thanx andrew first of all your messages are in Plain Text format! i hv monitored Asterisk both managerAPI console and Asterisk main console to see wht is actually going on .when a new incoming connection comes.
when the phone is ringing.it gives starting simple swithc on 'ZAP/1-1' and on manager API i get newExten event with exten 's' from channel 'zap/1-1' i hvnt picked up my fone. i think its its ringing 7-8 times and asterisk doesnt do anything . ----- Original Message ----- From: <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, September 16, 2004 9:22 AM Subject: Asterisk-Users Digest, Vol 2, Issue 152 > 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: E3 PCI Cards (Noah Miller) > 2. spandsp on current cvs? (Rich Adamson) > 3. Re: Broadvoice BYOD Plans - 3-way and Call Waiting (Chris Shaw) > 4. Re: No Caller Name sent from Asterisk over National or > DMS100? (Jason Kawakami) > 5. RE: Intertex IX66 (Chris HARIGA) > 6. how to get caller ID (vrushank) > 7. Re: how to get caller ID (Andrew Thompson) > 8. Re: E3 PCI Cards (Benjamin on Asterisk Mailing Lists) > 9. Beyond T1 (Christopher Jacob) > 10. call parking & forwarding (Maros RAJNOCH) > 11. What can you do with Asterisk in Brazil following the law > (Johannes van Hulst) > 12. ID for outgoing calls from DDI (DID) line (Maros RAJNOCH) > 13. Re: Beyond T1 (Andrew Thompson) > 14. Re: Beyond T1 (Steven P. Donegan) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 16 Sep 2004 11:03:48 -0400 > From: Noah Miller <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] E3 PCI Cards > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[EMAIL PROTECTED]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=US-ASCII; format=flowed > > >> Another promising candidate is Apple's dual G5 (PPC970) Xserve (a 1U > >> server). > >> http://www.apple.com/xserve > >> this one looks as if it might beat the price/performance ratio of a > >> high end Intel server. > > > > The Apple G5 Xserv system has a "PCI-X" interface. Does anyone know > > what that is and will a T405P or T410P card work? > > > >> Both systems run LinuxPPC. > > > > Does anyone have * running on PPC? > > Yeah, check out: > > http://www.voip-info.org/tiki-index.php?page=Asterisk%20MacOSX%20Support > > Specifically for OS X. There's a download link. The problem still is > that no one has written ppc drivers for the Digium cards. As I > understand, the only drivers are for GNU/Linux on i386. You wanna > write some for the good of the BSD and PPC communities? ;-) > > > > > > > > > > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ------------------------------ > > Message: 2 > Date: Thu, 16 Sep 2004 10:06:39 -0600 > From: Rich Adamson <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] spandsp on current cvs? > To: Asterisk-a-users-list <[EMAIL PROTECTED]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1 > > > Steve or anyone... > > Will spandsp install on the current cvs? > > Looked like the code at ftp.opencall.org/pub/spandsp was intended > to be applied to the old stable release. Anyone know? > > Rich > > > > > ------------------------------ > > Message: 3 > Date: Thu, 16 Sep 2004 08:11:10 -0700 > From: "Chris Shaw" <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] Broadvoice BYOD Plans - 3-way and Call > Waiting > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[EMAIL PROTECTED]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="iso-8859-1" > > > The other issue is that call waiting does not appear to work. The way I'm > > expecting it to work with Asterisk is to send the second call to me - I'm > > using SetGroup and CheckGroup within Asterisk to limit my calls to two at > a > > time total. However, if I'm on a phone call (incoming or outgoing), > Broadvoice > > transfers a second call to a "person you are calling is busy" message -- I > > don't see any additional SIP traffic to the Asterisk box. > > You must have call waiting turned off on your comm pilot control panel, go > to www.broadvoice.com and log into your control panel and make sure call > waiting is turned on. > > -Chris > > > > ------------------------------ > > Message: 4 > Date: Thu, 16 Sep 2004 09:20:09 -0600 > From: "Jason Kawakami" <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] Re: No Caller Name sent from Asterisk over > National or DMS100? > To: <[EMAIL PROTECTED]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="iso-8859-1" > > > ----- Original Message ----- > > Message: 3 > > Date: Thu, 16 Sep 2004 07:57:15 -0400 (EDT) > > From: David Troy <[EMAIL PROTECTED]> > > Subject: Re: [Asterisk-Users] No Caller Name sent from Asterisk over > > National or DMS100 PRI to a Norstar MICS? > > snip> > > > I have a PRI link up and running between Asterisk and a Nortel Norstar > MICS > > > v4.1 . I'm having a problem getting the textual Caller Name across the > link > > > from Ast to Ns, however numeric Caller ID arrives and displays fine. > >From Ns > > > to Ast both elements come through fine. I'm forcing dummy values for > testing > > > using: > <snip> > > everyone remember that we are talking about a private connection here. if i > read the original post here correctly the issue is between the * and the > Norstar not out to the PSTN. > > i have been tying NEC's together for 15+ years with a proprietary ISDN > protocol that sends station name across the d-channel without any reverse > lookup DB. > > Now that being said I am no expert on d-channel messaging so I can't really > answer the question on how/if we can pass the CALLERIDNAME across a private > d-channel connection between * and another PBX. > > Jason Kawakami > www.optellabs.com > > > > ------------------------------ > > Message: 5 > Date: Thu, 16 Sep 2004 11:21:41 -0400 > From: "Chris HARIGA" <[EMAIL PROTECTED]> > Subject: RE: [Asterisk-Users] Intertex IX66 > To: "'Jason Williams'" <[EMAIL PROTECTED]>, "'Asterisk Users > Mailing List - Non-Commercial Discussion'" > <[EMAIL PROTECTED]> > Message-ID: > <!~!UENERkVCMDkAAQACAAAAAAAAAAAAAAAAABgAAAAAAAAAgvaz9VLBY0Wot+jOKFJUmMKAAAAQ [EMAIL PROTECTED]> > > Content-Type: text/plain; charset="us-ascii" > > Lolllllll, > > That's a good one :)) > > U make my day :) > > Best regards, > > Chris HARIGA > > P.S.: I send my ethereal log to Intertex.se and I hope to fix the problem > asap. I will post on the list the "solution". > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Jason Williams > Sent: Thursday, September 16, 2004 4:50 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Intertex IX66 > > On Thu, 9 Sep 2004 12:26:56 -0400, Chris HARIGA <[EMAIL PROTECTED]> > wrote: > > Hi, > > > > I have Asterisk w/ 192.168.1.1 and I setup IX66 to be 192.168.1.2 (I'm > using > > pppoe client and dyndns.org on IX66) > > I setup on Local DNS Server my * box and after that I was able to register > > my phones from the Internet. > > I cannot understand my problem with one way sound... what is wrong on my > > configuration :(( > > As the IX66 is a sip aware router make sure you have no entries for > nat in your sip.conf, and let the ix66 deal with the nat, not * . I > hope this helps. > > > Jason > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -------------- next part -------------- > A non-text attachment was scrubbed... > Name: smime.p7s > Type: application/x-pkcs7-signature > Size: 3179 bytes > Desc: not available > Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040916/066420 44/smime-0001.bin > > ------------------------------ > > Message: 6 > Date: Fri, 17 Sep 2004 21:05:08 -0700 > From: "vrushank" <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] how to get caller ID > To: <[EMAIL PROTECTED]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="iso-8859-1" > > i cannot see caller ID of the call originated from outside zap channel. > i hv configured both zapata.conf and extensions.conf. > i m right now in india > i think asterisk only supports Bellcore enable caller ID. > so is it the same bug of BT caller ID problem in UK? > or it is the bug of my asterisk configuration? > i hv enabled callerID from my TELCO. > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040917/abdd5b 46/attachment-0001.html > > ------------------------------ > > Message: 7 > Date: Thu, 16 Sep 2004 11:45:21 -0400 > From: Andrew Thompson <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] how to get caller ID > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[EMAIL PROTECTED]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > vrushank wrote: > > i cannot see caller ID of the call originated from outside zap channel. > > i hv configured both zapata.conf and extensions.conf. > > i m right now in india > > i think asterisk only supports Bellcore enable caller ID. > > so is it the same bug of BT caller ID problem in UK? > > or it is the bug of my asterisk configuration? > > i hv enabled callerID from my TELCO. > > Have you monitored the console while the line is ringing to verify that > asterisk is not seeing the callerid and not paying attention to it? > > > PS: I'm testing a new email client, please forgive me if this message is > not in Plain Text. (And someone please let me know!) > > -- > Andrew Thompson > http://aktzero.com/ > > > ------------------------------ > > Message: 8 > Date: Fri, 17 Sep 2004 00:45:55 +0900 > From: Benjamin on Asterisk Mailing Lists > <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] E3 PCI Cards > To: Noah Miller <[EMAIL PROTECTED]> > Cc: Asterisk Users Mailing List - Non-Commercial Discussion > <[EMAIL PROTECTED]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=US-ASCII > > On Thu, 16 Sep 2004 11:03:48 -0400, Noah Miller <[EMAIL PROTECTED]> wrote: > > > Does anyone have * running on PPC? > > > > http://www.voip-info.org/tiki-index.php?page=Asterisk%20MacOSX%20Support > > > > Specifically for OS X. There's a download link. The problem still is > > that no one has written ppc drivers for the Digium cards. As I > > understand, the only drivers are for GNU/Linux on i386. > > That's not entirely correct. The Zaptel drivers work on LinuxPPC. > > Further, there is some work in progress on Zaptel drivers for BSD and > some folks use X100P and TDM400 on FreeBSD already. Since OSX is BSD > based, it will eventually benefit from the work done to bring Zaptel > to BSD. We have made an Xserve available for Rich Murphey, one of the > main contributors to the Asterisk on BSD effort, specifically for him > to test things on OSX. > > What's needed is more contributors to the BSD effort, or so it would > seem. Since driver development requires skills that are less common > than those required for many other development tasks, there are fewer > people who can do it. It also takes more time to move drivers from one > platform to another. I think a sponsorship fund could do some good > because it might give somebody the ability to work fulltime on drivers > for BSD in general and OSX in particular. > > I believe that it should be possible to raise significant sponsorship > funds for drivers (especially for OSX) from end user donations alone. > In order to do that, a few people need to come together, think about > how to organise this, set up a website, open a kagi and/or paypal > account and get the word out. I am discussing this idea at present > with some Mac folks who seem to be willing to put a bit of time and > effort into this. Anybody who would like to join in on this, please > contact me directly. > > rgds > benjk > > -- > Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, > Tokyo, Japan. > > NB: Spam filters in place. Messages unrelated to the * mailing lists > may get trashed. > > > ------------------------------ > > Message: 9 > Date: Thu, 16 Sep 2004 11:54:38 -0400 > From: "Christopher Jacob" <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] Beyond T1 > To: <[EMAIL PROTECTED]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii" > > All, > > This may be a stupid question, but here it is... > > What interface gives the most density? Do I top out at T1's? For instance, 4 > t1's to the Digium Quad span t1 card. Is there an interface available for T3 > or DS3? > > Thanks, > > Chris > > > > > > > ------------------------------ > > Message: 10 > Date: Thu, 16 Sep 2004 18:02:51 +0200 > From: Maros RAJNOCH <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] call parking & forwarding > To: ASTERISK <[EMAIL PROTECTED]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii" > > Hi everbody, > > I have problem with configuring call parking and forwarding. > > firstly my setup: > I have one asterisk with gnu-gatekeeper on the same PC. > As phones I use voip-phones with H323 support. > > phones are registered on gatekeeper as terminal and > asterisk as gateway. > > > I setup features.conf (parking.conf) like: > > [general] > parkext => 700 > parkpos => 701-720 > context => parkedcalls > parkingtime => 45 > > and include parkedcalls context to extensions.conf > but without any success > > for example: somebody call me from PSTN, and I pick up call on my h323 phone in room #1 > Now I want to go to another room (room#2), so I dial #700 (in hope to transfer call to parking queue) > At this time I hear tone (one tone for any one keystroke -- I think tones are simulated by phone - not by asterisk) > but nothing to happen. Also no records in asterisk logs. > > Have anybody idea what may be wrong? > > Another situation: call forward. > > I have no idea how to do it. There's no any reference in any documentation!? > I mean: Somebody call me from PSTN and I pick up this call by my h323 phone. > Now I want forward this call to my colleague to another h323 phone. > > ANY IDEA HOW TO DO IT? > > Thanks for any help. > -------------- next part -------------- > A non-text attachment was scrubbed... > Name: not available > Type: application/pgp-signature > Size: 189 bytes > Desc: not available > Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040916/b59773 99/attachment-0001.pgp > > ------------------------------ > > Message: 11 > Date: Thu, 16 Sep 2004 13:06:10 -0300 > From: "Johannes van Hulst" <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] What can you do with Asterisk in Brazil > following the law > To: <[EMAIL PROTECTED]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii" > > Has anybody any idea what I can do with asterisk following the Brazilian > law. > > I do not have a multimedia license or a telecom license, but I ace asterisk. > > > > Are there companies who are looking for asterisk expertise in Rio de > Janeiro? > > > > > > Greeting Han > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040916/f0c5dd fc/attachment-0001.html > > ------------------------------ > > Message: 12 > Date: Thu, 16 Sep 2004 18:16:46 +0200 > From: Maros RAJNOCH <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] ID for outgoing calls from DDI (DID) line > To: ASTERISK <[EMAIL PROTECTED]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii" > > Hi again, > > in my * I have one ISDN BRI line with DID (DDI) preselection. > so in fact I have 100 extensions: +421 33 12 34 56 xx > > Q: Is in my power -- or in power of * -- to influence which of these > extensions will occur in my cellular display? > > THANKS. > -------------- next part -------------- > A non-text attachment was scrubbed... > Name: not available > Type: application/pgp-signature > Size: 189 bytes > Desc: not available > Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040916/adc3c2 a4/attachment-0001.pgp > > ------------------------------ > > Message: 13 > Date: Thu, 16 Sep 2004 12:17:20 -0400 > From: Andrew Thompson <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] Beyond T1 > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[EMAIL PROTECTED]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Christopher Jacob wrote: > > All, > > > > This may be a stupid question, but here it is... > > > > What interface gives the most density? Do I top out at T1's? For instance, 4 > > t1's to the Digium Quad span t1 card. Is there an interface available for T3 > > or DS3? > > Depending on where you using the circuits, you might try an E1. It uses > the same total bandwidth as a T1(I think), but splits the channels at > 56K instead of 64K, yielding more channels. (And now I can't remember > the number.) > > I haven't heard of direct DS3 connectivity... > > Just stretching my imagination a little bit, you might be able to plug a > DS3 into a H323 box, and then feed the IP-side of the calls to > asterisk.... > > -- > Andrew Thompson > http://aktzero.com/ > > > ------------------------------ > > Message: 14 > Date: Thu, 16 Sep 2004 09:19:57 -0700 > From: "Steven P. Donegan" <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] Beyond T1 > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[EMAIL PROTECTED]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Andrew Thompson wrote: > > > Christopher Jacob wrote: > > > >> All, > >> > >> This may be a stupid question, but here it is... > >> > >> What interface gives the most density? Do I top out at T1's? For > >> instance, 4 > >> t1's to the Digium Quad span t1 card. Is there an interface available > >> for T3 > >> or DS3? > > > > > > Depending on where you using the circuits, you might try an E1. It > > uses the same total bandwidth as a T1(I think), but splits the > > channels at 56K instead of 64K, yielding more channels. (And now I > > can't remember the number.) > > > > I haven't heard of direct DS3 connectivity... > > > > Just stretching my imagination a little bit, you might be able to plug > > a DS3 into a H323 box, and then feed the IP-side of the calls to > > asterisk.... > > > Actually T1 is 24x64k and E1 is 30x64k - 1.536 megabits/sec -vs- 2.0 if > I recall correctly... > > > > ------------------------------ > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > > End of Asterisk-Users Digest, Vol 2, Issue 152 > ********************************************** _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
