterisk --> SPANDSP
> 
> Both do the same error...  About a quarter of the page is ok then garbage.
> The sending machine say that the fax was sent ok.
> 
> Here is some info that might help troubleshot my problem.
> 
> 
> 
> 
> ___________________________
> Jean-Fran?ois Rousseau
> Sys-Tech
> www.sys-tech.net
> jrousseau at sys-tech.net
> T?l. 24h (418) 520-0739
> T?lec.???(418) 520-4554
> Ligne directe 1-866-274-4870
> B?tisseurs de solutions informatiques et ?lectroniques
> urgence at sys-tech.net
> 1-877-969-tech
> Messages de confidentialit? : Ce courriel (de m?me que les fichiers joints)
> est strictement r?serv? ? l'usage de la personne ou de l'entit? ? qui il est
> adress? et peut contenir de l'information privil?gi?e et confidentielle.
> Toute divulgation, distribution ou copie de ce courriel est strictement
> prohib?e. Si vous avez re?u ce courriel par erreur, veuillez nous en aviser
> sur-le-champ, d?truire toutes les copies et le supprimer de votre syst?me
> informatique. If you can not understand this clause, please contact us for
> further information because it contain a legal notice.
> 
> 
> -----Message d'origine-----
> De : asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] De la part de Steve
> Underwood
> Envoy? : 26 ao?t 2004 06:12
> ? : Asterisk Users Mailing List - Non-Commercial Discussion
> Objet : Re: [Asterisk-Users] Faxing with SPANDSP or any other mean ? Is
> itpossible ? Am I dreaming ?
> 
> Stopping in mid page is usually a timing problem. See the spandsp FAQ.
> 
> Regards,
> Steve
> 
> 
> Jean-Fran?ois Rousseau wrote:
> 
>> Hi , does anybody have successfully received a full fax with spandsp ?
>> I keep having only about a quarter of the page and then the other part
>> is garbage. Does anybody have any solution for this ?
>> 
>> Right now I've tried:
>> 
>> FAX ---> IAXy ---> ASTERISK ---> SPANDSP
>> 
>> And
>> 
>> FAX ---> PSTN ---> X100P --> ASTERISK ---> SPANDSP
>> 
>> 
>> And both don't work, they give me only part of the page
>> 
>> 
>> 
>> 
>> BTW, I also tried the fax on a local lan over an IAXy or on the PSTN
>> with an X100P. Is there something I should know about faxing and theses
>> two interfaces ?
>> 
>> I also tried to Fax thru asterisk and it didn't work either   FAX ---->
> IAXy
>> ---> ASTERISK ---> X100P ---> PSTN ---> FAX
>> 
>> Finally my last test: FAX --> IAXy --> ASTERISK --> SIP (Iconnecthere)
>> --> PSTN --> FAX didn't work too.
>> 
>> Is there something I should know about faxing and Asterisk ? Should I
>> use a Sipura SIP FXS ?
>> 
>> P.S. I did start the ntp server to make sure timing was ok.
>> 
>> Thanks in advance
>> 
>> ___________________________
>> Jean-Fran?ois Rousseau
>> Sys-Tech
>> www.sys-tech.net
>> jrousseau at sys-tech.net
>> 
>> 
>> 
>> _______________________________________________
>> 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
>> 
>>  
>> 
> 
> _______________________________________________
> 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: 12
> Date: Fri, 27 Aug 2004 14:04:31 -0400
> From: Jean-Fran?ois Rousseau <jrousseau at sys-tech.net>
> Subject: RE: [Asterisk-Users] Faxing with SPANDSP or any other mean ?
> Is    itpossible ? Am I dreaming ?
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <asterisk-users at lists.digium.com>
> Message-ID: <20040827180458.BE6D02FE454 at lists.digium.com>
> Content-Type: text/plain;    charset="iso-8859-1"
> 
> Hi,
> 
> I've read the FAQ and tried to find a timing source. So far, I've compiled
> ztdummy and loaded it sucessfully. But it still not working. All I have is
> the beginning of the fax.
> 
> I've tried (HP FAX) --> PSTN --> X100P --> Asterisk -- > SPANDSP
> 
> And (HP FAX) --> IAXy --> Asterisk --> SPANDSP
> 
> Both do the same error...  About a quarter of the page is ok then garbage.
> The sending machine say that the fax was sent ok.
> 
> Here is some info that might help troubleshot my problem.
> 
> 
> 
> 
> 
> ___________________________
> Jean-Fran?ois Rousseau
> Sys-Tech
> www.sys-tech.net
> jrousseau at sys-tech.net
> 
> 
> -----Message d'origine-----
> De : asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] De la part de Steve
> Underwood
> Envoy? : 26 ao?t 2004 06:12
> ? : Asterisk Users Mailing List - Non-Commercial Discussion
> Objet : Re: [Asterisk-Users] Faxing with SPANDSP or any other mean ? Is
> itpossible ? Am I dreaming ?
> 
> Stopping in mid page is usually a timing problem. See the spandsp FAQ.
> 
> Regards,
> Steve
> 
> 
> Jean-Fran?ois Rousseau wrote:
> 
>> Hi , does anybody have successfully received a full fax with spandsp ?
>> I keep having only about a quarter of the page and then the other part
>> is garbage. Does anybody have any solution for this ?
>> 
>> Right now I've tried:
>> 
>> FAX ---> IAXy ---> ASTERISK ---> SPANDSP
>> 
>> And
>> 
>> FAX ---> PSTN ---> X100P --> ASTERISK ---> SPANDSP
>> 
>> 
>> And both don't work, they give me only part of the page
>> 
>> 
>> 
>> 
>> BTW, I also tried the fax on a local lan over an IAXy or on the PSTN
>> with an X100P. Is there something I should know about faxing and theses
>> two interfaces ?
>> 
>> I also tried to Fax thru asterisk and it didn't work either   FAX ---->
> IAXy
>> ---> ASTERISK ---> X100P ---> PSTN ---> FAX
>> 
>> Finally my last test: FAX --> IAXy --> ASTERISK --> SIP (Iconnecthere)
>> --> PSTN --> FAX didn't work too.
>> 
>> Is there something I should know about faxing and Asterisk ? Should I
>> use a Sipura SIP FXS ?
>> 
>> P.S. I did start the ntp server to make sure timing was ok.
>> 
>> Thanks in advance
>> 
>> ___________________________
>> Jean-Fran?ois Rousseau
>> Sys-Tech
>> www.sys-tech.net
>> jrousseau at sys-tech.net
>> 
>> 
>> 
>> _______________________________________________
>> 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
>> 
>>  
>> 
> 
> _______________________________________________
> 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: 13
> Date: Fri, 27 Aug 2004 14:07:52 -0400
> From: Jean-Fran?ois Rousseau <jrousseau at sys-tech.net>
> Subject: RE: [Asterisk-Users] Faxing with SPANDSP or any other mean ?
> Is    itpossible ? Am I dreaming ?
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <asterisk-users at lists.digium.com>
> Message-ID: <20040827180820.9E5E42FE60B at lists.digium.com>
> Content-Type: text/plain;    charset="iso-8859-1"
> 
> Hi,
> 
> I've read the FAQ and tried to find a timing source. So far, I've compiled
> ztdummy and loaded it sucessfully. But it still not working. All I have is
> the beginning of the fax.
> 
> I've tried (HP FAX) --> PSTN --> X100P --> Asterisk -- > SPANDSP
> 
> And (HP FAX) --> IAXy --> Asterisk --> SPANDSP
> 
> Both do the same error...  About a quarter of the page is ok then garbage.
> The sending machine say that the fax was sent ok.
> 
> Here is some info that might help troubleshot my problem.
> 
> #lsmod
> Module                  Size  Used by    Not tainted
> wcfxo                   8416   3
> ztdummy                 1928   0  (unused)
> zaptel                177792  10  [wcfxo ztdummy]
> usb-uhci               24496   0  [ztdummy]
> usbcore                65260   1  [usb-uhci]
> hisax                 517072   0  (unused)
> isdn                  126924   0  [hisax]
> slhc                    5168   0  [isdn]
> ide-scsi               10416   0
> 8139too                16072   1
> mii                     2432   0  [8139too]
> crc32                   2880   0  [8139too]
> 3c509                  11572   1
> isa-pnp                32688   0  [hisax 3c509]
> agpgart                47364   0  (unused)
> 
> cat /proc/interrupts
>          CPU0
> 0:     150695          XT-PIC  timer
> 1:          2          XT-PIC  keyboard
> 2:          0          XT-PIC  cascade
> 5:       3408          XT-PIC  eth0
> 8:          1          XT-PIC  rtc
> 9:    1478580          XT-PIC  wcfxo
> 10:    1478684          XT-PIC  wcfxo
> 11:    1478767          XT-PIC  wcfxo
> 14:       4798          XT-PIC  ide0
> 15:    1499931          XT-PIC  eth1, ztdummy, usb-uhci
> NMI:          0
> LOC:     150656
> ERR:          0
> MIS:          0
> 
> ___________________________
> Jean-Fran?ois Rousseau
> Sys-Tech
> www.sys-tech.net
> jrousseau at sys-tech.net
> T?l. 24h (418) 520-0739
> T?lec.???(418) 520-4554
> Ligne directe 1-866-274-4870
> B?tisseurs de solutions informatiques et ?lectroniques
> urgence at sys-tech.net
> 1-877-969-tech
> Messages de confidentialit? : Ce courriel (de m?me que les fichiers joints)
> est strictement r?serv? ? l'usage de la personne ou de l'entit? ? qui il est
> adress? et peut contenir de l'information privil?gi?e et confidentielle.
> Toute divulgation, distribution ou copie de ce courriel est strictement
> prohib?e. Si vous avez re?u ce courriel par erreur, veuillez nous en aviser
> sur-le-champ, d?truire toutes les copies et le supprimer de votre syst?me
> informatique. If you can not understand this clause, please contact us for
> further information because it contain a legal notice.
> 
> 
> -----Message d'origine-----
> De : asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] De la part de Steve
> Underwood
> Envoy? : 26 ao?t 2004 06:12
> ? : Asterisk Users Mailing List - Non-Commercial Discussion
> Objet : Re: [Asterisk-Users] Faxing with SPANDSP or any other mean ? Is
> itpossible ? Am I dreaming ?
> 
> Stopping in mid page is usually a timing problem. See the spandsp FAQ.
> 
> Regards,
> Steve
> 
> 
> Jean-Fran?ois Rousseau wrote:
> 
>> Hi , does anybody have successfully received a full fax with spandsp ?
>> I keep having only about a quarter of the page and then the other part
>> is garbage. Does anybody have any solution for this ?
>> 
>> Right now I've tried:
>> 
>> FAX ---> IAXy ---> ASTERISK ---> SPANDSP
>> 
>> And
>> 
>> FAX ---> PSTN ---> X100P --> ASTERISK ---> SPANDSP
>> 
>> 
>> And both don't work, they give me only part of the page
>> 
>> 
>> 
>> 
>> BTW, I also tried the fax on a local lan over an IAXy or on the PSTN
>> with an X100P. Is there something I should know about faxing and theses
>> two interfaces ?
>> 
>> I also tried to Fax thru asterisk and it didn't work either   FAX ---->
> IAXy
>> ---> ASTERISK ---> X100P ---> PSTN ---> FAX
>> 
>> Finally my last test: FAX --> IAXy --> ASTERISK --> SIP (Iconnecthere)
>> --> PSTN --> FAX didn't work too.
>> 
>> Is there something I should know about faxing and Asterisk ? Should I
>> use a Sipura SIP FXS ?
>> 
>> P.S. I did start the ntp server to make sure timing was ok.
>> 
>> Thanks in advance
>> 
>> ___________________________
>> Jean-Fran?ois Rousseau
>> Sys-Tech
>> www.sys-tech.net
>> jrousseau at sys-tech.net
>> 
>> 
>> 
>> _______________________________________________
>> 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
>> 
>>  
>> 
> 
> _______________________________________________
> 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
> 
> 
> 
> ------------------------------
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> End of Asterisk-Users Digest, Vol 1, Issue 5082
> ***********************************************
> 




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