Hello again,
I am using mix monitor and the majority of the sound records perfectly. I then get a "Internal Data Stream Error" near the end of the sound file. Has anyone ever seen this? I am allowing the ULAW amd ALAW codecs and an example dialplan entry is ; ; phone line phone1 exten => phone1,1,Answer() exten => phone1,2,MixMonitor(test.wav|av(0)V(0)) exten => phone1,3,Dial(SIP/phone1) exten => phone1,4,Busy(10) exten => phone1,5,Hangup() Many many thanks -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 23 October 2007 01:31 To: [email protected] Subject: asterisk-users Digest, Vol 39, Issue 76 *** WARNING *** This mail has originated outside your organization, either from an external partner or the Global Internet. Keep this in mind if you answer this message. Send asterisk-users mailing list submissions to [email protected] 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 [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than "Re: Contents of asterisk-users digest..." Today's Topics: 1. Re: A linksys SPA921 behind NAT and firewall ([EMAIL PROTECTED]) 2. Re: Making Asterisk a "Voice Router" (end1r) 3. Split asterisk in two ?? One TDM and One IP only?? (Steven) 4. Authenticate by IP? (Carlos Chavez) 5. Polycom 601 + Headset (Dovid B) 6. Re: tech prefix (Philipp Kempgen) 7. Re: Authenticate by IP? ([EMAIL PROTECTED]) 8. [France CID] Does Zaptel support ETSI FSK? (Vincent) 9. Re: Authenticate by IP? (Rurouni Alucard) 10. Re: Prompting for number when CID number not sent? (Vincent) 11. dial-out call queue (Joao Pereira) 12. Re: Authenticate by IP? (Carlos Chavez) 13. Split asterisk in two ?? One TDM and One IP only?? (BerkHolz, Steven) 14. Re: [France CID] Does Zaptel support ETSI FSK? (Vincent) 15. Re: Extensions.conf for basic IVR? (Vincent) 16. Re: 16 ports wanted (Christian Victor) 17. Re: Extensions.conf for basic IVR? (Erik Anderson) 18. Re: [France CID] Does Zaptel support ETSI FSK? (Jared Smith) 19. Re: Authenticate by IP? (Carlos Chavez) 20. Re: Extensions.conf for basic IVR? (Vincent) 21. Re: Authenticate by IP? ([EMAIL PROTECTED]) 22. Re: Authenticate by IP? (Victor Toofic) 23. bristuff: music on hold but no dialoptions tT defined. (Thomas Winter) 24. Re: [France CID] Does Zaptel support ETSI FSK? (Vincent) 25. Re: G729a codecs + Asterisk 1.4.11 (bilal ghayyad) 26. Re: [France CID] Does Zaptel support ETSI FSK? (Ira) 27. Voicemail playback on iPhone (Jason Lixfeld) 28. NAT traversal packet loss measurement (Yitzhak Bar Geva) 29. Re: Voicemail playback on iPhone (Ron Stephan) 30. Re: NAT traversal packet loss measurement (Matt Riddell) ---------------------------------------------------------------------- Message: 1 Date: Mon, 22 Oct 2007 13:25:20 -0400 From: "[EMAIL PROTECTED]" <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] A linksys SPA921 behind NAT and firewall To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=UTF-8 Check out again http://spc.pifiu.com it seems the owner of the site has added the latest admin guide for SPA-900 series & the spc.exe for 5.1.5 & 5.1.7 firmware. On 10/21/07, Per Jessen <[EMAIL PROTECTED]> wrote: > [EMAIL PROTECTED] wrote: > > > If you are trying to use non-complied ("XML") profiles... don't even > > bother wasting your time. > > Oh. I _am_ using the XML format. When I initiate a resync over the > http server, it works fine, except the SPA doesn't start the regular > resync. > > > > /Per Jessen, Z?rich ------------------------------ Message: 2 Date: Mon, 22 Oct 2007 14:05:01 -0400 From: "end1r" <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Making Asterisk a "Voice Router" To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="utf-8" Is this free? I see the tuner is free.. but the speech rec isn?t? -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of lenz Sent: Monday, October 22, 2007 11:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Making Asterisk a "Voice Router" Nice job! I took the liberty to post it on AstPligg as well: http://tinyurl.com/268bac Thanks l. In data Mon, 22 Oct 2007 16:22:13 +0200, Jared Smith <[EMAIL PROTECTED]> ha scritto: > On Mon, 2007-10-22 at 09:39 -0400, end1r wrote: >> I?m interested in what software (Free or course) that people use when >> they want to add a ?dial by voice? service to their asterisk system. >> Meaning I pick up the phone.. dial some extension? it prompts me for >> name.. I say ?John Smith?.. and it dials his extension and connects >> the call.. > > I've done this using Asterisk and the LumenVox speech engine... in > fact, I spoke about it at AstriCon Europe in 2006. My slides are > available at http://www.astricon.net/files/Jared_Smith_EUR06.pdf. > (They may be slightly out of date, but it should at least get you > started.) > > -- Home of QueueMetrics - http://queuemetrics.com _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ------------------------------ Message: 3 Date: Mon, 22 Oct 2007 14:09:49 -0400 From: "Steven" <[EMAIL PROTECTED]> Subject: [asterisk-users] Split asterisk in two ?? One TDM and One IP only?? To: [email protected] Message-ID: <[EMAIL PROTECTED]> I have built an asterisk server with a TE412P card on a Dell 2950. It does incoming calls via DID over PRI, our IVR, SIP/IAX extensions, Fax/Analog extensions via an old PBX via PRI, voicemail, etc. My issue now is that I find it difficult to test/upgrade to new versions. This is what I am thinking of doing. Server1 Keep one physical server just for TDM functions. PRI to Telco PRI to old PBX for Fax. (basically using it as a mux) Keep meetme here for Digium card timing. Server2 Build a new asterisk install within Xen VM with data stored on an iSCSI SAN. This would be all IP. IAX and SIP extensions. IAX and SIP providers. IVR Voicemail Web access to voicemail CDR This way I can test different versions of the features of Server2 (clone with different IP) without affecting production. I assume that I just use an IAX or SIP trunk between the two asterisk servers. Does this make sense? Are others doing similar? Are there any other features that require the TDM card besides PRI, Fax and Meetme? I have heard of people using Xen for IP only asterisk, but are there any known gotchas? Thanks, -- -- Steven http://www.glimasoutheast.org ------------------------------ Message: 4 Date: Mon, 22 Oct 2007 13:24:31 -0500 From: Carlos Chavez <[EMAIL PROTECTED]> Subject: [asterisk-users] Authenticate by IP? To: Asterisk <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="utf-8" I have a customer that needs an Asterisk server to sell minutes for cell phones in Mexico. I do not see a problem with that since he will get the calls by SIP and then use GSM adapters to get the calls into the GSM network. My problem is that his customers only want to be identified by IP and not by a username and password. Is there a way to authenticate just by using an IP address? -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20071022/44 0cf5e6/attachment-0001.pgp ------------------------------ Message: 5 Date: Mon, 22 Oct 2007 20:31:19 +0200 From: "Dovid B" <[EMAIL PROTECTED]> Subject: [asterisk-users] Polycom 601 + Headset To: <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1" Hi List, I am using a Plantronics CS50 head set with my Polycom 601. I use the button on it to pick up calls. Is there any way to have the phone set up that if I pick up with the button on the headset that it sends the call to the headset and that I don't have to press the headset button on the phone every time ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071022/87 300565/attachment-0001.htm ------------------------------ Message: 6 Date: Mon, 22 Oct 2007 21:00:17 +0200 From: Philipp Kempgen <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] tech prefix To: Asterisk Users <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-15 Jon Weisman wrote: > Here's what worked: > > exten=>_X.,1,Dial(SIP/"prefix"[EMAIL PROTECTED] trunk) > > substitute "prefix" for the tech prefix you would like to append. > ----- Original Message ----- > From: "Philipp Kempgen" <[EMAIL PROTECTED]> > To: "Asterisk Users" <[email protected]> > Sent: Tuesday, October 16, 2007 3:09 PM > Subject: Re: [asterisk-users] tech prefix > > > Jon Weisman wrote: > > How can I add a prefix to an outbound call? > > _X. => { > Dial(tech/123{EXTEN}); > } That's what I said. 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 Gesch?ftsf?hrer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ------------------------------ Message: 7 Date: Mon, 22 Oct 2007 15:13:04 -0400 From: "[EMAIL PROTECTED]" <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Authenticate by IP? To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=UTF-8 On 10/22/07, Carlos Chavez <[EMAIL PROTECTED]> wrote: > I have a customer that needs an Asterisk server to sell minutes for > cell phones in Mexico. I do not see a problem with that since he will > get the calls by SIP and then use GSM adapters to get the calls into the > GSM network. My problem is that his customers only want to be > identified by IP and not by a username and password. Is there a way to > authenticate just by using an IP address? > There certainly is. ------------------------------ Message: 8 Date: Mon, 22 Oct 2007 21:19:27 +0200 From: Vincent <[EMAIL PROTECTED]> Subject: [asterisk-users] [France CID] Does Zaptel support ETSI FSK? To: [email protected] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=us-ascii Hello I've been googling for a couple of days now, but still can't figure out what to put in zapata.conf to get it to report CID. Unless I'm mistaken, France uses ETSI FSK for CID method and bell 202 as CID FSK Standard: http://img219.imageshack.us/img219/7207/linksys3102cid1jj7.jpg http://img219.imageshack.us/img219/4625/linksys3102cid2ld5.jpg Does Zaptel support those on Digium TDM400 clones like those from OpenVox? Thank you. ------------------------------ Message: 9 Date: Mon, 22 Oct 2007 15:35:40 -0400 From: Rurouni Alucard <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Authenticate by IP? To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="utf-8" Saludos Carlos, Como vas a recibir las llamadas via SIP, puedes especificar el IP del host que te enviara las llamadas, por ej. Este es un bloque que tengo definido en el SIP.conf de uno de mis servers para enrutar las llamadas internacionales y a telefonos moviles utilizando un proveedor de terminacion. [oficina] type=peer context=from_office ; Esto va a mi 'extensions.conf' host=200.88.42.29 ; Este es el ip publico en la oficina (estatico) nat=no canreinvite=no qualify=yes disallow=all allow=g729 allow=ulaw Creo que eso contesta tu pregunta. -- Jose P. Espinal slackware-es.com Carlos Chavez wrote: > I have a customer that needs an Asterisk server to sell minutes for > cell phones in Mexico. I do not see a problem with that since he will > get the calls by SIP and then use GSM adapters to get the calls into the > GSM network. My problem is that his customers only want to be > identified by IP and not by a username and password. Is there a way to > authenticate just by using an IP address? > > > ------------------------------------------------------------------------ > > _______________________________________________ > --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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071022/d9 478cc1/attachment-0001.htm ------------------------------ Message: 10 Date: Mon, 22 Oct 2007 21:56:09 +0200 From: Vincent <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Prompting for number when CID number not sent? To: [email protected] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=us-ascii On Mon, 22 Oct 2007 10:14:41 -0400, Jared Smith <[EMAIL PROTECTED]> wrote: >Instead of ${callerid} here (which probably isn't working for you >anyway), you probably want to use the CALLERID dialplan function to >retrieve the CallerID number, like this: Thanks for the tip. It'll come in handy... once I finally get the TDM card to report CID :-) ------------------------------ Message: 11 Date: Mon, 22 Oct 2007 21:57:47 +0100 From: Joao Pereira <[EMAIL PROTECTED]> Subject: [asterisk-users] dial-out call queue To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Is it possible to implement a dial-out call queue in Asterisk? My idea is to give Asterisk a list of numbers, and then he makes the calls and delivers the calls to a call queue. Then, the agents will answer the calls. Is this possible? Thanks Regards Joao pereira ------------------------------ Message: 12 Date: Mon, 22 Oct 2007 15:59:18 -0500 From: Carlos Chavez <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Authenticate by IP? To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="utf-8" On Mon, 2007-10-22 at 15:35 -0400, Rurouni Alucard wrote: > Saludos Carlos, > > Como vas a recibir las llamadas via SIP, puedes especificar el IP del > host que te enviara las llamadas, por ej. > > Este es un bloque que tengo definido en el SIP.conf de uno de mis > servers para enrutar las llamadas internacionales y a telefonos > moviles utilizando un proveedor de terminacion. > > [oficina] > type=peer > context=from_office ; Esto va a mi 'extensions.conf' > host=200.88.42.29 ; Este es el ip publico en la oficina (estatico) > > nat=no > canreinvite=no > qualify=yes > disallow=all > allow=g729 > allow=ulaw > > Creo que eso contesta tu pregunta. > > Hola Jos?. Gracias por tu contestaci?n. Lo que me estas especificando el para hacer llamadas de salida (PEER). Yo necesito autentificar a un usuario de entrada, voy a intentar haciendo algo parecido solo cambiando a type=user para ver si as? funciona. -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20071022/73 97e46b/attachment-0001.pgp ------------------------------ Message: 13 Date: Mon, 22 Oct 2007 17:12:54 -0400 From: "BerkHolz, Steven" <[EMAIL PROTECTED]> Subject: [asterisk-users] Split asterisk in two ?? One TDM and One IP only?? To: "[email protected]" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1" I have built an asterisk server with a TE412P card on a Dell 2950. It does incoming calls via DID over PRI, our IVR, SIP/IAX extensions, Fax/Analog extensions via an old PBX via PRI, voicemail, etc. My issue now is that I find it difficult to test/upgrade to new versions. This is what I am thinking of doing. Server1 Keep one physical server just for TDM functions. PRI to Telco PRI to old PBX for Fax. (basically using it as a mux) Keep meetme here for Digium card timing. Server2 Build a new asterisk install within Xen VM with data stored on an iSCSI SAN. This would be all IP. IAX and SIP extensions. IAX and SIP providers. IVR Voicemail Web access to voicemail CDR This way I can test different versions of the features of Server2 (clone with different IP) without affecting production. I assume that I just use an IAX or SIP trunk between the two asterisk servers. Does this make sense? Are others doing similar? Are there any other features that require the TDM card besides PRI, Fax and Meetme? I have heard of people using Xen for IP only asterisk, but are there any known gotchas? Thanks, Thank You, Steven BerkHolz ------------------------------ Message: 14 Date: Mon, 22 Oct 2007 23:18:06 +0200 From: Vincent <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] [France CID] Does Zaptel support ETSI FSK? To: [email protected] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=us-ascii On Mon, 22 Oct 2007 21:19:27 +0200, Vincent <[EMAIL PROTECTED]> wrote: >Does Zaptel support those on Digium TDM400 clones like those from >OpenVox? Pff, finally found what it was: It had nothing to do with zaptel, and everything to do with extensions.conf: ======== exten => s,1,NoOp(Got a call) ;nothing displayed exten => s,n,Verbose(${CALLERID}) exten => s,n,Verbose(${CALLERIDNAME}) exten => s,n,Verbose(${CALLERIDNUM}) exten => s,n,NoOp(${CALLERID}) exten => s,n,Verbose(${CALLERID}) ;CID at last! exten => s,n,Verbose(${CALLERID(num)}) ======== I'm running Asterisk 1.4. Does someone know why only the last statement does display the CID number while the others print nothing? Thank you. ------------------------------ Message: 15 Date: Mon, 22 Oct 2007 23:20:19 +0200 From: Vincent <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Extensions.conf for basic IVR? To: [email protected] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=us-ascii On Mon, 22 Oct 2007 09:06:00 +0200, randulo <[EMAIL PROTECTED]> wrote: >The first ten sites that come up, including voip-info.org, usually a >good place to look first, each have full examples. Look also for the >background application wich is used to play the file, get input and >jump to the extension entered. Thanks. The problem with information on the Net is that the development of Asterisk moves quite fast, making some/a lot of information obsolete, something newbies aren't necessarily aware of. 2008 might be a good year to update "* - The future of telephony" :-) ------------------------------ Message: 16 Date: Mon, 22 Oct 2007 23:33:34 +0200 From: Christian Victor <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] 16 ports wanted To: Gergo Csibra <[EMAIL PROTECTED]>, Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Gergo Csibra schrieb: > Well, using more than one TDM card in your PC is not a good idea, > because of interrupts. If you have to have 16 FXO you can more > options: > > 1. Using TDM2400P with 4 FXO modules ($1775) > 2. Using Xorcom's Astribank (external) ($1170) > 3. Using some T1/E1 card with Channel Bank (more expensive) > 4. Using Sangoma's A200 with 8 (up to 12) dual-FXO modules (ca. $1.200) 5. Using Sangoma's A400 with 8 (up to 24) dual-FXO modules (ca. $1.350) Both Sangoma cards can be equipped with Octasic hardware-EC for ca. $300 more and are available in PCI(-X) and PCIexpress versions. For the A200 you need an additional case slot (does not need another PCI connector) for every 4 ports over 4. The same goes for the A400 on every 12 ports over 12. Christian ------------------------------ Message: 17 Date: Mon, 22 Oct 2007 16:41:19 -0500 From: "Erik Anderson" <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Extensions.conf for basic IVR? To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1 On 10/22/07, Vincent <[EMAIL PROTECTED]> wrote: > > 2008 might be a good year to update "* - The future of telephony" :-) Version 2 of TFOT was just released a few weeks ago... http://downloads.oreilly.com/books/9780596510480.pdf -- Erik Anderson http://andersonfam.org ------------------------------ Message: 18 Date: Mon, 22 Oct 2007 17:57:44 -0400 From: Jared Smith <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] [France CID] Does Zaptel support ETSI FSK? To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain On Mon, 2007-10-22 at 23:18 +0200, Vincent wrote: > ======== > exten => s,1,NoOp(Got a call) > > ;nothing displayed > exten => s,n,Verbose(${CALLERID}) > exten => s,n,Verbose(${CALLERIDNAME}) > exten => s,n,Verbose(${CALLERIDNUM}) > exten => s,n,NoOp(${CALLERID}) > exten => s,n,Verbose(${CALLERID}) > > ;CID at last! > exten => s,n,Verbose(${CALLERID(num)}) > ======== > > I'm running Asterisk 1.4. Does someone know why only the last > statement does display the CID number while the others print nothing? Beginning with Asterisk 1.4, we moved all of the CallerID functionality from channel variables and applications to a single CALLERID dialplan function. This should have been noted in UPGRADE.txt. I also tried to warn you about it in my last email in this thread, but I guess I should have been more specific. -- Jared Smith Community Relations Manager Digium, Inc. ------------------------------ Message: 19 Date: Mon, 22 Oct 2007 16:00:19 -0500 From: Carlos Chavez <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Authenticate by IP? To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="utf-8" On Mon, 2007-10-22 at 15:13 -0400, [EMAIL PROTECTED] wrote: > On 10/22/07, Carlos Chavez <[EMAIL PROTECTED]> wrote: > > I have a customer that needs an Asterisk server to sell minutes for > > cell phones in Mexico. I do not see a problem with that since he will > > get the calls by SIP and then use GSM adapters to get the calls into the > > GSM network. My problem is that his customers only want to be > > identified by IP and not by a username and password. Is there a way to > > authenticate just by using an IP address? > > > > There certainly is. > And could you please point me in the right direction? -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20071022/c0 ad07f8/attachment-0001.pgp ------------------------------ Message: 20 Date: Tue, 23 Oct 2007 00:04:14 +0200 From: Vincent <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Extensions.conf for basic IVR? To: [email protected] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=us-ascii On Mon, 22 Oct 2007 16:41:19 -0500, "Erik Anderson" <[EMAIL PROTECTED]> wrote: >Version 2 of TFOT was just released a few weeks ago... Just had to ask :-) Thanks. ------------------------------ Message: 21 Date: Mon, 22 Oct 2007 18:06:41 -0400 From: "[EMAIL PROTECTED]" <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Authenticate by IP? To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=UTF-8 La configuraci?n de Jose esta correcta. Cuando usas un "peer" en sip.conf Asterisk usa el hostname or el IP para autenticar. Cuando usas un "user" la autenticaci?n se basa en el usuario y la contrase?a, cual en su caso no existe. On 10/22/07, Carlos Chavez <[EMAIL PROTECTED]> wrote: > Hola Jos?. Gracias por tu contestaci?n. Lo que me estas especificando > el para hacer llamadas de salida (PEER). Yo necesito autentificar a un > usuario de entrada, voy a intentar haciendo algo parecido solo cambiando > a type=user para ver si as? funciona. > > -- > Telecomunicaciones Abiertas de M?xico S.A. de C.V. > Carlos Ch?vez Prats > Director de Tecnolog?a > +52-55-91169161 ext 2001 > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > ------------------------------ Message: 22 Date: Mon, 22 Oct 2007 17:07:18 -0500 From: Victor Toofic <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Authenticate by IP? To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=iso-8859-1 El Mon, Oct 22 de 2007 a las 15:59 -0500, Carlos Chavez comentaba: > Hola Jos?. Gracias por tu contestaci?n. Lo que me estas especificando > el para hacer llamadas de salida (PEER). Yo necesito autentificar a un > usuario de entrada, voy a intentar haciendo algo parecido solo cambiando > a type=user para ver si as? funciona. type=peer also works for incoming calls. In this case (peer) asterisk only checks the IP the call is coming from and uses the context you defined there. If you use type=user you will need to specify a username and a secret. -- Greetings.. V?ctor Toofic ------------------------------ Message: 23 Date: Tue, 23 Oct 2007 00:11:39 +0200 From: Thomas Winter <[EMAIL PROTECTED]> Subject: [asterisk-users] bristuff: music on hold but no dialoptions tT defined. To: [email protected] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="utf-8" Hi, Iam dialing from NT ptp to SIP provider. Sometimes Asterisk is doing music on hold but there are no options like t or T in the dial command. As an result the channel got lost and an Hangup occurs. Iam using bristuff-0.3.0-PRE-1y-i on an QuadBri card. Any solution for this? Oct 22 11:20:06 VERBOSE[911] logger.c: -- SIP/sip-08206a30 answered Zap/8-1 Oct 22 11:20:23 VERBOSE[29983] logger.c: -- Started music on hold, class 'default', on channel 'Zap/8-1' Oct 22 11:20:46 VERBOSE[29983] logger.c: -- Stopped music on hold on Zap/8-1 Oct 22 11:20:46 VERBOSE[29983] logger.c: -- Started music on hold, class 'default', on channel 'Zap/8-1' Oct 22 11:20:55 VERBOSE[911] logger.c: == Spawn extension (macro-call, s, 2) exited non-zero on 'Zap /8-1' in macro 'tmp_call' ------------------------------ Message: 24 Date: Tue, 23 Oct 2007 00:15:28 +0200 From: Vincent <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] [France CID] Does Zaptel support ETSI FSK? To: [email protected] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=us-ascii On Mon, 22 Oct 2007 17:57:44 -0400, Jared Smith <[EMAIL PROTECTED]> wrote: >Beginning with Asterisk 1.4, we moved all of the CallerID functionality >from channel variables and applications to a single CALLERID dialplan >function. This should have been noted in UPGRADE.txt. I also tried to >warn you about it in my last email in this thread, but I guess I should >have been more specific. No problem. I should have read it more closely, but due to the number of people having problems with Zaptel and CID, I was focused on that part. Should have started asking people what the correct way was to read CID information in Asterisk 1.4... Thanks. ------------------------------ Message: 25 Date: Mon, 22 Oct 2007 16:05:06 -0700 (PDT) From: bilal ghayyad <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] G729a codecs + Asterisk 1.4.11 To: [email protected] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=iso-8859-1 Dear Marc; I readed your email about the codec G729a and I am now also need to install the codec on my Asterisk. I typed from Asterisk CLI: core show version and I got the following: Asterisk SVN-branch-1.4-r72556 built by root @ localhost.localdomain on a i686 running Linux on 2007-06-30 13:08:08 UTC So I beleive that my processor is i686, correct? But I am not able to know which one to download: The x86-32 or x86-64 ? Can you please advise. Also, the nocona or the opteron versions? Regards Bilal ------------------- Good Morning, Any help would be grateful to help me understanding what's wrong... I have bought 2 g729a licenses to digium and I would like to have them works... My processor is an Intel(R) Xeon(R) CPU E5310 @ 1.60GHz (4 processors) so I have downloaded the http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.4/x86-64 /codec_g729a_v32_nocona.tar.gz codec I have registered my license, copied the codec_g729a.so into the /usr/lib/asterisk/modules folder and restarted my asterisk But on the CLI when I type asterisk*CLI> show modules like 72 Module Description Use Count codec_g726.so ITU G.726-32kbps G726 Transcoder 0 format_g729.so Raw G729 data 0 format_g726.so Raw G.726 (16/24/32/40kbps) data 0 format_g723.so G.723.1 Simple Timestamp File Format 0 The codec_g729a.so doesn't appear.......... Any idea how to solve the problem..... Thanks Best Regards, Marc LEURENT __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ------------------------------ Message: 26 Date: Mon, 22 Oct 2007 16:09:39 -0700 From: Ira <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] [France CID] Does Zaptel support ETSI FSK? To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii"; format=flowed At 02:18 PM 10/22/2007, you wrote: >;nothing displayed >exten => s,n,Verbose(${CALLERID}) >exten => s,n,Verbose(${CALLERIDNAME}) >exten => s,n,Verbose(${CALLERIDNUM}) >exten => s,n,NoOp(${CALLERID}) >exten => s,n,Verbose(${CALLERID}) > >;CID at last! >exten => s,n,Verbose(${CALLERID(num)}) >======== > >I'm running Asterisk 1.4. Does someone know why only the last >statement does display the CID number while the others print nothing? try adding a wait(1) right in the beginning, worked for me. Ira ------------------------------ Message: 27 Date: Mon, 22 Oct 2007 19:15:55 -0400 From: Jason Lixfeld <[EMAIL PROTECTED]> Subject: [asterisk-users] Voicemail playback on iPhone To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=US-ASCII; format=flowed Anyone managed to get this to work? What's the recipe? ------------------------------ Message: 28 Date: Tue, 23 Oct 2007 01:35:13 +0200 From: "Yitzhak Bar Geva" <[EMAIL PROTECTED]> Subject: [asterisk-users] NAT traversal packet loss measurement To: [email protected] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1" How can one measure the effect of NAT traversal packet loss? We currently have no solution for NAT traversal for our SIP clients. There is no doubt that packets are getting lost. What is not clear is how much damage this does. On the face of it, everything seems fine. Could this be so? Perhaps we're suffering a degradation in quality or our call setup times could be improved. How can we measure this? What's the simplest method of preventing packet loss due to NAT traversal in a SIP environment? Thanks, Yitzhak -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071023/13 da9c41/attachment-0001.htm ------------------------------ Message: 29 Date: Mon, 22 Oct 2007 16:38:17 -0700 From: "Ron Stephan" <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Voicemail playback on iPhone To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii" Trick question I assume? It was mind numbingly simple on my iPhone...(though none of the voice mail worked when London a few weeks ago). - tap voice mail - - tap speaker (upper right) until it turns blue (is activate) - tap the message you want to playback - use assorted controls to delete - replay etc. Now...if the question is ... how do you get asterisk voice mail to show up on an iPhone...I am all ears. Groovy concept - if anybody has a hack - I'd love to see it. Elvis -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Lixfeld Sent: Monday, October 22, 2007 4:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Voicemail playback on iPhone Anyone managed to get this to work? What's the recipe? _______________________________________________ --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 __________ NOD32 2607 (20071022) Information __________ This message was checked by NOD32 antivirus system. http://www.eset.com ------------------------------ Message: 30 Date: Tue, 23 Oct 2007 13:30:55 +1300 From: Matt Riddell <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] NAT traversal packet loss measurement To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1 -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Yitzhak Bar Geva wrote: > How can one measure the effect of NAT traversal packet loss? > We currently have no solution for NAT traversal for our SIP clients. There > is no doubt that packets are getting lost. What is not clear is how much > damage this does. On the face of it, everything seems fine. Could this be > so? Perhaps we're suffering a degradation in quality or our call setup times > could be improved. How can we measure this? > What's the simplest method of preventing packet loss due to NAT traversal in > a SIP environment? NAT is unlikely to cause a percentage of packets to get lost. Normally you'd have one way audio if NAT was causing a problem (i.e. 100% packet loss). The only other situation in which it might happen is where the NAT router decides to close a port mapping (thereby blocking incoming calls to the customer's device). But if you're looking for packet loss there are a number of other things to check first. I wouldn't do VoIP across the WAN without at least some packet shaping but hey. - -- 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 iD8DBQFHHUC+DQNt8rg0Kp4RAv0uAJ9Q41eQ+7RuqzFvgtxEhQOIU0QFggCaAlkD GMVdY/n58wHsciuHihZCCHY= =6L87 -----END PGP SIGNATURE----- ------------------------------ _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users End of asterisk-users Digest, Vol 39, Issue 76 ********************************************** ******************************************************************** This email and any attachments are confidential to the intended recipient and may also be privileged. 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