Francisco Perez-Landaeta
2004-Sep-09 09:34 UTC
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 1, Issue 5082
Anyone using the recently MAC OS X ? Version of asterisk ? Thanks, Francisco Perez-Landaeta> From: asterisk-users-request@lists.digium.com > Reply-To: asterisk-users@lists.digium.com > Date: Fri, 27 Aug 2004 13:08:24 -0500 (CDT) > To: asterisk-users@lists.digium.com > Subject: Asterisk-Users Digest, Vol 1, Issue 5082 > > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > > You can reach the person managing the list at > asterisk-users-owner@lists.digium.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Asterisk-Users digest..." > > > 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@adelphia.net> > Subject: [Asterisk-Users] Re: how to fetch a call? (Tony Mountifield) > To: asterisk-users@lists.digium.com > Cc: tony@softins.clara.co.uk, roger@planinternet.de > Message-ID: <1093626430.2021.182.camel@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@softins.clara.co.uk (Tony Mountifield) >> Subject: [Asterisk-Users] Re: how to fetch a call? >> To: asterisk-users@lists.digium.com >> Message-ID: <cgnfpm$56k$1@softins.clara.co.uk> >> >> In article <412F4122.6070401@planinternet.de>, >> Roger Schreiter <roger@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@yahoo.co.jp> > Subject: [Asterisk-Users] Asterisk Assistants Custom Icon > To: astusr <asterisk-users@lists.digium.com> > Cc: chriss@watertech.com > Message-ID: <20040827171445.8897.qmail@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@skimmilk.net> > Subject: RE: [Asterisk-Users] Overhead Paging > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users@lists.digium.com> > Message-ID: <134401c48c5b$1e09ae40$c8fea8c0@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@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@freebsdcluster.net> > Subject: [Asterisk-Users] libr2 > To: asterisk-users@lists.digium.com > Message-ID: <20040827172229.GA40041@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@gendesign.com> > Subject: [Asterisk-Users] Speech Recognition and Asterisk > To: Asterisk Users Group <asterisk-users@lists.digium.com> > Message-ID: <1093627611.2871.391.camel@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@terra.com.br> > Subject: Re: [Asterisk-Users] FXOs > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: <023e01c48c5d$4161eb80$7a0aa8c0@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@mstvp.com> > To: <asterisk-users@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@pixelpower.com >> o(713)861-4005 >> o(800)905-6412 >> f(713)864-8668 >> c(713)201-1262 >> >> >> >> _______________________________________________ >> Asterisk-Users mailing list >> Asterisk-Users@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@GoRemote.com> > Subject: RE: [Asterisk-Users] sip change? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <C0E23A9B1942314FAEC0134C189CF79BC5B8B9@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@lists.digium.com > [mailto:asterisk-users-bounces@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@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@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@identitymine.com> > Subject: RE: [Asterisk-Users] sip change? > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: > <93D6FFFFE7DE7142962E7D4A331538E8034048@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@lists.digium.com > [mailto:asterisk-users-bounces@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@lists.digium.com > [mailto:asterisk-users-bounces@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@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@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@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@routers.com> > Subject: RE: [Asterisk-Users] Overhead Paging > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <Chameleon.1093629021.adar0@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@accessgate.net> > Subject: Re: [Asterisk-Users] sip change? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <412F760F.60800@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@lists.digium.com >> [mailto:asterisk-users-bounces@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@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@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@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 >> >> >> >> > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20040827/cd2184de > /attachment-0001.html > > ------------------------------ > > Message: 11 > Date: Fri, 27 Aug 2004 14:03:03 -0400 > From: Jean-Fran?ois Rousseau <jrousseau@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@lists.digium.com> > Message-ID: <20040827180331.2740B2FE60B@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@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@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@lists.digium.com > [mailto:asterisk-users-bounces@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@sys-tech.net >> >> >> >> _______________________________________________ >> Asterisk-Users mailing list >> Asterisk-Users@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@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@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@lists.digium.com> > Message-ID: <20040827180458.BE6D02FE454@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@sys-tech.net > > > -----Message d'origine----- > De : asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@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@sys-tech.net >> >> >> >> _______________________________________________ >> Asterisk-Users mailing list >> Asterisk-Users@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@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@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@lists.digium.com> > Message-ID: <20040827180820.9E5E42FE60B@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@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@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@lists.digium.com > [mailto:asterisk-users-bounces@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@sys-tech.net >> >> >> >> _______________________________________________ >> Asterisk-Users mailing list >> Asterisk-Users@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@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@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > > > End of Asterisk-Users Digest, Vol 1, Issue 5082 > *********************************************** >
Apparently Analagous Threads
Wisdom of the Ancients
