[Asterisk-Users] G.729 on YDL and MacOSX
Re: G.729 codec on Yellow Dog Linux for various PPC Kristian Kielhofner [EMAIL PROTECTED] wrote: This is probably a good time to ask if there is any planned support for a g729 binary for YDL and G3/G4, etc. I would love to start playing with apple hardware, YDL, and asterisk. But I need that binary! Indeed it is a good time to ask (but always start a new thread ;-) I have mentioned this before, and I would like to ask EVERYBODY who is interested to VOICE your interest directly with the respective vendors. This is the first step and it is VERY IMPORTANT. I am confindent that an Altivec optimised G.729 codec for the PPC970 CPUs (aka G5) on YDL4 would so clearly trash any Intel or AMD based system that most serious deployments that require G.729 will end up using Xserve instead of Intel toyz. Combine this with the fact that the x86 architecture has hit the wall while IBM is only getting started. Even Microsoft have recognised the leadership of IBM by going PPC with their new game console. Before this background it is quite apparent that there is an interesting market potential for G.729 binaries for LinuxPPC. However, without requests from customers for a G.729 codec for LinuxPPC it will take so much longer for an x86 centric shop like Digium to recognise this potential and consider spending time and effort on it. Therefore, please, send an email to Digium and tell them that you want this binary for PPC and continue to nag them about it again and again and again and again. If as a result, Digium realise that there is demand, then they will quite possibly provide that binary. At the same time, let's also remind TerraSoft (http://www.terrasoftsolutions.com) that Asterisk on their YDL platform is alive and that their sponsorship to bring Asterisk to LinuxPPC was not in vain, that there is finally an opportunity to get a return on their investment. Let's assume that Digium is simply too busy with other things and that even if they wanted to, they couldn't do the G.729 codec for PPC. So, in lieu of Digium providing the codec for PPC, TerraSoft may recognise the opportunity and step in. But again, in order for this to happen, it will take requests from customers. Therefore, please, send an email to Kai Staats at TerraSoft and tell them that you'd be very interested to buy G.729 codec binaries for Asterisk on YDL if they were to offer them, then follow up on that with reminders to show that you are serious about it. TerraSoft have been working together with Digium to bring Asterisk to YDL, so there shouldn't be a problem for the two companies to get together again and bring the G.729 codec to YDL as well. All it takes for that to happen is visible customer demand. Perhaps we should set up some kind of petition page on the Wiki. Re: G.729 codec on MacOSX for Apple Macintosh Darren Sessions [EMAIL PROTECTED] wrote: Or for that matter, is there a planned G729 binary for Mac OSX ? It will probably take a LinuxPPC port first, but here again, why don't you send email to Apple and tell them that you would rather purchase oodles of Xserve instead of x86 based servers if only there was a G.729 codec for OSX. It will take a lot more noise to get Apple to recognise that there is a market potential than it will take to get Digium or TerraSoft to do so, but that's no reason not to make a request. So, please, send email to Apple and tell them that you have tested Asterisk on MacOSX -- they have listed our installer on their website http://www.apple.com/downloads/macosx/unix_open_source/asteriskinstallpackageformacosx.html, that you found it runs circles around any other product, such as Cisco Call Manager -- Apple just loves to hear that sort of thing -- and that the only thing that's missing is the G.729 codec which the open source community is unable to provide on its own due to the patent royalties that need to be paid on a reseller-to-patent-holder basis because there is no end-user-to-patent-holder scheme, that you would love to buy many many Xserves if Apple was to sell you the missing codec. 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. ___ 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
[Asterisk-Users] Asterisk, ATA-186 Sipgate.de / sipgate.co.uk
I try to get the following to work: Sipgate.de and sipgate.co.uk are configured as gateway, while the ATA-186 has two phone sets attached. I tried: ATA settings as described at: http://www.loligo.com/asterisk/Cisco/ATA-186-guide.v20030628.txt (just with a fixed IP) sip.conf: == [general] context=default port=5060 bindaddr=0.0.0.0 srvlookup=yes register = 5552220:[EMAIL PROTECTED]/5552220 register = 4782156:[EMAIL PROTECTED]/4782156 externip = 61.220.121.xx localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks [601] type=friend username=601 secret=my_password1 canreinvite=no host=dynamic dtmfmode=rfc2833 mailbox=601 nat=yes [602] type=friend username=602 secret=my_password2 canreinvite=no host=dynamic dtmfmode=rfc2833 mailbox=601 nat=yes [sipgate.de] type=friend username=5552220 secret=my_password3 host=sipgate.de fromuser=5552220 fromdomain=sipgate.net nat=yes context=incomingsipgate canreinvite=no [sipgate.co.uk] type=friend username=4782156 secret=my_password4 host=sipgate.co.uk fromuser=4782156 fromdomain=sipgate.net nat=yes context=incomingsipgate canreinvite=no extensions.conf: === [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp ; Console interface for demo [incomingsipgate] exten = h,1,Hangup exten = 800,1,Dial(SIP/internestelefon,20,tr) [sipgate.de] exten = _0049.,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr) exten = _0049.,2,Playback(invalid) exten = _0049.,3,Hangup [sipgate.co.uk] exten = _0044.,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr) exten = _0044.,2,Playback(invalid) exten = _0044.,3,Hangup I did not understand the paragraph of [incomingsipgate]. I also do not understand EXTEN:1 (should the second phone be EXTEN:2 ???) starting the server with asterisk -vvvcg brings a lots of lines ;-) sip show users: sipgate.co.uk my_password2 incomingsipgateNo Alway sipgate.demy_password1 incomingsipgateNo Alway 602 my_password4 incomingsipgateNo Alway 601 my_password3 incomingsipgateNo Alway sip show registry: sipgate.co.uk:5060 4782156 105Registered sipgate.de:5060 5552220 105Registered Tests: 601 calls 602busy 00491 busy(1 at sipgate.de should play a tape) No info on the screen (asterisk: *CLI ) What have I forgotten / made wrong? bye Ronald begin:vcard fn:Ronald Wiplinger n:Wiplinger;Ronald email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE version:2.1 end:vcard ___ 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
Re: [Asterisk-Users] KSS/BLF on Asterisk
[EMAIL PROTECTED] wrote: Folks, I am trying to determine the best way to allow a station to monitor the status of another station. For example: a reception set needing to see the status of 20 or 30 phones OR an executive assistant wanting to have appearances of several other extensions, in order to monitor their status and assist with call handling. I know Snom has a phone that you can attach an add-on module to, but I don't know how you'd program Asterisk to deliver status information to those buttons. Get a hint! :-) Check out the hint priority in extensions.conf. There are also some details in the wiki. Nick ___ 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
Re: [Asterisk-Users] Asterisk, ATA-186 Sipgate.de / sipgate.co.uk
Ronald Wiplinger schrieb: [sipgate.de] type=friend username=5552220 secret=my_password3 host=sipgate.de fromuser=5552220 fromdomain=sipgate.net should be fromdomain=sipgate.de nat=yes context=incomingsipgate canreinvite=no [sipgate.co.uk] type=friend username=4782156 secret=my_password4 host=sipgate.co.uk fromuser=4782156 fromdomain=sipgate.net should be fromdomain=sipgate.co.uk nat=yes context=incomingsipgate canreinvite=no extensions.conf: === [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp; Console interface for demo [incomingsipgate] exten = h,1,Hangup exten = 800,1,Dial(SIP/internestelefon,20,tr) should be [incomingsipgate] exten = 5552220,1,Dial(SIP/internestelefon,20,r) exten = 4782156,1,Dial(SIP/internestelefon,20,r) [sipgate.de] exten = _0049.,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr) exten = _0049.,2,Playback(invalid) exten = _0049.,3,Hangup should be (you forgot to number prio 1 !) exten = _0049.,1,Dial(SIP/0${EXTEN:[EMAIL PROTECTED],30,r) ; do not dial international prefix 0049 with Sipgate, if you call from same national net ! exten = _0049.,2,Playback(invalid) exten = _0049.,3,Hangup [sipgate.co.uk] exten = _0044.,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr) exten = _0044.,2,Playback(invalid) exten = _0044.,3,Hangup exten = _0044.,1,Dial(SIP/0${EXTEN:[EMAIL PROTECTED],30,tr) ; do not dial international prefix 0044 with Sipgate, if you call from same national net ! exten = _0044.,2,Playback(invalid) exten = _0044.,3,Hangup I did not understand the paragraph of [incomingsipgate]. This is the context to which all incoming calls from Sipgate will be sent to be handled. I also do not understand EXTEN:1 (should the second phone be EXTEN:2 ???) please regard correct expression ${EXTEN:1} ! This means take the variable ($) called {EXTEN} (this is the dialed number) and cut the FIRST digit (:1) So - if you dial 0049341234567 and then DIAL(SIP/${EXTEN:1}...) this will result in dialing 0493411234567 which is not a valid number. Regards -- Please visit http://www.ip-phone-forum.de -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk, ATA-186 Sipgate.de / sipgate.co.uk
BetaTeilchen wrote: Ronald Wiplinger schrieb: Thanks for helping me, but it still does not work. [sipgate.de] type=friend username=5552220 secret=my_password3 host=sipgate.de fromuser=5552220 fromdomain=sipgate.net should be fromdomain=sipgate.de nat=yes context=incomingsipgate canreinvite=no [sipgate.co.uk] type=friend username=4782156 secret=my_password4 host=sipgate.co.uk fromuser=4782156 fromdomain=sipgate.net should be fromdomain=sipgate.co.uk nat=yes context=incomingsipgate canreinvite=no extensions.conf: === [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp; Console interface for demo [incomingsipgate] exten = h,1,Hangup exten = 800,1,Dial(SIP/internestelefon,20,tr) should be [incomingsipgate] exten = 5552220,1,Dial(SIP/internestelefon,20,r) exten = 4782156,1,Dial(SIP/internestelefon,20,r) What is the difference between tr and r ? What does the 20 mean? [sipgate.de] exten = _0049.,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr) exten = _0049.,2,Playback(invalid) exten = _0049.,3,Hangup should be (you forgot to number prio 1 !) exten = _0049.,1,Dial(SIP/0${EXTEN:[EMAIL PROTECTED],30,r) ; do not dial international prefix 0049 with Sipgate, if you call from same national net ! exten = _0049.,2,Playback(invalid) exten = _0049.,3,Hangup [sipgate.co.uk] exten = _0044.,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr) exten = _0044.,2,Playback(invalid) exten = _0044.,3,Hangup exten = _0044.,1,Dial(SIP/0${EXTEN:[EMAIL PROTECTED],30,tr) ; do not dial international prefix 0044 with Sipgate, if you call from same national net ! exten = _0044.,2,Playback(invalid) exten = _0044.,3,Hangup I did not understand the paragraph of [incomingsipgate]. This is the context to which all incoming calls from Sipgate will be sent to be handled. I also do not understand EXTEN:1 (should the second phone be EXTEN:2 ???) please regard correct expression ${EXTEN:1} ! This means take the variable ($) called {EXTEN} (this is the dialed number) and cut the FIRST digit (:1) So - if you dial 0049341234567 and then DIAL(SIP/${EXTEN:1}...) this will result in dialing 0493411234567 which is not a valid number. BTW, when I stop Asterisk with stop now I get a Yuck! Error in buffer handling ...: What does this mean? bye Ronald begin:vcard fn:Ronald Wiplinger n:Wiplinger;Ronald email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE version:2.1 end:vcard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk, ATA-186 Sipgate.de / sipgate.co.uk
Maybe you should start reading here: http://www.voip-info.org/wiki-Asterisk+introduction to get basic knowledges of Asterisk Ronald Wiplinger schrieb: BetaTeilchen wrote: Ronald Wiplinger schrieb: Thanks for helping me, but it still does not work. [sipgate.de] type=friend username=5552220 secret=my_password3 host=sipgate.de fromuser=5552220 fromdomain=sipgate.net should be fromdomain=sipgate.de nat=yes context=incomingsipgate canreinvite=no [sipgate.co.uk] type=friend username=4782156 secret=my_password4 host=sipgate.co.uk fromuser=4782156 fromdomain=sipgate.net should be fromdomain=sipgate.co.uk nat=yes context=incomingsipgate canreinvite=no extensions.conf: === [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp; Console interface for demo [incomingsipgate] exten = h,1,Hangup exten = 800,1,Dial(SIP/internestelefon,20,tr) should be [incomingsipgate] exten = 5552220,1,Dial(SIP/internestelefon,20,r) exten = 4782156,1,Dial(SIP/internestelefon,20,r) What is the difference between tr and r ? What does the 20 mean? [sipgate.de] exten = _0049.,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr) exten = _0049.,2,Playback(invalid) exten = _0049.,3,Hangup should be (you forgot to number prio 1 !) exten = _0049.,1,Dial(SIP/0${EXTEN:[EMAIL PROTECTED],30,r) ; do not dial international prefix 0049 with Sipgate, if you call from same national net ! exten = _0049.,2,Playback(invalid) exten = _0049.,3,Hangup [sipgate.co.uk] exten = _0044.,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr) exten = _0044.,2,Playback(invalid) exten = _0044.,3,Hangup exten = _0044.,1,Dial(SIP/0${EXTEN:[EMAIL PROTECTED],30,tr) ; do not dial international prefix 0044 with Sipgate, if you call from same national net ! exten = _0044.,2,Playback(invalid) exten = _0044.,3,Hangup I did not understand the paragraph of [incomingsipgate]. This is the context to which all incoming calls from Sipgate will be sent to be handled. I also do not understand EXTEN:1 (should the second phone be EXTEN:2 ???) please regard correct expression ${EXTEN:1} ! This means take the variable ($) called {EXTEN} (this is the dialed number) and cut the FIRST digit (:1) So - if you dial 0049341234567 and then DIAL(SIP/${EXTEN:1}...) this will result in dialing 0493411234567 which is not a valid number. BTW, when I stop Asterisk with stop now I get a Yuck! Error in buffer handling ...: What does this mean? bye Ronald ___ 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 ___ 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
[Asterisk-Users] bristtuff segfault
Hi list, I'd like to have comments from the bristuff / QuadBRI users, others are welcome to as I'm really lost and need to move on. I have the following setup: a first asterisk is connected to the legacy Alcatel PaBX to connect to a remote site with a second asterisk server. PSTN | Legacy phones == Alcatel Omnipcx == Asterisk1 | | IAX | Asterisk2 == 25 SIP phones Both servers are dell 400sc (Pentium(R) 4 CPU 2.80GHz, 2 IDE disks in software RAID1). The system has been working flawlessly for a week with 4 analog lines between the omnipcx and asterisk1 (using TDM04B). Now I've moved to a 4 BRI link between omnipcx and asterisk1, using QuadBRI from Junghanns.net. It does work... for about 3 minutes, and then asterisk segfault after hanging up a successful call to the omnipcx. Backtrace shows the error is in libpri function q931_destroy (q931.c: 1908), which is coherent with crash after hangup. Asterisk / Zaptel / Libpri / Qozap on first server have been built with the scripts from bri-stuff-0.1.0-RC4a.tar.gz. On the card, I only changed all groups of 5 jumpers to NT mode. Linux Distro is Mandrake-10.0. I tried with kernels (from kernel.org) 2.6.8.1 and 2.4.27 with and without SMP, and got exactly same results. (more details below) Any hints would be much appreciated. Thanks for a prompt reply. Jean-Denis Asterisk1 logs shows warnings on all hangups, eg. Oct 22 16:39:47 WARNING[1101196208]: PRI: Can't destroy call 133! Oct 22 16:39:47 WARNING[1101196208]: Hangup on bad channel 0/1 on span 1 Oct 22 16:39:54 WARNING[1101196208]: Hangup on bad channel 0/1 on span 1 Oct 22 16:39:58 WARNING[1101196208]: Hangup on bad channel 0/1 on span 1 Oct 22 16:40:04 WARNING[1101196208]: PRI: Can't destroy call 135! Oct 22 16:40:04 WARNING[1101196208]: Hangup on bad channel 0/1 on span 1 Oct 22 16:40:41 WARNING[1101196208]: PRI: Can't destroy call 137! Oct 22 16:40:41 WARNING[1101196208]: Hangup on bad channel 0/2 on span 1 Oct 22 16:41:09 WARNING[1101196208]: Hangup on bad channel 0/1 on span 1 Oct 22 16:41:13 WARNING[1101196208]: Hangup on bad channel 0/1 on span 1 Oct 22 16:41:19 NOTICE[1121725360]: I should never be called! Oct 22 16:41:53 WARNING[1101196208]: PRI: Can't destroy call 140! Oct 22 16:41:53 WARNING[1101196208]: Hangup on bad channel 0/2 on span 1 Loading qozap shows no error except for devfs, but I doubt this is the source of the problem: Oct 22 16:29:43 asterisk1 kernel: Zapata Telephony Interface Registered on major 196 Oct 22 16:29:56 asterisk1 kernel: PCI: Enabling device 02:02.0 ( - 0003) Oct 22 16:29:56 asterisk1 kernel: PCI: Found IRQ 10 for device 02:02.0 Oct 22 16:29:56 asterisk1 kernel: PCI: Sharing IRQ 10 with 00:1f.3 Oct 22 16:29:56 asterisk1 kernel: PCI: Sharing IRQ 10 with 00:1f.5 Oct 22 16:29:56 asterisk1 kernel: qozap: Junghanns.NET quadBRI card configured at mem 0xe08e IRQ 10 HZ 100 CardID 0 Oct 22 16:29:56 asterisk1 kernel: qozap: S/T ports: 4 [ NT NT NT NT ] Oct 22 16:29:56 asterisk1 kernel: card 1 span 1 state G0 (A_ST_RD_STA = 0x0) Oct 22 16:29:56 asterisk1 kernel: card 1 span 2 state G0 (A_ST_RD_STA = 0x0) Oct 22 16:29:56 asterisk1 kernel: card 1 span 3 state G0 (A_ST_RD_STA = 0x0) Oct 22 16:29:56 asterisk1 kernel: card 1 span 4 state G0 (A_ST_RD_STA = 0x0) Oct 22 16:29:56 asterisk1 kernel: qozap: 1 multiBRI card(s) in this box, 4 BRI ports total. Oct 22 16:29:56 asterisk1 devfsd[168]: error copying: /lib/dev-state/zap/1 to /dev/zap/1 Oct 22 16:29:56 asterisk1 devfsd[168]: error copying: /lib/dev-state/zap/2 to /dev/zap/2 Oct 22 16:29:56 asterisk1 devfsd[168]: error copying: /lib/dev-state/zap/3 to /dev/zap/3 Oct 22 16:29:56 asterisk1 devfsd[168]: error copying: /lib/dev-state/zap/4 to /dev/zap/4 Oct 22 16:29:56 asterisk1 devfsd[168]: error copying: /lib/dev-state/zap/5 to /dev/zap/5 Oct 22 16:29:56 asterisk1 devfsd[168]: error copying: /lib/dev-state/zap/6 to /dev/zap/6 Oct 22 16:29:56 asterisk1 devfsd[168]: error copying: /lib/dev-state/zap/7 to /dev/zap/7 Oct 22 16:29:56 asterisk1 devfsd[168]: error copying: /lib/dev-state/zap/8 to /dev/zap/8 Oct 22 16:29:56 asterisk1 devfsd[168]: error copying: /lib/dev-state/zap/9 to /dev/zap/9 Oct 22 16:29:56 asterisk1 devfsd[168]: error copying: /lib/dev-state/zap/10 to /dev/zap/10 Oct 22 16:29:56 asterisk1 devfsd[168]: error copying: /lib/dev-state/zap/11 to /dev/zap/11 Oct 22 16:29:56 asterisk1 devfsd[168]: error copying: /lib/dev-state/zap/12 to /dev/zap/12 Oct 22 16:30:36 asterisk1 kernel: Registered tone zone 2 (France) Oct 22 16:30:36 asterisk1 kernel: card 1 span 1 state G2 (A_ST_RD_STA = 0x82) Oct 22 16:30:36 asterisk1 kernel: card 1 span 2 state G2 (A_ST_RD_STA = 0xc2) Oct 22 16:30:36 asterisk1 kernel: card 1 span 3 state G2 (A_ST_RD_STA = 0x82) Oct 22 16:30:36 asterisk1 kernel: card 1 span 4 state G2 (A_ST_RD_STA = 0xc2) Oct 22
Re: [Asterisk-Users] iLBC/PCM16 Huge Cost
Todd Lieberman wrote: Wo trevor, Format and start over? Don't go crazy, just remove the files created by make install. Fighting for weeks to get a more-or-less stable telephone system can drive a man to do extraordinary things like rebuilding a server from scratch! We are making process, however. With the 0.59r mpg123, I see no processes consuming all sorts of CPU power once * has been running for a few hours. show translation still reveals the iLBC column in the 8700 to 9600 range though. LPC10's row is also in the 900s. show translation recalc 10 still causes the * console to stop responding as well. When I first bootup *, it consumes nearly 100% of CPU, all in user... system is less than 1%. The time that it monopolizes the processor varies as well. ___ 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
Re: [Asterisk-Users] Hardware
On Sat, Oct 23, 2004 at 06:40:12PM -0600, Michael Loftis wrote: --On Saturday, October 23, 2004 19:56 -0400 Stan Brinkerhoff [EMAIL PROTECTED] wrote: Look for support by whatever operating system you plan on running. I second thatpretty much any P4 based hardware should be perfectly fine for asterisk. I'd tend to lean towards SCSI drives though, but other than that go to town! Why scsi? I thought that Asterisk doesn't have much disk IO. At least that this is not a bottleneck. -- Tzafrir Cohen +---+ http://www.technion.ac.il/~tzafrir/ |vim is a mutt's best friend| mailto:[EMAIL PROTECTED] +---+ ___ 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
Re: [Asterisk-Users] Hardware
I would use a Western Digital Raptor SATA Harddisk, also gives you a performance boost of your system + it aint that expensive as scsi. And my dream setup for asterisk would be: dual xeon, intel xeon motherboard, 2gig ram for each cpu and a few raptor or scsi disks + some wildcard digium telephony cards to call with 10users at a time to a normal phone number. On Sun, 2004-10-24 at 11:49, Tzafrir Cohen wrote: On Sat, Oct 23, 2004 at 06:40:12PM -0600, Michael Loftis wrote: --On Saturday, October 23, 2004 19:56 -0400 Stan Brinkerhoff [EMAIL PROTECTED] wrote: Look for support by whatever operating system you plan on running. I second thatpretty much any P4 based hardware should be perfectly fine for asterisk. I'd tend to lean towards SCSI drives though, but other than that go to town! Why scsi? I thought that Asterisk doesn't have much disk IO. At least that this is not a bottleneck. -- Christophe De Coninck | Zarek K http://www.zarekk.be mailto: [EMAIL PROTECTED] mailto: [EMAIL PROTECTED] attachment: banner.gif___ 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
Re: [Asterisk-Users] Fedora 2, Kudzu and X100P
On Sat, Oct 23, 2004 at 09:29:00PM -0500, Carlos Chavez wrote: I am installing a new * server using Fedora Core 2 but I ran into a problem after I installed the X100P. When FC2 boots it runs KUDZU to detect new hardware and it detected the card and insists on loading the module crc_ccitt before the zaptel module. Because of this I cannot load the wcfxo module without the computer crashing. I have already erase the entry in /etc/sysconfig/hwconf and turned kudzu off during boot. Anyone know of a way to fix this (short or reinstalling FC2)? One obvious solution is not to automatically load kudzu. chkconfig --remove kudzu Another obvious solution of the same sort is modprobing the zaptel module earlier in the boot process. I can't seem to figure out , though, where kudzu takes its modue names from. I haven't bothred reading th source yet, though (not from /usr/share/hwdata, it seems) -- Tzafrir Cohen +---+ http://www.technion.ac.il/~tzafrir/ |vim is a mutt's best friend| mailto:[EMAIL PROTECTED] +---+ ___ 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
[Asterisk-Users] Digium TheVoice recordings' sound terrible
A customer has ordered some voice prompts from Digium's TheVoice online store. They say the recordings' sound was good when they listened to it on their Windoze boxes. However, then Asterisk is playing back the recordings, the volume is far too high and they sound really bad. This is particularly noticeable since the IVR menu mixes those ordered recordings with recordings that are already part of the Asterisk distribution. The volume of the included recordings are much lower and they sound much better than the ordered recordings. I wonder why Digium would deliver recordings that differ so much from the included set of recordings. However, the format the customer ordered was WAV, whereas all the included recordings are of course GSM. Has anybody had similar experiences? I tried to convert the WAV files to GSM using sox but since I don't know what parameters are best in this case, the results weren't satisfactory. Any suggestions? thanks 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. ___ 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
Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible
Things that are productive. 1. I am sure there are free programs that will allow you to adjust the files to sound more like the originial recordings as well as converting them to gsm. Do some searching and learning and fix it yourself. Mail Digium directly so that they are aware of the problem and can correct it for future recordings. Possibly something like this http://www.softpicks.net/software/Complete-Audio-Converter-Lite-1470.htm 2 . If you dont want to go through all of that, kindly ask Digium to have the files fixed for you. I seriously doubt they have their own sound stage and most likely outsource this type of business. Chances are the people they outsource the business to are experts and have sophisticated equipment to do this with very fast turn around time. 3. Email thousands of people that will probably ignore you or not know the answer (with the exception of myself). I would choose number one. Thanks, Steve - Original Message - From: Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 6:37 AM Subject: [Asterisk-Users] Digium TheVoice recordings' sound terrible A customer has ordered some voice prompts from Digium's TheVoice online store. They say the recordings' sound was good when they listened to it on their Windoze boxes. However, then Asterisk is playing back the recordings, the volume is far too high and they sound really bad. This is particularly noticeable since the IVR menu mixes those ordered recordings with recordings that are already part of the Asterisk distribution. The volume of the included recordings are much lower and they sound much better than the ordered recordings. I wonder why Digium would deliver recordings that differ so much from the included set of recordings. However, the format the customer ordered was WAV, whereas all the included recordings are of course GSM. Has anybody had similar experiences? I tried to convert the WAV files to GSM using sox but since I don't know what parameters are best in this case, the results weren't satisfactory. Any suggestions? thanks 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. ___ 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 ___ 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
Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible
Option #4, send me the files and $100 via paypal and I will fix them for you. - Original Message - From: Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 6:37 AM Subject: [Asterisk-Users] Digium TheVoice recordings' sound terrible A customer has ordered some voice prompts from Digium's TheVoice online store. They say the recordings' sound was good when they listened to it on their Windoze boxes. However, then Asterisk is playing back the recordings, the volume is far too high and they sound really bad. This is particularly noticeable since the IVR menu mixes those ordered recordings with recordings that are already part of the Asterisk distribution. The volume of the included recordings are much lower and they sound much better than the ordered recordings. I wonder why Digium would deliver recordings that differ so much from the included set of recordings. However, the format the customer ordered was WAV, whereas all the included recordings are of course GSM. Has anybody had similar experiences? I tried to convert the WAV files to GSM using sox but since I don't know what parameters are best in this case, the results weren't satisfactory. Any suggestions? thanks 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. ___ 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 ___ 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
[Asterisk-Users] Call Waiting
Hi, I have just set up an Asterisk box.it sure is a big job to get everything perfect, especially when you have picky users. Anyway, the box has 2 X100P's and a couple of sipura spa-2000's connected to the LAN. 1 of the lines connected to the X100P's goes straight to extension 1000 after a short greeting. Anyway, say ext 1000 is talking to ext 1001, and the line rings. Ext 1000 hears a small beep every few seconds. This is obviously call waiting. My question is how do I answer that incoming call whilst on a call? I have looked around, tried *0 and even 0*, the flash key, but to no avail :( ___ 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
Re: [Asterisk-Users] Digium Wildcard T1 Compatibility
A flex grow is like a channel bank. A normal PRI comes into a router. The router breaks out some channels for data and the other voice channels become analog POTS lines. You will need POTS cards. I am positive that you could have your T100P and asterisk provide this function so that you wouldnt need their equipment or POTS. Just depends on the tech you get whether they will help or not. Just read the first paragraph of the product description on Digium's site. http://www.digium.com/index.php?menu=wildcard_t100p - Original Message - From: Cirelle Enterprises [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, October 22, 2004 4:06 PM Subject: Re: [Asterisk-Users] Digium Wildcard T1 Compatibility - Original Message - From: Daniel Daley [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, October 21, 2004 5:49 PM Subject: [Asterisk-Users] Digium Wildcard T1 Compatibility | Hi, | | I have a quick question about the T100P. I've used the card before in a | PRI setup and it worked great. I'm now trying to figure out a setup for | another company that gets services from Verizon. They offer what they | call a flexgrow T1 where they say the voice lines are delivered as just | standard POTS channels. Will the wildcard handle this kind of T1 or is | that something you would need to break out into separate lines and go | into POTS cards? | | Thanks, | | --Dan-- | for what it's worth, we were told to use RJ48C (Std Ethernet Cable) ___ 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 ___ 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
[Asterisk-Users] chan_sip CallerPres support?
hi would it be hard to implement CallerPres support in chan_sip? roy ___ 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
Re: [Asterisk-Users] Re: Grandstream Flashing (different issue)
May be risky if your email is screwy but it solves your problem Add: delete=yes in your voicemail.conf. - Original Message - From: Stephen R. Besch [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, October 22, 2004 11:55 AM Subject: [Asterisk-Users] Re: Grandstream Flashing (different issue) Todd Routhier - Lightwave Technologies, LLC. wrote: OK, this is a different flashing issue than the one that's being talked about. I have a few of these phones (GrandStream 101) and when a voicemail is received the light on the LED starts blinking and the dial tine stutters, this is cool. BUT. How the do I get it to stop, I have had mine covered with paper for the last 2 days because the blinking LED panel is driving me nuts. I have received the messages in my email and looked at them on the web. I am thinking that I have to check them by calling into the Asterisk system and mark them as read or something in order for this to quite. Two problems, I don't know how to check them by phone just yet and I will likely never check them by phone. You absolutely must get the message file deleted from the mail store. It doesn't matter how you delete the message either. You can use the * phone interface or simply delete the files associated with the message. As soon as the files are gone from the INBOX, * sends a SIP command to the phone to turn off the MWI. As long as there are any active messages in the INBOX, the light stays on. If you would like to be a guinea pig for my VB program that allows you to manage your mail folders and messages from a Windows GUI, I'll send you a copy. The only caveat is that I haven't yet found a way to get perfect synchronization with file access to the mail store. The user needs to be careful not to modify the INBOX while * is taking a new message. For me this is usually not a problem, since I would never be not answering the phone when I am listening to/moving/deleting messages - as a result, * would never be writing a new message when I was using the VB program. The program requires the VB6 runtime (available from MS for free) and that you have SAMBA running on the * server. Sincerely, Stephen R. Besch ___ 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 ___ 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
[Asterisk-Users] Problem compiling ZPAHFC with Suse 9.1, Kernel 2.6.5
Hi, I've some problems compiling/installing the ZAPHFC-Driver. I've download the actuell version bristuff-0.1.0-RC4a from junghanns.net. I use SUSE 9.1 with kernel 2.6.5.-111. I've made the symbolic link to Linux-2.6 and test the link successfully. I've done make oldconig, make menuconfig and make in the linux-directory. When I start ./compile.sh in the bri-stuff-directory (./download.sh alraedy done before), zaptel and libpri will be compiled without problems. But compiling of qozap and zaphfc will end wit error: zt_register, -_unregister, -_transmit, -_receive and -_chunk are not defined It's possible to install the ZAPHFS-driver with make loadNT but it reports 0 channels configured. I've alraedy googled this problem but find only users with the same problem, no resolution. Have anyone one? Joachim ___ 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
Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible
Hi Steve, On Sun, 24 Oct 2004 07:47:17 -0400, Steve Totaro [EMAIL PROTECTED] wrote: 1. I am sure there are free programs that will allow you to adjust the files to sound more like the originial recordings as well as converting them to gsm. that's all very cool, but if you read my post carefully -- you did read more than just the subject line, did you not?! -- then you will find that I had mentioned I was unable to get acceptable results using sox because I don't know what parameters to use. Are you suggesting that sox is not the right tool for the job, no matter what parameters? I would be very surprised if that was the case. Since I didn't think this is the only time that Digium delivered WAV recordings that are out of sync with the Asterisk GSM sound library, in other words, that this was a known problem, I honestly expected that somebody else had already discovered what parameters to use with sox to do this conversion properly. Do some searching and learning and fix it yourself. I am sure you will appreciate that not everybody is a sound engineer and not everybody has the time to spend more than a full day of experimenting to fix five recordings. You will also appreciate that in such cases people go to mailing lists like this one where they can reasonably expect that somebody else has already discovered what the proper parameters are. Last but not least, my dear friend, if you had the courtesy to browse the list archive and check out how many times I came here for help versus how many times I have helped others, then you will probably find that your remarks are misplaced. Digium directly so that they are aware of the problem and can correct it for future recordings. Done that already. No response. Hence my posting here. 3. Email thousands of people that will probably ignore you or not know the answer (with the exception of myself). Well, if you know the answer, then why do you have to write a novel instead of posting what -- from your experience -- the correct sox parameters should look like, for example? I would choose number one. And quite possibly revert to asking the list after you have failed to produce any acceptable results and Digium didn't respond to you. But then again, I feel sorry that you had such a bad day today. While I am glad that your wrath didn't hit some poor newbie who would probably be scared away from ever posting to this list again, I sincerely hope that you feel better now. kind regards benjamin -- 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. ___ 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
[Asterisk-Users] How to create Groups/members and do Conferencing?
Hi, I have just been able to compile asterisk, so that says that I'm fairnly new to Asterisk. I'm still figuring out how to use it with my User Agents. My requirement is:1. To make a "Group" containing some agents (SIP User Agents) as members.2. To start a conference between the members of the Group. (The asterisk server shoulddo the conferencing between these SIP User Agents. So the asterisk server shouldbe able to understand the request from one member to be redirected to all the group members).I'll be really grateful if I could have some suggestions as to how can I create the Group and start the conference?Note: The SIP User agents and the asterisk server are running on the same machine. Join Excite! - http://www.excite.comThe most personalized portal on the Web! ___ 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
Re: [Asterisk-Users] Call Waiting
You are supposed to be able to either press flash or quickly push the actual hook switch. - Original Message - From: Nikhil Jogia [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 7:54 AM Subject: [Asterisk-Users] Call Waiting Hi, I have just set up an Asterisk box.it sure is a big job to get everything perfect, especially when you have picky users. Anyway, the box has 2 X100P's and a couple of sipura spa-2000's connected to the LAN. 1 of the lines connected to the X100P's goes straight to extension 1000 after a short greeting. Anyway, say ext 1000 is talking to ext 1001, and the line rings. Ext 1000 hears a small beep every few seconds. This is obviously call waiting. My question is how do I answer that incoming call whilst on a call? I have looked around, tried *0 and even 0*, the flash key, but to no avail :( ___ 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 ___ 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
RE: [Asterisk-Users] Digium TheVoice recordings' sound terrible
Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] wrote: A customer has ordered some voice prompts from Digium's TheVoice online store. They say the recordings' sound was good when they listened to it on their Windoze boxes. However, then Asterisk is playing back the recordings, the volume is far too high and they sound really bad. This is particularly noticeable since the IVR menu mixes those ordered recordings with recordings that are already part of the Asterisk distribution. The volume of the included recordings are much lower and they sound much better than the ordered recordings. I wonder why Digium would deliver recordings that differ so much from the included set of recordings. Did you ask Digium? There's probably not a lot of point in asking for customer support from any specific vendor in this mail list. You should contact the vendor's support department directly. However, the format the customer ordered was WAV, whereas all the included recordings are of course GSM. Has anybody had similar experiences? I tried to convert the WAV files to GSM using sox but since I don't know what parameters are best in this case, the results weren't satisfactory. Any suggestions? You could try using sox. That will convert WAV files to GSM and adjust the volume for you during the conversion process. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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
Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible
However, the format the customer ordered was WAV, whereas all the included recordings are of course GSM. Has anybody had similar experiences? I tried to convert the WAV files to GSM using sox but since I don't know what parameters are best in this case, the results weren't satisfactory. Any suggestions? Benjamin, Don't know if this helps you or not, but this is taken right from Jtodd's wiki page. **my disclaimer** Might want to backup your sounds before doing this though. #!/bin/sh tmpfile=/tmp/rescale$$.wav for i in *.wav; do scale=$(sox $i /tmp/foo.wav stat -v 21) if [ $scale != 1.000 ]; then echo -n Rescale $i... cp $i $tmpfile sox $tmpfile -v $scale $i echo fi done The wiki page is at http://www.voip-info.org/tiki-index.php?page=Asterisk%20sound%20files Hope this helps, -Chuji ___ 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
[Asterisk-Users] Re: Direct SIP connection to Vonage service
Hi Benjamin, I looked at NuFone.net and some others, but it appears that IAX is not right for my system. I'd say this is only because you don't know enough about IAX yet ;-) [Many comments explaining how IAX would work wonderfully if all my VoIP hardware were replaced with IAX-compatible equipment] No, I don't want to replace existing gear. It would be expensive, disrupt operations, take lots of time to set up, and I don't want the administrative hassle of running multiple Asterisk systems for such a small network. There are other reasons, too. For example, the Cisco 827-4V is very reliable, because it has no hard drive and no fans. If *your* Asterisk system fails, you can zip over to Akihabara and get what you need, even on Sunday. Rue Montgallet is not the same! IAX is probably the ideal protocol for an interoffice trunk carrying many calls at once, but for me, it seems better to gradually migrate to a SIP-based system, with a single Asterisk server in Reno, and retaining present hardware. If NuFone service is reliable, good quality, competitively priced, and they also support the open source community, they'll get my business. But I'll connect via SIP, so existing equipment can be better utilized. When I'm in a hotel, stuck behind a NAT over which I have no control, sure, I'll use IAX to connect to the server (and tolerate the media proxy delays in that case.) Have I missed something? Regards, Stewart ___ 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
RE: [Asterisk-Users] Digium TheVoice recordings' sound terrible
Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] wrote: On Sun, 24 Oct 2004 07:47:17 -0400, Steve Totaro [EMAIL PROTECTED] wrote: 1. I am sure there are free programs that will allow you to adjust the files to sound more like the originial recordings as well as converting them to gsm. that's all very cool, but if you read my post carefully -- you did read more than just the subject line, did you not?! -- then you will find that I had mentioned I was unable to get acceptable results using sox because I don't know what parameters to use. It looks as if I didn't read your article carefully enough either. You can ignore the followup I just posted. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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
[Asterisk-Users] random crash at startup
Hi folks, I have upgraded asterisk from 0.8 to 1.0 on my gentoo server and it won't start now. It crashes on random points while loading the modules somewhere between res_crypto and chan_iax2 the last messages are either: === Yuck! Error in buffer handling...: Success Asterisk cleanly ending (2). === [app_random.so] = (Random goto) == Registered application 'Random' [app_transfer.so]Yuck! Error in buffer handling...: Broken pipe === [chan_zap.so] = (Zapata Telephony w/PRI) Asterisk cleanly ending (2). === [res_crypto.so] = (Cryptographic Digital Signatures) Found new ID3 Header Beginning asterisk shutdown Executing last minute cleanups == Destroying any remaining musiconhold processes Asterisk cleanly ending (2). === Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found [res_musiconhold.so] = (Music On Hold Resource) == Parsing '/etc/asterisk/musiconhold.conf': Found Oct 24 15:44:28 WARNING[1077071984]: res_musiconhold.c:561 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. Found new ID3 Header Warning, flexible rate not heavily tested! Beginning asterisk shutdown Executing last minute cleanups Asterisk cleanly ending (2). Ouch ... error while writing audio data: : Broken pipe === [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2)) Oct 24 15:44:49 WARNING[1077071984]: chan_iax2.c:7409 load_module: Unable to open IAX timing interface: No such device or address == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == Using TOS bits 16 == IAX Ready and Listening on 0.0.0.0 port 4569 Yuck! Error in buffer handling...: Success Asterisk cleanly ending (2). === [chan_sip.so] = (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found -- SIP Seeding 'janat' at [EMAIL PROTECTED]:5060 for 1800 Beginning asterisk shutdown Executing last minute cleanups == Destroying any remaining musiconhold processes Yuck! Error in buffer handling...: Broken pipe Asterisk cleanly ending (2). === and so on... what is it? tom ___ 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
Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible
Helpful URLS about SOX/wav/gsm Have you seen these? Converting: http://www.voip-info.org/wiki- Convert+WAV+audio+files+for+use+in+Asterisk Volume: http://www.voip-info.org/wiki-Asterisk+sound+files Other bits and bobs: http://www.marko.net/asterisk/archives/0212/0384.html e. On 24 Oct 2004, at 11:37, Benjamin on Asterisk Mailing Lists wrote: A customer has ordered some voice prompts from Digium's TheVoice online store. They say the recordings' sound was good when they listened to it on their Windoze boxes. However, then Asterisk is playing back the recordings, the volume is far too high and they sound really bad. This is particularly noticeable since the IVR menu mixes those ordered recordings with recordings that are already part of the Asterisk distribution. The volume of the included recordings are much lower and they sound much better than the ordered recordings. I wonder why Digium would deliver recordings that differ so much from the included set of recordings. However, the format the customer ordered was WAV, whereas all the included recordings are of course GSM. Has anybody had similar experiences? I tried to convert the WAV files to GSM using sox but since I don't know what parameters are best in this case, the results weren't satisfactory. Any suggestions? thanks 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. ___ 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 ___ 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
Re: [Asterisk-Users] random crash at startup
Did you upgrade zaptel and libpri before upgrading asterisk? - Original Message - From: Tomas Carnecky [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 9:44 AM Subject: [Asterisk-Users] random crash at startup Hi folks, I have upgraded asterisk from 0.8 to 1.0 on my gentoo server and it won't start now. It crashes on random points while loading the modules somewhere between res_crypto and chan_iax2 the last messages are either: === Yuck! Error in buffer handling...: Success Asterisk cleanly ending (2). === [app_random.so] = (Random goto) == Registered application 'Random' [app_transfer.so]Yuck! Error in buffer handling...: Broken pipe === [chan_zap.so] = (Zapata Telephony w/PRI) Asterisk cleanly ending (2). === [res_crypto.so] = (Cryptographic Digital Signatures) Found new ID3 Header Beginning asterisk shutdown Executing last minute cleanups == Destroying any remaining musiconhold processes Asterisk cleanly ending (2). === Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found [res_musiconhold.so] = (Music On Hold Resource) == Parsing '/etc/asterisk/musiconhold.conf': Found Oct 24 15:44:28 WARNING[1077071984]: res_musiconhold.c:561 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. Found new ID3 Header Warning, flexible rate not heavily tested! Beginning asterisk shutdown Executing last minute cleanups Asterisk cleanly ending (2). Ouch ... error while writing audio data: : Broken pipe === [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2)) Oct 24 15:44:49 WARNING[1077071984]: chan_iax2.c:7409 load_module: Unable to open IAX timing interface: No such device or address == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == Using TOS bits 16 == IAX Ready and Listening on 0.0.0.0 port 4569 Yuck! Error in buffer handling...: Success Asterisk cleanly ending (2). === [chan_sip.so] = (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found -- SIP Seeding 'janat' at [EMAIL PROTECTED]:5060 for 1800 Beginning asterisk shutdown Executing last minute cleanups == Destroying any remaining musiconhold processes Yuck! Error in buffer handling...: Broken pipe Asterisk cleanly ending (2). === and so on... what is it? tom ___ 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 ___ 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
[Asterisk-Users] Error when compiling asterisk-oh323
When I try to compile asterisk-oh323, errors as following: for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done make[1]: Entering directory `/root/willis/asterisk-oh323-0.6.3b/wrapper' ./check_ver /root/willis/pwlib pwlib ./check_ver /root/willis/openh323 openh323 gcc -shared -Wl,-soname,liboh323wrap.so -o liboh323wrap.so wrapper_misc.o asteriskaudio.o wrapendpoint.o wrapconnection.o wrapper.o wrapcaps.o make[1]: Leaving directory `/root/willis/asterisk-oh323-0.6.3b/wrapper' make[1]: Entering directory `/root/willis/asterisk-oh323-0.6.3b/asterisk-driver' gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -I/root/src/asterisk/include -I../wrapper -g -c -o chan_oh323.o chan_oh323.c chan_oh323.c: In function `oh323_call': chan_oh323.c:1237: structure has no member named `callerid' chan_oh323.c:1239: structure has no member named `callerid' chan_oh323.c:1241: structure has no member named `callerid' chan_oh323.c: In function `ast_oh323_new': chan_oh323.c:2381: structure has no member named `dnid' chan_oh323.c:2391: structure has no member named `callerid' chan_oh323.c:2392: structure has no member named `callerid' chan_oh323.c:2393: structure has no member named `callerid' chan_oh323.c:2398: structure has no member named `callerid' chan_oh323.c:2399: structure has no member named `callerid' chan_oh323.c:2400: structure has no member named `callerid' chan_oh323.c:2402: structure has no member named `callerid' chan_oh323.c:2407: structure has no member named `callerid' chan_oh323.c:2408: structure has no member named `callerid' chan_oh323.c:2410: structure has no member named `callerid' chan_oh323.c:2412: structure has no member named `callerid' chan_oh323.c:2416: structure has no member named `callerid' chan_oh323.c:2419: structure has no member named `ani' chan_oh323.c:2419: structure has no member named `callerid' chan_oh323.c:2425: structure has no member named `callerid' chan_oh323.c:2426: structure has no member named `callerid' chan_oh323.c: In function `load_module': chan_oh323.c:4697: warning: passing arg 4 of `ast_channel_register' from incompatible pointer type make[1]: *** [chan_oh323.o] Error 1 make[1]: Leaving directory `/root/willis/asterisk-oh323-0.6.3b/asterisk-driver' make: *** [subdirs_all] Error 1 ___ 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
Re: [Asterisk-Users] random crash at startup
Steve Totaro wrote: Did you upgrade zaptel and libpri before upgrading asterisk? do I need zaptel? I have libpri-1.0.0 but no zaptel installed. in the gentoo ebuild the dependecy is like thik: DEPEND=virtual/libc media-sound/mpg123 dev-libs/newt doc? ( app-doc/doxygen ) alsa? ( media-libs/alsa-lib ) mysql? ( dev-db/mysql ) gtk? ( =x11-libs/gtk+-1.2* ) !nopri? ( =net-libs/libpri-1.0.0 ) !nozaptel? ( =net-misc/zaptel-1.0.0 =net-libs/zapata-1.0.0 ) and I did USE=~doc ~alsa ~mysql ~gtk nopri nozaptel emerge asterisk do I need alsa, mysql, libpri or zaptel? tom ___ 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
[Asterisk-Users] (iax|sip)friends in extconfig?
hi I'm currently using sipfriends from asterisk-stable and I've enabled MYSQL_USERS as well. Are mysql/odbc/whatever _users_ available in extconfig yet? roy ___ 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
Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible
On October 24, 2004 07:47 am, Steve Totaro wrote: 2 . If you dont want to go through all of that, kindly ask Digium to have the files fixed for you. I seriously doubt they have their own sound stage and most likely outsource this type of business. Chances are the people they outsource the business to are experts and have sophisticated equipment to do this with very fast turn around time. Actually The Voice is Alison, and she does work at Digium. -A. ___ 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
Re: [Asterisk-Users] iLBC/PCM16 Huge Cost
On October 23, 2004 10:58 pm, Michael Loftis wrote: mmm... any packaging is better than none. I regularly destroy things on systems when it's not been put into proper packaging because we upgrade the system, and there's no record of something being installed, nor what it depends on, so it gets broken. http://asic-linux.com.mx/~izto/checkinstall/ Checkinstall is your friend. RPM, DEB and Slackware .TGZ formats supported. -A. ___ 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
[Asterisk-Users] Re: chan_sip CallerPres support?
would it be hard to implement CallerPres support in chan_sip? There is support for outgoing calls, but this patch breakes incoming callerid: http://bugs.digium.com/bug_view_page.php?bug_id=0002471 Greez Andreas _ Listen to music online with the Xtra Broadband Channel http://xtra.co.nz/broadband ___ 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
Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible
I know she works at Digium but they probably go down the street to a real sound stage to do the recordings via 3rd party. A sound stage is a facility used to create and process professional recordings. They can be used by anyone employed by an company. - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 10:05 AM Subject: Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible On October 24, 2004 07:47 am, Steve Totaro wrote: 2 . If you dont want to go through all of that, kindly ask Digium to have the files fixed for you. I seriously doubt they have their own sound stage and most likely outsource this type of business. Chances are the people they outsource the business to are experts and have sophisticated equipment to do this with very fast turn around time. Actually The Voice is Alison, and she does work at Digium. -A. ___ 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 ___ 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
Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible
I know she works at Digium but they probably go down the street to a real sound stage to do the recordings via 3rd party. A sound stage is a facility used to create and process professional recordings. They can be used by anyone employed by an company. http://www.theivrvoice.com/ would seem to imply otherwise. I'd be a bit surprised if any company had enough work to keep her employed full-time, so the works at Digium line sounds a bit fishy to me. Regards, ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ 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
Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible
On October 24, 2004 10:24 am, Steve Totaro wrote: I know she works at Digium but they probably go down the street to a real sound stage to do the recordings via 3rd party. Oh I dunno, for telephone IVR you don't need much of a sound stage. Convert a bathroom into one with a lot of insulation, thick carpet and soft walls. Throw in a good mic, a decent preamp and mixer to adjust levels and that's about all you'd need for doing good quality IVR voicing. It's not like this is going on to a CD or something. :-) -A. ___ 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
Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible
On Sun, 24 Oct 2004 14:47:56 +0100, Elliot Moore [EMAIL PROTECTED] wrote: Helpful URLS about SOX/wav/gsm Have you seen these? [snip URLs] yes, I have played with those and all I did achieve was making the recordings worse, but thanks anyway. However, it seems now that this is not a common problem so I have to check back with the customer whether these files are really the originals. You never know, they might have messed around with those. If anything else fails I will ask them to reorder the prompts once more and get them delivered as GSM. thanks again 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. ___ 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
Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible
On Sun, 24 Oct 2004 09:27:49 -0500 (CDT), Joe Greco [EMAIL PROTECTED] wrote: http://www.theivrvoice.com/ would seem to imply otherwise. I'd be a bit surprised if any company had enough work to keep her employed full-time, so the works at Digium line sounds a bit fishy to me. I think when he wrote 'She does work at Digium' it was meant in the sense of She is doing work at Digium', or 'She does (some) work for Digium ;-) 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. ___ 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
Re: [Asterisk-Users] random crash at startup
get the new versions of libpri zaptel and asterisk and install them in that order. should work. - Original Message - From: Tomas Carnecky [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 10:05 AM Subject: Re: [Asterisk-Users] random crash at startup Steve Totaro wrote: Did you upgrade zaptel and libpri before upgrading asterisk? do I need zaptel? I have libpri-1.0.0 but no zaptel installed. in the gentoo ebuild the dependecy is like thik: DEPEND=virtual/libc media-sound/mpg123 dev-libs/newt doc? ( app-doc/doxygen ) alsa? ( media-libs/alsa-lib ) mysql? ( dev-db/mysql ) gtk? ( =x11-libs/gtk+-1.2* ) !nopri? ( =net-libs/libpri-1.0.0 ) !nozaptel? ( =net-misc/zaptel-1.0.0 =net-libs/zapata-1.0.0 ) and I did USE=~doc ~alsa ~mysql ~gtk nopri nozaptel emerge asterisk do I need alsa, mysql, libpri or zaptel? tom ___ 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 ___ 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
RE: [Asterisk-Users] KSS/BLF on Asterisk
[EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: Folks, I am trying to determine the best way to allow a station to monitor the status of another station. For example: a reception set needing to see the status of 20 or 30 phones OR an executive assistant wanting to have appearances of several other extensions, in order to monitor their status and assist with call handling. I know Snom has a phone that you can attach an add-on module to, but I don't know how you'd program Asterisk to deliver status information to those buttons. Get a hint! :-) Check out the hint priority in extensions.conf. There are also some details in the wiki. Thanks. I've taken a look at that beastie and I'll hack around with it to see what it can do. Nick ___ 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 ___ 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
Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible
Hi Benj, On Sun, 24 Oct 2004 23:39:06 +0900, Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] wrote: [snip URLs] yes, I have played with those and all I did achieve was making the recordings worse, but thanks anyway. However, it seems now that this is not a common problem so I have to check back with the customer whether these files are really the originals. You never know, they might have messed around with those. If anything else fails I will ask them to reorder the prompts once more and get them delivered as GSM. I've never ordered from thevoice, but I have converted some MP3 to gsm and after fighting with sox parameteres I came up with this: sox in.wav -t gsm -r 8000 -g -b -c 1 out.gsm resample -ql vol 0.6 You can adjust the volume changing the last parameter. It works good enough for me. With the wiki examples I got mixed result, like a changed pitch, or the same pitch but a really slow speed for some recordings. Best regards, -- Nicolás Gudiño Buenos Aires - Argentina ___ 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
Re: [Asterisk-Users] Re: Direct SIP connection to Vonage service
On Sun, 24 Oct 2004 15:27:53 +0200, Stewart Nelson [EMAIL PROTECTED] wrote: No, I don't want to replace existing gear. fair enough. It would be expensive that I don't agree with, especially not if you do it yourself, but anyway. There are other reasons, too. For example, the Cisco 827-4V is very reliable, because it has no hard drive and no fans. If *your* Asterisk system fails, you can zip over to Akihabara and get what you need, even on Sunday. Rue Montgallet is not the same! Unattended remote systems I run off Compact Flash instead of hard drives. I can't beat the no fan thing though ;-) for me, it seems better to gradually migrate to a SIP-based system, with a single Asterisk server in Reno, and retaining present hardware. I still think it would make sense to run one Asterisk server in Paris and another in Reno. You don't really need more than that, but of course it's your choice. When I'm in a hotel, stuck behind a NAT over which I have no control, sure, I'll use IAX to connect to the server (and tolerate the media proxy delays in that case.) There are no media proxy delays if you use an IAX provider. IAX calls skip intermediary nodes and go from end to end. That was the point of my previous message. If you use H.323 gateway with a SIP provider or vice versa, then you will also have a media proxy in between because SIP gear cannot reinvite H.323 gear, there's got to be something in between that translates. If your entire route is all SIP, then you can go from end to end. Like wise if youre entire route is all IAX, then you can go from end to end, too. Have I missed something? Yeah, wine and cheese is better, cheaper and more plentiful in Paris than in Tokyo and that beats shopping in Akihabara anytime ;-) 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. ___ 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
Re: [Asterisk-Users] random crash at startup
Steve Totaro wrote: get the new versions of libpri zaptel and asterisk and install them in that order. should work. I have libpri-1.0.0 and now I've reinstalled asterisk-1.0.0 but it still doesn't work, and there is no Digium hardware in the server so I don't need zaptel. tom ___ 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
[Asterisk-Users] Failed to authenticate on INVITE to '601 ...
I have installed the first time Asterisk, (forgive me simple questions) I have also installed the demo. After testing demo (call 1000, call 600, ...) I changed in the extensions.conf: ; include = demo include = incomingsipgate include = sipgate.de include = sipgate.col.uk [incomingsipgate] exten = 5552220,1,Dial(SIP/601,20,r) exten = 4782156,1,Dial(SIP/602,20,r) [sipgate.de] exten = _0049X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r) exten = _0049X.,2,Playback(invalid) exten = _0049X.,3,Hangup [sipgate.co.uk] exten = _0044X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr) exten = _0044X.,2,Playback(invalid) exten = _0044X.,3,Hangup in sip.conf I have: register = 5552220:[EMAIL PROTECTED]/5552220 register = 4782156:[EMAIL PROTECTED]/4782156 [601] type=friend username=601 secret=pwd-601 canreinvite=no host=dynamic defaultip=61.220.121.19 dtmfmode=rfc2833 mailbox=601 nat=yes caller-id=Ronald 1 601 [602] type=friend username=602 secret=pwd-602 canreinvite=no host=dynamic defaultip=61.220.121.19 dtmfmode=rfc2833 mailbox=602 nat=yes caller-id=Ronald 2 602 [sipgate.de] type=friend username=5552220 secret=pwd-de host=sipgate.de fromuser=5552220 fromdomain=sipgate.de nat=yes context=incomingsipgate canreinvite=no [sipgate.co.uk] type=friend username=4782156 secret=pwd-uk host=sipgate.co.uk fromuser=4782156 fromdomain=sipgate.co.uk nat=yes context=incomingsipgate canreinvite=no The console shows when I want to dial at sipgate.de the number 1 (test) or 5 (Voicemail): 00491 -- Executing Dial(SIP/601-ea8b, SIP/[EMAIL PROTECTED]|30|r) in new stack -- Called [EMAIL PROTECTED] Oct 24 23:00:48 NOTICE[213005]: c6779 handle_response: Failed to authenticate on INVITE to '601 sip:[EMAIL PROTECTED];tag=as25c3e254' -- Nobody picked up in 3 ms -- Executing Playback(SIP/601-ea8b, invalid) in new stack -- Playing 'invalid' (language 'en') -- Got SIP response 481 Call Leg Does Not Exist back from 217.10.79.9 -- Executing Hangup(SIP/601-ea8b, ) in new stack == Spawn extension (default, 00491, 3) exited non-zero on 'SIP/601-ea8b' -- Executing Hangup(SIP/601-ea8b, ) in new stack == Spawn extension (default, h , 1) exited non-zero on 'SIP/601-ea8b' What do I miss? bye Ronald begin:vcard fn:Ronald Wiplinger n:Wiplinger;Ronald email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE version:2.1 end:vcard ___ 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
Re: [Asterisk-Users] Fedora 2, Kudzu and X100P
On Sun, 2004-10-24 at 05:10, Tzafrir Cohen wrote: One obvious solution is not to automatically load kudzu. chkconfig --remove kudzu Another obvious solution of the same sort is modprobing the zaptel module earlier in the boot process. I can't seem to figure out , though, where kudzu takes its modue names from. I haven't bothred reading th source yet, though (not from /usr/share/hwdata, it seems) The problem is that once Kudzu runs it configures linux to always load the incorrect module for the card. I have already erased kudzu from the server, recompiled Zaptel and modified modules.dep by hand but if any application runs a depmod -a the configuration for the other module returns. If I do a modprobe zaptel it will always load the other module. -- Carlos Chvez Director de Tecnologa Telecomunicaciones Abiertas de Mxico ___ 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
Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible
On Sun, 24 Oct 2004 12:01:19 -0300, Nicolás Gudiño [EMAIL PROTECTED] wrote: I've never ordered from thevoice, but I have converted some MP3 I guess you mean WAV to gsm and after fighting with sox parameteres I came up with this: sox in.wav -t gsm -r 8000 -g -b -c 1 out.gsm resample -ql vol 0.6 thanks a lot, I will try that. 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. ___ 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
Re: [Asterisk-Users] New Channel Driver: chan_bluetooth
Yep, same here with Ericsson T610 Reason: AT+BRSF is not implemented in Ericsson Cellphones. Kind regards, Martin List-Petersen http://www.marlow.dk/ On Wed, 2004-10-20 at 22:20, Jon Radon wrote: Running Asterisk CVS-HEAD-10/19/04-04:34:45, just tested with my Sony Ericsson T68i. Couldn't get it to connect, got the error message below. It then just sat saying it was negotiating. I tried to disable BTP, but that didn't help. Oct 20 17:10:14 NOTICE[196620]: chan_bluetooth.c:1769 try_connect: Initialised bluetooth link to device t68i Oct 20 17:10:14 ERROR[196620]: chan_bluetooth.c:2152 handle_rd_data: Expected '\n' Oct 20 17:10:14 ERROR[196620]: chan_bluetooth.c:2083 handle_rd_data: Expected '\r' Oct 20 17:10:14 ERROR[196620]: chan_bluetooth.c:2083 handle_rd_data: Expected '\r' Oct 20 17:10:14 ERROR[196620]: chan_bluetooth.c:2083 handle_rd_data: Expected '\r' Oct 20 17:10:14 ERROR[196620]: chan_bluetooth.c:2083 handle_rd_data: Expected '\r' Oct 20 17:10:14 ERROR[196620]: chan_bluetooth.c:2083 handle_rd_data: Expected '\r' ___ 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
Re: [Asterisk-Users] random crash at startup
just try it. - Original Message - From: Tomas Carnecky [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 11:05 AM Subject: Re: [Asterisk-Users] random crash at startup Steve Totaro wrote: get the new versions of libpri zaptel and asterisk and install them in that order. should work. I have libpri-1.0.0 and now I've reinstalled asterisk-1.0.0 but it still doesn't work, and there is no Digium hardware in the server so I don't need zaptel. tom ___ 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 ___ 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
Re: [Asterisk-Users] Failed to authenticate on INVITE to '601 ... solved
Ronald Wiplinger wrote: I have installed the first time Asterisk, (forgive me simple questions) I have also installed the demo. I solved it with the newest cvs version !!! bye Ronald After testing demo (call 1000, call 600, ...) I changed in the extensions.conf: ; include = demo include = incomingsipgate include = sipgate.de include = sipgate.col.uk [incomingsipgate] exten = 5552220,1,Dial(SIP/601,20,r) exten = 4782156,1,Dial(SIP/602,20,r) [sipgate.de] exten = _0049X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r) exten = _0049X.,2,Playback(invalid) exten = _0049X.,3,Hangup [sipgate.co.uk] exten = _0044X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr) exten = _0044X.,2,Playback(invalid) exten = _0044X.,3,Hangup in sip.conf I have: register = 5552220:[EMAIL PROTECTED]/5552220 register = 4782156:[EMAIL PROTECTED]/4782156 [601] type=friend username=601 secret=pwd-601 canreinvite=no host=dynamic defaultip=61.220.121.19 dtmfmode=rfc2833 mailbox=601 nat=yes caller-id=Ronald 1 601 [602] type=friend username=602 secret=pwd-602 canreinvite=no host=dynamic defaultip=61.220.121.19 dtmfmode=rfc2833 mailbox=602 nat=yes caller-id=Ronald 2 602 [sipgate.de] type=friend username=5552220 secret=pwd-de host=sipgate.de fromuser=5552220 fromdomain=sipgate.de nat=yes context=incomingsipgate canreinvite=no [sipgate.co.uk] type=friend username=4782156 secret=pwd-uk host=sipgate.co.uk fromuser=4782156 fromdomain=sipgate.co.uk nat=yes context=incomingsipgate canreinvite=no The console shows when I want to dial at sipgate.de the number 1 (test) or 5 (Voicemail): 00491 -- Executing Dial(SIP/601-ea8b, SIP/[EMAIL PROTECTED]|30|r) in new stack -- Called [EMAIL PROTECTED] Oct 24 23:00:48 NOTICE[213005]: c6779 handle_response: Failed to authenticate on INVITE to '601 sip:[EMAIL PROTECTED];tag=as25c3e254' -- Nobody picked up in 3 ms -- Executing Playback(SIP/601-ea8b, invalid) in new stack -- Playing 'invalid' (language 'en') -- Got SIP response 481 Call Leg Does Not Exist back from 217.10.79.9 -- Executing Hangup(SIP/601-ea8b, ) in new stack == Spawn extension (default, 00491, 3) exited non-zero on 'SIP/601-ea8b' -- Executing Hangup(SIP/601-ea8b, ) in new stack == Spawn extension (default, h , 1) exited non-zero on 'SIP/601-ea8b' What do I miss? bye Ronald ___ 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 -- Ronald Wiplinger Senior Software Engineer AGP Telecom Co. Ltd. Tel. (O) +886-2-2741-7890 # 7303, (M) +886-939-77-55-16 (from USA dial (408)253-3153 # 7303) -Disclaimer--- This document is intended for transmission to the named recipient only. If you are not that person, you should note that legal rights reside in this document and you are not authorized to access, read, disclose, copy, use or otherwise deal with it and any such actions are prohibited and may be unlawful. The views expressed in this document are not necessarily those of AGP Telecom Co., Ltd. Notice is hereby given that no representation, contract or other binding obligation shall be created by this e-mail, which must be interpreted accordingly. Any representations, contractual rights or obligations shall be separately communicated in writing and signed in the original by a duly authorized officer of the relevant company. -- begin:vcard fn:Ronald Wiplinger n:Wiplinger;Ronald email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE version:2.1 end:vcard ___ 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
Re: [Asterisk-Users] Fedora 2, Kudzu and X100P
Just an idea, couldnt you remove the zaptel hardware, run kudzu and remove the hardware module via kudzu then disable kudzu again? - Original Message - From: Carlos Chavez [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 11:15 AM Subject: Re: [Asterisk-Users] Fedora 2, Kudzu and X100P On Sun, 2004-10-24 at 05:10, Tzafrir Cohen wrote: One obvious solution is not to automatically load kudzu. chkconfig --remove kudzu Another obvious solution of the same sort is modprobing the zaptel module earlier in the boot process. I can't seem to figure out , though, where kudzu takes its modue names from. I haven't bothred reading th source yet, though (not from /usr/share/hwdata, it seems) The problem is that once Kudzu runs it configures linux to always load the incorrect module for the card. I have already erased kudzu from the server, recompiled Zaptel and modified modules.dep by hand but if any application runs a depmod -a the configuration for the other module returns. If I do a modprobe zaptel it will always load the other module. -- Carlos Chvez Director de Tecnologa Telecomunicaciones Abiertas de Mxico ___ 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 ___ 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
Re: [Asterisk-Users] random crash at startup
Steve Totaro wrote: just try it. I installed the old 0.9 version of asterisk and now it works, even with libpri-1.0.0. I've found out thet the kernel module ztdummy wasn't loaded while I tried to start asterisk, could this have been the problem? tom ___ 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
Re: [Asterisk-Users] random crash at startup
more info on ztdummy and zaptel i am sure will solve your issue. http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer%20ztdummy - Original Message - From: Tomas Carnecky [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 11:42 AM Subject: Re: [Asterisk-Users] random crash at startup Steve Totaro wrote: just try it. I installed the old 0.9 version of asterisk and now it works, even with libpri-1.0.0. I've found out thet the kernel module ztdummy wasn't loaded while I tried to start asterisk, could this have been the problem? tom ___ 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 ___ 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
[Asterisk-Users] Re: doublehash patch for 1.0.1
Just tried the patch you made with the latest CVS and it patches fine although it does not work. Now when I hit # it does not send the DTMF to the other side at all. Although hitting ## does get the transfer. Now # doesn't do ANYTHING :) I'm not sure why that is, it works with all our phones (Grandstream BT101s and analog phones on Grandstream ATA286s). I just tested by calling my bank's IVR. applied patch. went great. now single # does not transfer and double does. but, i am having the same problem as matthew, the # does not go through at dmtf. all other keys go through as dmtf, just not the #. this is on a spa3k. clearly * is receiving the #, as ## does do a transfer. so why is a single # not being sent onward as dtmf? randy --- ps. and i have a general wonder/question about this. is someone who uses a commercial pbx, say a meridian or whatever, unable to use ivr systems because # is not sent? ___ 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
Re: [Asterisk-Users] Re: doublehash patch for 1.0.1
99% of the companies I call that say hit # after entering your response, doesnt actually require the #.. I've only encountered a few places that if you don't hit the # it ignores your response, eventually sending you to an operator or hangs up on you. On Sun, 24 Oct 2004 09:37:09 -0700, Randy Bush [EMAIL PROTECTED] wrote: Just tried the patch you made with the latest CVS and it patches fine although it does not work. Now when I hit # it does not send the DTMF to the other side at all. Although hitting ## does get the transfer. Now # doesn't do ANYTHING :) I'm not sure why that is, it works with all our phones (Grandstream BT101s and analog phones on Grandstream ATA286s). I just tested by calling my bank's IVR. applied patch. went great. now single # does not transfer and double does. but, i am having the same problem as matthew, the # does not go through at dmtf. all other keys go through as dmtf, just not the #. this is on a spa3k. clearly * is receiving the #, as ## does do a transfer. so why is a single # not being sent onward as dtmf? randy --- ps. and i have a general wonder/question about this. is someone who uses a commercial pbx, say a meridian or whatever, unable to use ivr systems because # is not sent? ___ 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 -- MBM ___ 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
RE: [Asterisk-Users] Call Waiting
If this is call waiting on the CO line, I found to flash the CO line you have to (flash *0) to answer it. If it is another station calling your phone while you are on , a normal flash will do. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nikhil Jogia Sent: Sunday, October 24, 2004 6:54 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Call Waiting Hi, I have just set up an Asterisk box.it sure is a big job to get everything perfect, especially when you have picky users. Anyway, the box has 2 X100P's and a couple of sipura spa-2000's connected to the LAN. 1 of the lines connected to the X100P's goes straight to extension 1000 after a short greeting. Anyway, say ext 1000 is talking to ext 1001, and the line rings. Ext 1000 hears a small beep every few seconds. This is obviously call waiting. My question is how do I answer that incoming call whilst on a call? I have looked around, tried *0 and even 0*, the flash key, but to no avail :( If this is call waiting on the CO line, I found to flash the CO line you have to (flash *0) to answer it. If it is another station calling your phone while you are on , a normal flash will do. ___ 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 ___ 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
Re: [Asterisk-Users] IAXy setup
Hummm.. thats cool.. I havent tougth about being re-provisioning the iaxy box :)... But how do you detect the dns change? wich ddns company are u using? Jim Van Meggelen wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wilson Pickett Sent: October 21, 2004 5:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXy setup A little late, but about DNS for IAXY request name resolution, no? Also, what happens if your IP address changes at the Asterisk end? You'll have to connect to all your IAXys to Here's what I do: when the server ip changes, I auto reprovision the IAXy. It works better than the Grandstreams, which have to be rebooted when the ip changes. They do DNS, but only at boot time apparently! ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.775 / Virus Database: 522 - Release Date: 08/10/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.775 / Virus Database: 522 - Release Date: 08/10/2004 ___ 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 ___ 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
Re: [Asterisk-Users] Call Waiting
ah good thinking, i didnt even factor CO call waiting into the equation - Original Message - From: Henry Devito [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 1:19 PM Subject: RE: [Asterisk-Users] Call Waiting If this is call waiting on the CO line, I found to flash the CO line you have to (flash *0) to answer it. If it is another station calling your phone while you are on , a normal flash will do. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nikhil Jogia Sent: Sunday, October 24, 2004 6:54 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Call Waiting Hi, I have just set up an Asterisk box.it sure is a big job to get everything perfect, especially when you have picky users. Anyway, the box has 2 X100P's and a couple of sipura spa-2000's connected to the LAN. 1 of the lines connected to the X100P's goes straight to extension 1000 after a short greeting. Anyway, say ext 1000 is talking to ext 1001, and the line rings. Ext 1000 hears a small beep every few seconds. This is obviously call waiting. My question is how do I answer that incoming call whilst on a call? I have looked around, tried *0 and even 0*, the flash key, but to no avail :( If this is call waiting on the CO line, I found to flash the CO line you have to (flash *0) to answer it. If it is another station calling your phone while you are on , a normal flash will do. ___ 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 ___ 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 ___ 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
Re: [Asterisk-Users] Re: doublehash patch for 1.0.1
clearly * is receiving the #, as ## does do a transfer. so why is a single # not being sent onward as dtmf? I noticed on X-Lite that # in a dialstring is sent URL-encoded or similar, and Asterisk doesn't understand it. Could this be something similar? Perhaps sip debug will reveal? Steve ___ 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
Re: [Asterisk-Users] Uniden UIP 200 Phone and NAT?
On Sat, 2004-23-10 at 19:43 -0500, Me wrote: Any chance you can pass me the Beta Version or let me know how to get it myself? I'm sorry, I can't distribute the beta. You can ask Uniden support - although I doubt they'd hand it out willingly. I wouldn't expect much more of a wait before an official version is released. Ryan ___ 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
Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible
Benjamin on Asterisk Mailing Lists wrote: On Sun, 24 Oct 2004 09:27:49 -0500 (CDT), Joe Greco [EMAIL PROTECTED] wrote: http://www.theivrvoice.com/ would seem to imply otherwise. I'd be a bit surprised if any company had enough work to keep her employed full-time, so the works at Digium line sounds a bit fishy to me. I think when he wrote 'She does work at Digium' it was meant in the sense of She is doing work at Digium', or 'She does (some) work for Digium ;-) rgds benjk I would lean towards she does some work for Digium. Did you check out her webpage? One, she lives in Canada, so she certainly does not work at Digium in the physical sense. Two, her client list leads me to believe that Digium is probably one of her smaller clients. Did you try to contact her directly? She seems to imply on her site that customer satisfaction is pretty important to her. Maybe she will fix them for you. You did after all pay for these files, right? ;) She must have a fairly good relationship with Digium, however, because she does have the official title of Asterisk Diva, and she was at Astricon. -- Kristian Kielhofner ___ 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
Re: [Asterisk-Users] MusicOnHold() - how to restart player from the beginning on each call? (fwd)
Looks like what you want is not music on-hold, but rather a streaming server On Oct 22, 2004, at 4:23 PM, Ryan Courtnage wrote: On Fri, 2004-22-10 at 16:05 -0400, Kanwar Ranbir Sandhu wrote: On Fri, 2004-10-22 at 05:56, Manfred Petz wrote: [snip] Is there a way to force MusicOnHold() to be restarted from the beginning for each call which has been answered? [snip] Why? What would be the point? off the top of my head ... promotional messages. Manfred - I don't think there is a graceful way to do this. I know that if you do a killall mpg123 at your command line, the next call MOH answers will start playing the mp3s at the beginning. Of course this would affect others that are listening, but if you build out some logic you might be able to make some use of it. Ryan ___ 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 ___ 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
Re: [Asterisk-Users] MusicOnHold() - how to restart player from the beginning on each call? (fwd)
Why not just create a context that plays static msgs whenever someone is transfered thereThank you for calling Monthly special etc ... then transfer them back when the person at the biz picks up On Sun, 24 Oct 2004 14:23:04 -0400, Emilio Panighetti [EMAIL PROTECTED] wrote: Looks like what you want is not music on-hold, but rather a streaming server On Oct 22, 2004, at 4:23 PM, Ryan Courtnage wrote: On Fri, 2004-22-10 at 16:05 -0400, Kanwar Ranbir Sandhu wrote: On Fri, 2004-10-22 at 05:56, Manfred Petz wrote: [snip] Is there a way to force MusicOnHold() to be restarted from the beginning for each call which has been answered? [snip] Why? What would be the point? off the top of my head ... promotional messages. Manfred - I don't think there is a graceful way to do this. I know that if you do a killall mpg123 at your command line, the next call MOH answers will start playing the mp3s at the beginning. Of course this would affect others that are listening, but if you build out some logic you might be able to make some use of it. Ryan ___ 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 ___ 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 ___ 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
RE: [Asterisk-Users] Geotel integration with Asterisk
Geotel is a company that Cisco bought which provides call control across geographically dispersed locations. The simplest application is being able to query call queue status at another location. For example, a call comes in and can be sent to one of three call center locations. Geotel can query each location to see who is the least busy for this type of call. Traditionally it has been VERY expensive. We provide some primitive Geotel functions in-the-cloud right now. For example, we can know how many live calls are going to a location before we send the call. We can set thresholds (e.g. if a location A has over 100 concurrent calls send them to location B). Geotel can theoretically provide this and carry it further. I think there is some nice enterprise reporting that can come from the Geotel as well. G. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Saturday, October 23, 2004 4:30 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Geotel integration with Asterisk Ok lets get this out of the way... WTF is Geotel? bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Greg Smith Sent: Saturday, October 23, 2004 4:19 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Geotel integration with Asterisk Has any one integrated to a Geotel with Asterisk? Thanks. Greg Advanta ___ 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 ___ 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
[Asterisk-Users] getting cid from spa3k pstn to *
in order to get the cid from the spa3k to *, i need to turn on PSTN / PSTN-To-VoIP Gateway Setup / PSTN CID For VoIP CID: = YES this produces a sip invite as follows: Frame 1 (1092 bytes on wire, 1092 bytes captured) Ethernet II, Src: 00:90:69:6d:e8:00, Dst: 00:30:48:80:b3:72 Internet Protocol, Src Addr: 42.666.11.7 (42.666.11.7), Dst Addr: 666.42.7.11 (666.42.7.11) User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Method: INVITE Resent Packet: False Message Header Via: SIP/2.0/UDP 42.666.11.7:5061;branch=z9hG4bK-388cd0e7 From: CID Namesip:[EMAIL PROTECTED];tag=42d678b4c352ea69o1 To: sip:[EMAIL PROTECTED] Remote-Party-ID: CID Namesip:[EMAIL PROTECTED];screen=yes;party=calling Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE Max-Forwards: 70 Contact: spa3k pstn sip:[EMAIL PROTECTED]:5061 Expires: 240 User-Agent: Sipura/SPA3000-2.0.11(GWa) Content-Length: 430 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp Message body Session Description Protocol note that the From: has the cid, as does the Remote-Party-ID:. and the Contact: has the spa3k's id and display name. as the sip.conf entry looks like [spa3k] type=friend host=dynamic port=5061 auth=md5 secret=hidden qualify=1000 dtmfmode=rfc2833 canreinvite=yes context=spa3k-ext the From: and/or Remote-Party-ID: cause asterisk respond with 407 Proxy Authentication Required, to which the spa3k responds Frame 3 (450 bytes on wire, 450 bytes captured) Ethernet II, Src: 00:90:69:6d:e8:00, Dst: 00:30:48:80:b3:72 Internet Protocol, Src Addr: 42.666.11.7 (42.666.11.7), Dst Addr: 666.42.7.11 (666.42.7.11) User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: ACK sip:[EMAIL PROTECTED] SIP/2.0 Method: ACK Resent Packet: False Message Header Via: SIP/2.0/UDP 42.666.11.7:5061;branch=z9hG4bK-388cd0e7 From: CID Namesip:[EMAIL PROTECTED];tag=42d678b4c352ea69o1 To: sip:[EMAIL PROTECTED];tag=as2741cf03 Call-ID: [EMAIL PROTECTED] CSeq: 101 ACK Max-Forwards: 70 Contact: spa3k pstn sip:[EMAIL PROTECTED]:5061 User-Agent: Sipura/SPA3000-2.0.11(GWa) Content-Length: 0 and it all goes to hell from there. if i set the spa3k config to have PSTN / PSTN-To-VoIP Gateway Setup / PSTN CID For VoIP CID: = NO Frame 1 (1072 bytes on wire, 1072 bytes captured) Ethernet II, Src: 00:90:69:6d:e8:00, Dst: 00:30:48:80:b3:72 Internet Protocol, Src Addr: 42.666.11.7 (42.666.11.7), Dst Addr: 666.42.7.11 (666.42.7.11) User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Method: INVITE Resent Packet: False Message Header Via: SIP/2.0/UDP 42.666.11.7:5061;branch=z9hG4bK-f5998d8a From: spa3k pstn sip:[EMAIL PROTECTED];tag=8fc58211a0dc60f2o1 To: sip:[EMAIL PROTECTED] Remote-Party-ID: spa3k pstn sip:[EMAIL PROTECTED];screen=yes;party=calling Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE Max-Forwards: 70 Contact: spa3k pstn sip:[EMAIL PROTECTED]:5061 Expires: 240 User-Agent: Sipura/SPA3000-2.0.11(GWa) Content-Length: 430 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp Message body Session Description Protocol the connection completes, but asterisk does not have the pstn caller id. randy ___ 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
[Asterisk-Users] Re: [Asterisk-Dev] New Channel Driver: chan_bluetooth
On Wed, 20 Oct 2004 13:37:28 +0100, Theo Zourzouvillys [EMAIL PROTECTED] wrote: after a couple of days work banging my head against the wall (bloody standards my arse), i've got chan_bluetooth to a point where it's starting to function - certianly more than just proof of concept now. Will this need a mobile phone with bluetooth support or can you use a mere bluetooth headset as a client, if so how would you pick up and hang up or dial? I would really love to play with that, but as everybody knows, Japan is s incredibly far advanced in mobile phones that the only phone I could buy here which does have bluetooth works only as a data modem for a Windoze notebook, no voice, no headset support, no addressbook syncing, which is why I then bought an iPod for the $250 the phone would have cost. 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. ___ 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
[Asterisk-Users] Asterisk Prepaid with MySQL
Hi, Anyone could use Asterisk Prepaid with a MySQL database? Thanks. Nahuel Ramos. ___ 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
[Asterisk-Users] Compiling zaptel
Hi, I've been running * for a couple of month now. However, now i want to run ztdummy. Compiling works (apart from some warning regarding strict-aliasing), however installation gives missing Unresolved symbols: if [ -d /etc/modutils ]; then \ /sbin/update-modules ; \ fi depmod: *** Unresolved symbols in /lib/modules/2.4.27-1-686/misc/wcusb.o depmod: *** Unresolved symbols in /lib/modules/2.4.27-1-686/misc/zaptel.o depmod: *** Unresolved symbols in /lib/modules/2.4.27-1-686/misc/ztd-eth.o /sbin/depmod -a depmod: *** Unresolved symbols in /lib/modules/2.4.27-1-686/misc/wcusb.o depmod: *** Unresolved symbols in /lib/modules/2.4.27-1-686/misc/zaptel.o depmod: *** Unresolved symbols in /lib/modules/2.4.27-1-686/misc/ztd-eth.o [ -f /etc/zaptel.conf ] || install -m 644 zaptel.conf.sample /etc/zaptel.conf I ran (as suggested on this list) depmod -ae with the following information: pbx:/usr/src/zaptel# depmod -ae depmod: *** Unresolved symbols in /lib/modules/2.4.27-1-686/misc/wcusb.o depmod: usb_submit_urb_R8c511495 depmod: usb_register_Re5a350f5 depmod: usb_deregister_R7002f0f3 depmod: usb_control_msg_R2ea78c08 depmod: usb_set_interface_Ra1eb4dc9 depmod: usb_set_configuration_Rd5d22e8d depmod: *** Unresolved symbols in /lib/modules/2.4.27-1-686/misc/zaptel.o depmod: create_proc_entry_Rfe079653 depmod: remove_wait_queue_R9bd53081 depmod: proc_mkdir_Reb28b76c depmod: remove_proc_entry_R63256fd5 depmod: __pollwait_R05dbf31f depmod: add_wait_queue_R27b3fac5 depmod: register_chrdev_R007abe28 depmod: *** Unresolved symbols in /lib/modules/2.4.27-1-686/misc/ztd-eth.o depmod: skb_over_panic_R0f38fc6b depmod: dev_queue_xmit_Rea2fb899 depmod: dev_remove_pack_Rdf0e2a5d depmod: alloc_skb_R0f71f7ff depmod: dev_get_by_name_R14968228 depmod: __kfree_skb_R72b0e92c depmod: dev_add_pack_R52914681 depmod: skb_under_panic_Rf1c9f235 Here is a list of the relevant modules loaded: pbx:/usr/src/zaptel# lsmod|grep usb usb-uhci 23344 0 (unused) usbcore62924 1 [usb-uhci] pbx:/usr/src/zaptel# lsmod|grep ppp ppp_generic20388 0 (unused) slhc4784 0 [ppp_generic isdn] Any and all help is greatly appreciated. -- Best regards Peer Oliver Schmidt the internet company ___ 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
[Asterisk-Users] Reload cause Sound Volumn becomes very loud
Hi all, I am running the Asterisk with CVS-HEAD-10/25/04. When I type reload in console, whatever the incoming/outgoing sound volumn becomes very loud until I stop the asterisk and restart it. It's running no problem before I've upgrade the asterisk. Is there any configuration I need to modify? Since I'm not able to find any information in this list nor voip-info.org (maybe I've overlook, if so please point me to the correct URL..) Thanks! Regards, R Wong ___ 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
Re: [Asterisk-Users] Re: doublehash patch for 1.0.1
I don't know but it's IMHO, this should be just the opposite. Single # for a transfer and double ## to send the key on as DTMF. How many objects in a dialplan start with a #? Lyle - Original Message - From: Randy Bush [EMAIL PROTECTED] To: Barton Hodges [EMAIL PROTECTED] Cc: splatters [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 11:37 AM Subject: [Asterisk-Users] Re: doublehash patch for 1.0.1 Just tried the patch you made with the latest CVS and it patches fine although it does not work. Now when I hit # it does not send the DTMF to the other side at all. Although hitting ## does get the transfer. Now # doesn't do ANYTHING :) I'm not sure why that is, it works with all our phones (Grandstream BT101s and analog phones on Grandstream ATA286s). I just tested by calling my bank's IVR. applied patch. went great. now single # does not transfer and double does. but, i am having the same problem as matthew, the # does not go through at dmtf. all other keys go through as dmtf, just not the #. this is on a spa3k. clearly * is receiving the #, as ## does do a transfer. so why is a single # not being sent onward as dtmf? randy --- ps. and i have a general wonder/question about this. is someone who uses a commercial pbx, say a meridian or whatever, unable to use ivr systems because # is not sent? ___ 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 ___ 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
RE: [Asterisk-Users] Re: doublehash patch for 1.0.1
What if you call an external system and get a voicemail. Press # to finish your message . you would have to press ##. IMHO I think most users are not sophisticated enough to transfer calls. If they are they can press ##. Or am I missing something? :) S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Lyle Giese Sent: Sunday, October 24, 2004 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: doublehash patch for 1.0.1 I don't know but it's IMHO, this should be just the opposite. Single # for a transfer and double ## to send the key on as DTMF. How many objects in a dialplan start with a #? Lyle - Original Message - From: Randy Bush [EMAIL PROTECTED] To: Barton Hodges [EMAIL PROTECTED] Cc: splatters [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 11:37 AM Subject: [Asterisk-Users] Re: doublehash patch for 1.0.1 Just tried the patch you made with the latest CVS and it patches fine although it does not work. Now when I hit # it does not send the DTMF to the other side at all. Although hitting ## does get the transfer. Now # doesn't do ANYTHING :) I'm not sure why that is, it works with all our phones (Grandstream BT101s and analog phones on Grandstream ATA286s). I just tested by calling my bank's IVR. applied patch. went great. now single # does not transfer and double does. but, i am having the same problem as matthew, the # does not go through at dmtf. all other keys go through as dmtf, just not the #. this is on a spa3k. clearly * is receiving the #, as ## does do a transfer. so why is a single # not being sent onward as dtmf? randy --- ps. and i have a general wonder/question about this. is someone who uses a commercial pbx, say a meridian or whatever, unable to use ivr systems because # is not sent? ___ 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 ___ 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 ___ 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
Re: [Asterisk-Users] Re: doublehash patch for 1.0.1
I do some script type programing and have seen this in other uses. IMHO, it would be easier to program this way. Single # go to transfer function. Get # as first character in transfer, send out the DTMF tones instead and drop the request to transfer. I could be all wet on this, but my feeble mind sezs this makes sense from a programming perspective. Lyle - Original Message - From: Storm D. J. Petersen [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 3:04 PM Subject: RE: [Asterisk-Users] Re: doublehash patch for 1.0.1 What if you call an external system and get a voicemail. Press # to finish your message . you would have to press ##. IMHO I think most users are not sophisticated enough to transfer calls. If they are they can press ##. Or am I missing something? :) S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Lyle Giese Sent: Sunday, October 24, 2004 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: doublehash patch for 1.0.1 I don't know but it's IMHO, this should be just the opposite. Single # for a transfer and double ## to send the key on as DTMF. How many objects in a dialplan start with a #? Lyle - Original Message - From: Randy Bush [EMAIL PROTECTED] To: Barton Hodges [EMAIL PROTECTED] Cc: splatters [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 11:37 AM Subject: [Asterisk-Users] Re: doublehash patch for 1.0.1 Just tried the patch you made with the latest CVS and it patches fine although it does not work. Now when I hit # it does not send the DTMF to the other side at all. Although hitting ## does get the transfer. Now # doesn't do ANYTHING :) I'm not sure why that is, it works with all our phones (Grandstream BT101s and analog phones on Grandstream ATA286s). I just tested by calling my bank's IVR. applied patch. went great. now single # does not transfer and double does. but, i am having the same problem as matthew, the # does not go through at dmtf. all other keys go through as dmtf, just not the #. this is on a spa3k. clearly * is receiving the #, as ## does do a transfer. so why is a single # not being sent onward as dtmf? randy --- ps. and i have a general wonder/question about this. is someone who uses a commercial pbx, say a meridian or whatever, unable to use ivr systems because # is not sent? ___ 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 ___ 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 ___ 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 ___ 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
RE: [Asterisk-Users] Re: doublehash patch for 1.0.1
Personally I think like you... but I have to force myself to consider the dim wits that use my PBX. :) They are fat old men who barely understand what a telephone is... let alone VOIP. :) S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Lyle Giese Sent: Sunday, October 24, 2004 1:11 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: doublehash patch for 1.0.1 I do some script type programing and have seen this in other uses. IMHO, it would be easier to program this way. Single # go to transfer function. Get # as first character in transfer, send out the DTMF tones instead and drop the request to transfer. I could be all wet on this, but my feeble mind sezs this makes sense from a programming perspective. Lyle - Original Message - From: Storm D. J. Petersen [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 3:04 PM Subject: RE: [Asterisk-Users] Re: doublehash patch for 1.0.1 What if you call an external system and get a voicemail. Press # to finish your message . you would have to press ##. IMHO I think most users are not sophisticated enough to transfer calls. If they are they can press ##. Or am I missing something? :) S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Lyle Giese Sent: Sunday, October 24, 2004 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: doublehash patch for 1.0.1 I don't know but it's IMHO, this should be just the opposite. Single # for a transfer and double ## to send the key on as DTMF. How many objects in a dialplan start with a #? Lyle - Original Message - From: Randy Bush [EMAIL PROTECTED] To: Barton Hodges [EMAIL PROTECTED] Cc: splatters [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 11:37 AM Subject: [Asterisk-Users] Re: doublehash patch for 1.0.1 Just tried the patch you made with the latest CVS and it patches fine although it does not work. Now when I hit # it does not send the DTMF to the other side at all. Although hitting ## does get the transfer. Now # doesn't do ANYTHING :) I'm not sure why that is, it works with all our phones (Grandstream BT101s and analog phones on Grandstream ATA286s). I just tested by calling my bank's IVR. applied patch. went great. now single # does not transfer and double does. but, i am having the same problem as matthew, the # does not go through at dmtf. all other keys go through as dmtf, just not the #. this is on a spa3k. clearly * is receiving the #, as ## does do a transfer. so why is a single # not being sent onward as dtmf? randy --- ps. and i have a general wonder/question about this. is someone who uses a commercial pbx, say a meridian or whatever, unable to use ivr systems because # is not sent? ___ 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 ___ 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 ___ 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 ___ 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 ___ 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
[Asterisk-Users] Iaxy authentication
Hello, working on trying to get the Iaxy setup from behind a NAT. I have done everything the way I think it should be done but I can't seem to get dial tone and each time the device trys to register with * I get this message on the console: *** Oct 24 15:15:11 NOTICE[131080]: chan_iax2.c:3865 register_verify: No registration for peer '100' (from MyIPwasHere) *** Each time I actually pickup the phone connected to the Iaxy I get this: ** Oct 24 15:16:55 NOTICE[131080]: chan_iax2.c:5390 socket_read: Rejected connect attempt from 67.166.254.124 ** Seems that the authentication is not working.. I have set the user: and pass: fields in the iaxy_ext.conf which is fed into the iaxy and set the accountcode: in iax.conf to be the same as the user: field in the file that is fed into the Iaxy, is that correct or does the username go somewhere else. I have set the secret: the same as the password: field in the file that's fed into the iaxy. What have I missed here, I have dug and looked in the Wiki, Google, Digium site, list archives and there does not appear to be all that much documentation that I can find on setting this device up. Any help would be appreciated.. Thanks, Todd -- Start Your Own ISP! http://www.YourOwnISP.com ___ 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
RE: [Asterisk-Users] KSS/BLF on Asterisk
Get a hint! :-) Check out the hint priority in extensions.conf. There are also some details in the wiki. I've looked all over the wiki, and all the documentation I could get my hands on, Where did you find anything about the hint priority? I am interested in trying to make this work. ___ 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 ___ 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
Re: [Asterisk-Users] Re: Grandstream Flashing (different issue)
Ok, all of this makes sense but I guess the bigger question is.. How does one check their voice mail and delete it by using a phone and dialing into *? Is there a magic extension or series of buttons to push to get someone into their mailbox? Thanks, Todd -- Start Your Own ISP! http://www.YourOwnISP.com - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 7:22 AM Subject: Re: [Asterisk-Users] Re: Grandstream Flashing (different issue) May be risky if your email is screwy but it solves your problem Add: delete=yes in your voicemail.conf. - Original Message - From: Stephen R. Besch [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, October 22, 2004 11:55 AM Subject: [Asterisk-Users] Re: Grandstream Flashing (different issue) Todd Routhier - Lightwave Technologies, LLC. wrote: OK, this is a different flashing issue than the one that's being talked about. I have a few of these phones (GrandStream 101) and when a voicemail is received the light on the LED starts blinking and the dial tine stutters, this is cool. BUT. How the do I get it to stop, I have had mine covered with paper for the last 2 days because the blinking LED panel is driving me nuts. I have received the messages in my email and looked at them on the web. I am thinking that I have to check them by calling into the Asterisk system and mark them as read or something in order for this to quite. Two problems, I don't know how to check them by phone just yet and I will likely never check them by phone. You absolutely must get the message file deleted from the mail store. It doesn't matter how you delete the message either. You can use the * phone interface or simply delete the files associated with the message. As soon as the files are gone from the INBOX, * sends a SIP command to the phone to turn off the MWI. As long as there are any active messages in the INBOX, the light stays on. If you would like to be a guinea pig for my VB program that allows you to manage your mail folders and messages from a Windows GUI, I'll send you a copy. The only caveat is that I haven't yet found a way to get perfect synchronization with file access to the mail store. The user needs to be careful not to modify the INBOX while * is taking a new message. For me this is usually not a problem, since I would never be not answering the phone when I am listening to/moving/deleting messages - as a result, * would never be writing a new message when I was using the VB program. The program requires the VB6 runtime (available from MS for free) and that you have SAMBA running on the * server. Sincerely, Stephen R. Besch ___ 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 ___ 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 ___ 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
[Asterisk-Users] Unknown RTP codec 72 received
19 Question + this one and no answer; Does anyone have a clue what causes Unknown RTP codec 72 received notice and how to fix it? Regards Danny ___ 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
Re: [Asterisk-Users] Re: [Asterisk-Dev] New Channel Driver: chan_bluetooth
On Sun, 2004-10-24 at 20:08, Benjamin on Asterisk Mailing Lists wrote: On Wed, 20 Oct 2004 13:37:28 +0100, Theo Zourzouvillys [EMAIL PROTECTED] wrote: after a couple of days work banging my head against the wall (bloody standards my arse), i've got chan_bluetooth to a point where it's starting to function - certianly more than just proof of concept now. Will this need a mobile phone with bluetooth support or can you use a mere bluetooth headset as a client, if so how would you pick up and hang up or dial? It supports both. Regarding the headset, i have not seen how that works yet, but i would say you would need to enter the number somewhere (maybe special prefix on any phone + phoneno. to get the call to the headset) For the cellphone, this is really grand stuff: I originially was hoping for using the cellphone as a handset to the Asterisk box, but it's quite the oposite. Theo has basically obsoleted the expensive PSTN to GSM converters. Asterisk connects via bluetooth to the phone and uses it as a FXO line. Neat stuff. I'm seeing forward to, how this work progresses, because it's really promissing. Kind regards, Martin List-Petersen ___ 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
Re: [Asterisk-Users] Re: [Asterisk-Dev] New Channel Driver: chan_bluetooth
On Sun, 24 Oct 2004 21:42:49 +0100, Martin List-Petersen [EMAIL PROTECTED] wrote: Regarding the headset, i have not seen how that works yet, but i would say you would need to enter the number somewhere (maybe special prefix on any phone + phoneno. to get the call to the headset) Oh well, I guess I could use the manager API to initiate the call from my Powerbook. Cool! So, where do I get a Bluetooth headset now? I don't think they sell those over here either. Asterisk connects via bluetooth to the phone and uses it as a FXO line. Neat stuff. Too bad. Japan and Korea are the only two countries on the planet without GSM. Even Afghanistan has got GSM now. We don't even get a proper Bluetooth keitai. Frustrating. 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. ___ 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
RE: [Asterisk-Users] G.729 . . . I SMELL SMOKE!
On Sat, 2004-10-23 at 01:06 -0400, Jim Van Meggelen wrote: Few will disagree that the careful application of netiquette will be a benefit to any newsgroup/mailing list/board; and top posting is something that should be used sparingly. Nevertheless, top posting is not the horrid crime some might have us believe. When used appropriately, it serves very well, and only causes offense to the ideologues. Me too-type top posing is usually of no benefit, but when someone is commenting on a tangled and involved thread, it can make sense to frame the entirety of the thread in a thoughtful top post. Don't forget the same people who refuse to trim the bottom of the post and we end up with 20(your case only 1) copy of the mailing list footer. Then we get to the most dangerous beast, the abusive, expert troll. This is someone who clearly is very intelligent and articulate, and could argue their value due to a) their willingness to contribute, b) their level of knowledge and c) their fantastic writing skills. Unfortunately, these folks reduce their value to almost nothing by virtue of their pathetic lack of any manners whatsoever. They will drive away more people than they help -- and that doesn't bother them in the slightest. What a waste of talent. As I am sure to be painted by the above brush, let me offer just a small point here. I have had just a bit of time to think this over after politely listenening to the same argument from another person this weekend. You seem to not realize that those who are knowlegable are only so due to the vast amount of time we put into learning. I'm sure there are many people who are like me and are trying to spend a lot of time learning several projects that have no overlap. While we seek all this knowlege, I hope the others like me actually try and do things outside of the computer world as well. Now I want you to realize that many of the really newbie or lazy (these are NOT equal in the level I detest) questions that are answerable with a quick browse of the wiki or a simple google search end up being equivalent to SPAM in my mailbox as I try and search for information that furthers my knowlege. Understand that I learn from looking at what others are doing, and answers to others questions. So when you try and run off those who know a fair amount but don't meet your manners requirement, I want you to think about why you feel newbie or lazy users should be of higher value than those with the knowlege? Why do you wish to preserve their participation at the detriment to those who have more answers than questions? Hoping that the person I talked to this weekend is actually reading. I don't consider myself any more important than anyone else in this list, but rather I like others, wish to defend this channel of information from descending below useful signal to noise ratio. We all understand that it is generally best to avoid feeding the trolls, but every now and then the townsfolk must grab shovels and pitchforks, and drive these beasts back into the caves from whence they came; where, one hopes, they will contemplate the value of a few simple manners, and perhaps even one day to attempt to give to the community without the needless rancour. -- Steven Critchfield [EMAIL PROTECTED] ___ 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
Re: [Asterisk-Users] Re: Grandstream Flashing (different issue)
better get to reading. Basically you need to create an extension that points to voicemailmain. - Original Message - From: Me [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 4:30 PM Subject: Re: [Asterisk-Users] Re: Grandstream Flashing (different issue) Ok, all of this makes sense but I guess the bigger question is.. How does one check their voice mail and delete it by using a phone and dialing into *? Is there a magic extension or series of buttons to push to get someone into their mailbox? Thanks, Todd -- Start Your Own ISP! http://www.YourOwnISP.com - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 7:22 AM Subject: Re: [Asterisk-Users] Re: Grandstream Flashing (different issue) May be risky if your email is screwy but it solves your problem Add: delete=yes in your voicemail.conf. - Original Message - From: Stephen R. Besch [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, October 22, 2004 11:55 AM Subject: [Asterisk-Users] Re: Grandstream Flashing (different issue) Todd Routhier - Lightwave Technologies, LLC. wrote: OK, this is a different flashing issue than the one that's being talked about. I have a few of these phones (GrandStream 101) and when a voicemail is received the light on the LED starts blinking and the dial tine stutters, this is cool. BUT. How the do I get it to stop, I have had mine covered with paper for the last 2 days because the blinking LED panel is driving me nuts. I have received the messages in my email and looked at them on the web. I am thinking that I have to check them by calling into the Asterisk system and mark them as read or something in order for this to quite. Two problems, I don't know how to check them by phone just yet and I will likely never check them by phone. You absolutely must get the message file deleted from the mail store. It doesn't matter how you delete the message either. You can use the * phone interface or simply delete the files associated with the message. As soon as the files are gone from the INBOX, * sends a SIP command to the phone to turn off the MWI. As long as there are any active messages in the INBOX, the light stays on. If you would like to be a guinea pig for my VB program that allows you to manage your mail folders and messages from a Windows GUI, I'll send you a copy. The only caveat is that I haven't yet found a way to get perfect synchronization with file access to the mail store. The user needs to be careful not to modify the INBOX while * is taking a new message. For me this is usually not a problem, since I would never be not answering the phone when I am listening to/moving/deleting messages - as a result, * would never be writing a new message when I was using the VB program. The program requires the VB6 runtime (available from MS for free) and that you have SAMBA running on the * server. Sincerely, Stephen R. Besch ___ 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 ___ 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 ___ 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 ___ 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
RE: [Asterisk-Users] Unknown RTP codec 72 received
-Original Message- From: Danny Froberg [mailto:[EMAIL PROTECTED] Sent: Sunday, October 24, 2004 4:37 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Unknown RTP codec 72 received 19 Question + this one and no answer; Does anyone have a clue what causes Unknown RTP codec 72 received notice and how to fix it? Regards Danny I receive this message when I call from X-Lite. I notice that it is usually when I am sending DTMF digits. I could be using the wrong dtmfmode (using info), but I am not sure. This message is rather annoying so I would definitely like to see if anyone else has gotten it figured out. Robert Jackson ___ 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
Re: [Asterisk-Users] Re: Grandstream Flashing (different issue)
http://www.voip-info.org/tiki-searchresults.php?words=voicemailwhere=pages - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 5:27 PM Subject: Re: [Asterisk-Users] Re: Grandstream Flashing (different issue) better get to reading. Basically you need to create an extension that points to voicemailmain. - Original Message - From: Me [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 4:30 PM Subject: Re: [Asterisk-Users] Re: Grandstream Flashing (different issue) Ok, all of this makes sense but I guess the bigger question is.. How does one check their voice mail and delete it by using a phone and dialing into *? Is there a magic extension or series of buttons to push to get someone into their mailbox? Thanks, Todd -- Start Your Own ISP! http://www.YourOwnISP.com - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 7:22 AM Subject: Re: [Asterisk-Users] Re: Grandstream Flashing (different issue) May be risky if your email is screwy but it solves your problem Add: delete=yes in your voicemail.conf. - Original Message - From: Stephen R. Besch [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, October 22, 2004 11:55 AM Subject: [Asterisk-Users] Re: Grandstream Flashing (different issue) Todd Routhier - Lightwave Technologies, LLC. wrote: OK, this is a different flashing issue than the one that's being talked about. I have a few of these phones (GrandStream 101) and when a voicemail is received the light on the LED starts blinking and the dial tine stutters, this is cool. BUT. How the do I get it to stop, I have had mine covered with paper for the last 2 days because the blinking LED panel is driving me nuts. I have received the messages in my email and looked at them on the web. I am thinking that I have to check them by calling into the Asterisk system and mark them as read or something in order for this to quite. Two problems, I don't know how to check them by phone just yet and I will likely never check them by phone. You absolutely must get the message file deleted from the mail store. It doesn't matter how you delete the message either. You can use the * phone interface or simply delete the files associated with the message. As soon as the files are gone from the INBOX, * sends a SIP command to the phone to turn off the MWI. As long as there are any active messages in the INBOX, the light stays on. If you would like to be a guinea pig for my VB program that allows you to manage your mail folders and messages from a Windows GUI, I'll send you a copy. The only caveat is that I haven't yet found a way to get perfect synchronization with file access to the mail store. The user needs to be careful not to modify the INBOX while * is taking a new message. For me this is usually not a problem, since I would never be not answering the phone when I am listening to/moving/deleting messages - as a result, * would never be writing a new message when I was using the VB program. The program requires the VB6 runtime (available from MS for free) and that you have SAMBA running on the * server. Sincerely, Stephen R. Besch ___ 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 ___ 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 ___ 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 ___ 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
RE: [Asterisk-Users] Geotel integration with Asterisk
From what I read about a year ago was that it was a carrier hosted solution that actually controlled the ss7 switching at the exchange (basically no call costs from tromboning, and was only implemented into an ip-centrex or hosted call centre application. Are you saying that enterprises can buy something similar and control the carriers switching? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Smith Sent: Sunday, October 24, 2004 2:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Geotel integration with Asterisk Geotel is a company that Cisco bought which provides call control across geographically dispersed locations. The simplest application is being able to query call queue status at another location. For example, a call comes in and can be sent to one of three call center locations. Geotel can query each location to see who is the least busy for this type of call. Traditionally it has been VERY expensive. We provide some primitive Geotel functions in-the-cloud right now. For example, we can know how many live calls are going to a location before we send the call. We can set thresholds (e.g. if a location A has over 100 concurrent calls send them to location B). Geotel can theoretically provide this and carry it further. I think there is some nice enterprise reporting that can come from the Geotel as well. G. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Saturday, October 23, 2004 4:30 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Geotel integration with Asterisk Ok lets get this out of the way... WTF is Geotel? bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Greg Smith Sent: Saturday, October 23, 2004 4:19 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Geotel integration with Asterisk Has any one integrated to a Geotel with Asterisk? Thanks. Greg Advanta ___ 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 ___ 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 ___ 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
RE: [Asterisk-Users] Iaxy authentication
I bought out IAXy devices from NetXUSA, who also sent me this short installation document which I've copied below: Quick Start Guide for Digium IAXY Device Determine the IP address of each unit by viewing the logs on your DHCP Server to see which IP address your IAXY has taken, you cannot reserve IP address based on the MAC address. You may however define a static IP address after the initial setup. Login to Asterisks CVS and checkout iaxyprov, then do a make Once your in the iaxyprov directory edit the iaxy.sample.conf file or create a new configuration file Define the IP address or if you wish to continue to use DHCP, assign a unique user name for your device, password, codec, and the Asterisk Server it will register with. For an example see the iaxy.sample.conf file Once you have defined your iaxy.conf file proceed to run iaxyprov. The format is as follows: ./iaxyprov IP Address iaxy.conf File name You should receive a confirmation back that the data was received, Repeat these same steps for each IAXY device, changing only the user name and IP address if not using DHCP Next, you need to configure the Iax.conf file under /etc/asterisk. Create each user, see the sample configuration below [100] ;User 100 type=friend username=100 secret=1234 context=default disallow=all allow=ulaw mailbox=1234 Lastly, the format for the IAXY devices in extensions.conf is IAX2/User Exten = 100,1,Dial(IAX2/100) For further questions please contact NETXUSA Technical support Monday Friday 9 am est 6 pm est or refer to the Asterisk user list for help. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Me Sent: Sunday, October 24, 2004 3:24 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Iaxy authentication Hello, working on trying to get the Iaxy setup from behind a NAT. I have done everything the way I think it should be done but I can't seem to get dial tone and each time the device trys to register with * I get this message on the console: *** Oct 24 15:15:11 NOTICE[131080]: chan_iax2.c:3865 register_verify: No registration for peer '100' (from MyIPwasHere) *** Each time I actually pickup the phone connected to the Iaxy I get this: ** Oct 24 15:16:55 NOTICE[131080]: chan_iax2.c:5390 socket_read: Rejected connect attempt from 67.166.254.124 ** Seems that the authentication is not working.. I have set the user: and pass: fields in the iaxy_ext.conf which is fed into the iaxy and set the accountcode: in iax.conf to be the same as the user: field in the file that is fed into the Iaxy, is that correct or does the username go somewhere else. I have set the secret: the same as the password: field in the file that's fed into the iaxy. What have I missed here, I have dug and looked in the Wiki, Google, Digium site, list archives and there does not appear to be all that much documentation that I can find on setting this device up. Any help would be appreciated.. Thanks, Todd -- Start Your Own ISP! http://www.YourOwnISP.com ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.778 / Virus Database: 525 - Release Date: 10/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.778 / Virus Database: 525 - Release Date: 10/15/2004 ___ 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
Re: [Asterisk-Users] Re: Grandstream Flashing (different issue)
Thanks Steve, it's not that I have not been reading (ask my wife how many nights I have slept in the last week), and it's not that there is not a huge amount of info out there. The problem I am having is finding the info I need in any sort of organized way. The searches I do sometimes come up with solutions but otherwise I am left to browse the Wiki which has tons of info, it's just not very obvious how to find it most of the time. Thanks for the link, I am sure this will help! Todd -- Start Your Own ISP! http://www.YourOwnISP.com - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 4:37 PM Subject: Re: [Asterisk-Users] Re: Grandstream Flashing (different issue) http://www.voip-info.org/tiki-searchresults.php?words=voicemailwhere=pages - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 5:27 PM Subject: Re: [Asterisk-Users] Re: Grandstream Flashing (different issue) better get to reading. Basically you need to create an extension that points to voicemailmain. - Original Message - From: Me [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 4:30 PM Subject: Re: [Asterisk-Users] Re: Grandstream Flashing (different issue) Ok, all of this makes sense but I guess the bigger question is.. How does one check their voice mail and delete it by using a phone and dialing into *? Is there a magic extension or series of buttons to push to get someone into their mailbox? Thanks, Todd -- Start Your Own ISP! http://www.YourOwnISP.com - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 7:22 AM Subject: Re: [Asterisk-Users] Re: Grandstream Flashing (different issue) May be risky if your email is screwy but it solves your problem Add: delete=yes in your voicemail.conf. - Original Message - From: Stephen R. Besch [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, October 22, 2004 11:55 AM Subject: [Asterisk-Users] Re: Grandstream Flashing (different issue) Todd Routhier - Lightwave Technologies, LLC. wrote: OK, this is a different flashing issue than the one that's being talked about. I have a few of these phones (GrandStream 101) and when a voicemail is received the light on the LED starts blinking and the dial tine stutters, this is cool. BUT. How the do I get it to stop, I have had mine covered with paper for the last 2 days because the blinking LED panel is driving me nuts. I have received the messages in my email and looked at them on the web. I am thinking that I have to check them by calling into the Asterisk system and mark them as read or something in order for this to quite. Two problems, I don't know how to check them by phone just yet and I will likely never check them by phone. You absolutely must get the message file deleted from the mail store. It doesn't matter how you delete the message either. You can use the * phone interface or simply delete the files associated with the message. As soon as the files are gone from the INBOX, * sends a SIP command to the phone to turn off the MWI. As long as there are any active messages in the INBOX, the light stays on. If you would like to be a guinea pig for my VB program that allows you to manage your mail folders and messages from a Windows GUI, I'll send you a copy. The only caveat is that I haven't yet found a way to get perfect synchronization with file access to the mail store. The user needs to be careful not to modify the INBOX while * is taking a new message. For me this is usually not a problem, since I would never be not answering the phone when I am listening to/moving/deleting messages - as a result, * would never be writing a new message when I was using the VB program. The program requires the VB6 runtime (available from MS for free) and that you have SAMBA running on the * server. Sincerely, Stephen R. Besch ___ 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 ___ 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
RE: [Asterisk-Users] KSS/BLF on Asterisk
[EMAIL PROTECTED] wrote: Get a hint! :-) Check out the hint priority in extensions.conf. There are also some details in the wiki. I've looked all over the wiki, and all the documentation I could get my hands on, Where did you find anything about the hint priority? I am interested in trying to make this work. The only references I could find to it were here: http://www.voip-info.org/tiki-index.php?page=Asterisk%20standard%20exten sions here: http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20snom and here: http://www.mail-archive.com/[EMAIL PROTECTED]/msg49781.htm l Not much to work from, but the lack of documentation on this feature is probably signifigant. I realize that this is going to be something that requires more research on my part, as no one appears to be using it very much. ___ 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 ___ 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 ___ 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
Re: [Asterisk-Users] MusicOnHold() - how to restart player from the beginning on each call? (fwd)
What Manfred wants to do is not that uncommon. I've used the method that William has suggested in the past. On a lot of corporate phone systems this is a simple option in the programming. Another way is to simply advertise your specials over your music on hold and repeat them... Hoping that the person hears the special.. The only reason they wouldn't is if the customer was on hold for too small of an amount of time, which the customer will appreciate more. :) On Sun, 24 Oct 2004 14:26:50 -0400, William Suffill [EMAIL PROTECTED] wrote: Why not just create a context that plays static msgs whenever someone is transfered thereThank you for calling Monthly special etc ... then transfer them back when the person at the biz picks up On Sun, 24 Oct 2004 14:23:04 -0400, Emilio Panighetti [EMAIL PROTECTED] wrote: Looks like what you want is not music on-hold, but rather a streaming server On Oct 22, 2004, at 4:23 PM, Ryan Courtnage wrote: On Fri, 2004-22-10 at 16:05 -0400, Kanwar Ranbir Sandhu wrote: On Fri, 2004-10-22 at 05:56, Manfred Petz wrote: [snip] Is there a way to force MusicOnHold() to be restarted from the beginning for each call which has been answered? [snip] Why? What would be the point? off the top of my head ... promotional messages. Manfred - I don't think there is a graceful way to do this. I know that if you do a killall mpg123 at your command line, the next call MOH answers will start playing the mp3s at the beginning. Of course this would affect others that are listening, but if you build out some logic you might be able to make some use of it. Ryan ___ 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 ___ 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 ___ 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 -- MBM ___ 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
[Asterisk-Users] Connection to a H323 system
I found in Google a h323.conf file, but not on my Asterisk installation. Do I need to do more than h323.conf ??? I have a h323 phone and would like to replace it as one connection to my Asterisk, Thanks for your hints. bye Ronald begin:vcard fn:Ronald Wiplinger n:Wiplinger;Ronald email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE version:2.1 end:vcard ___ 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
[Asterisk-Users] ACT Gateways
Has anybody tested any gateways from ACT: http://www.act-tel.com.tw/Index2.htm They have four different configurations: 4xFXS - 4xFXO 2xFXS - 2xFXO 1xFXS - 1xFXO 4xFXS I emailed them but they didn't bother the respond. -- #Joseph ___ 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
Re: [Asterisk-Users] Connection to a H323 system
vi /usr/src/asterisk/channels/h323/h323.conf.sample vi /usr/src/asterisk/channels/h323/README - Original Message - From: Ronald Wiplinger [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 8:01 PM Subject: [Asterisk-Users] Connection to a H323 system I found in Google a h323.conf file, but not on my Asterisk installation. Do I need to do more than h323.conf ??? I have a h323 phone and would like to replace it as one connection to my Asterisk, Thanks for your hints. bye Ronald ___ 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 ___ 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
Re: [Asterisk-Users] iLBC/PCM16 Huge Cost
Okay, I have removed the IDE Controller and am now using onboard. The problems below still exist-- Trevor Peirce wrote: show translation still reveals the iLBC column in the 8700 to 9600 range though. LPC10's row is also in the 900s. show translation recalc 10 still causes the * console to stop responding as well. When I first bootup *, it consumes nearly 100% of CPU, all in user... system is less than 1%. The time that it monopolizes the processor varies as well. ___ 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
Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible
On Sun, 2004-10-24 at 10:24, Steve Totaro wrote: I know she works at Digium but they probably go down the street to a real sound stage to do the recordings via 3rd party. A sound stage is a facility used to create and process professional recordings. They can be used by anyone employed by an company. SNIP.. Take a look at the Astricon links, I believe that she is in Canada and works for several groups. Dave ___ 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
RE: [Asterisk-Users] Geotel integration with Asterisk
On Sun, 2004-10-24 at 17:52, dean collins wrote: From what I read about a year ago was that it was a carrier hosted solution that actually controlled the ss7 switching at the exchange (basically no call costs from tromboning, and was only implemented into an ip-centrex or hosted call centre application. Are you saying that enterprises can buy something similar and control the carriers switching? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Smith Sent: Sunday, October 24, 2004 2:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Geotel integration with Asterisk Geotel is a company that Cisco bought which provides call control across geographically dispersed locations. The simplest application is being able to query call queue status at another location. For example, a call comes in and can be sent to one of three call center locations. Geotel can query each location to see who is the least busy for this type of call. Traditionally it has been VERY expensive. We provide some primitive Geotel functions in-the-cloud right now. For example, we can know how many live calls are going to a location before we send the call. We can set thresholds (e.g. if a location A has over 100 concurrent calls send them to location B). Geotel can theoretically provide this and carry it further. I think there is some nice enterprise reporting that can come from the Geotel as well. G. SNIP... The GEOtel solution uses a type of interface that was originally designed for tie-in to the MCI network. The MCI network uses something call a DAP(data access point) the DAP performs a database lookup anytime that an 800,888,866,877 or virtual network number is dialed on their network. This lookup is done via SS7 and returns the appropriate routing information ie.. Switch and trunk group with appropriate DNIS to the originating switch which then routes the call to the proper termination location. The GEOtel solution actually works like a wedge into the call routing info. By using an adjunct processor that is in contact with the customers network switches/ACDs the DAP actually queries the adjunct processor for the proper routing data, and returns the appropriate info for call termination. The return data is based on whatever rules that the adjunct uses for the call lookup. The original trial for this service was used by MCI corporate for their own Customer service network Galaxy class ACD's made by Rockwell. The adjunct would poll the ACD's and determine queuing times as well as time of day number of operators etc, and return proper routing information. This was called Intelligent Routing Service (IRS) but the marketing group decided that Intelligent Call Routing was a better name. Hope this was informative in some way :) Dave ___ 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
RE: [Asterisk-Users] KSS/BLF on Asterisk
I am buying a Snom phone this week. I will play with this feature and see what I can get going. I will share my findings. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Van Meggelen Sent: Sunday, October 24, 2004 6:05 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] KSS/BLF on Asterisk [EMAIL PROTECTED] wrote: Get a hint! :-) Check out the hint priority in extensions.conf. There are also some details in the wiki. I've looked all over the wiki, and all the documentation I could get my hands on, Where did you find anything about the hint priority? I am interested in trying to make this work. The only references I could find to it were here: http://www.voip-info.org/tiki-index.php?page=Asterisk%20standard%20exten sions here: http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20snom and here: http://www.mail-archive.com/[EMAIL PROTECTED]/msg49781.htm l Not much to work from, but the lack of documentation on this feature is probably signifigant. I realize that this is going to be something that requires more research on my part, as no one appears to be using it very much. ___ 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 ___ 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 ___ 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 ___ 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
[Asterisk-Users] Howto get voicemail $VM_ vars into externnotify script?
Hi all, I am trying to slap together a script that will email2sms the details of the voicemails left on my * box to my gsm phone. I can't figure out how to get my script to pick up the voicemail vars like ${VM_MSGNUM}, ${VM_DATE}, ${VM_MAILBOX}, ${VM_CALLERID}, ${VM_DUR}. Right now I have this: #!/bin/sh /bin/mail -s Voicemail received with details: Msg nr : `echo $VM_MSGNUM` Msg date: `echo $VM_DATE` For : `echo $VM_MAILBOX` From: `echo $VM_CALLERID` Length : `echo $VM_DUR` [EMAIL PROTECTED] Which doesn't work as all the $VM_ vars show up blank. Any ideas? TIA, Patrick ___ 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
[Asterisk-Users] Several FXS Ports
hello list. looking for a way to have several FXS ports on an asterisk box, lets say oh... 300, just for shoots and giggles. would i need special telco equipment? if so, what kind? i already have a 23 inch cabinet, which i'm told telco equipment uses 23 inch. any insight on this would be greatly appreciated. Thanks in advance -- James W Dumais ABSS::Networks http://www.abss.ca/ 1(705)725-9124 / 1(800)473-2121 ___ 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
Re: [Asterisk-Users] Several FXS Ports
On Sun, 2004-10-24 at 22:02 -0400, James Dumais wrote: hello list. looking for a way to have several FXS ports on an asterisk box, lets say oh... 300, just for shoots and giggles. would i need special telco equipment? if so, what kind? i already have a 23 inch cabinet, which i'm told telco equipment uses 23 inch. any insight on this would be greatly appreciated. Do you realize this EXACT question came up as recent as this month on this EXACT list. http://lists.digium.com/pipermail/asterisk-users/2004-October/065518.html -- Steven Critchfield [EMAIL PROTECTED] ___ 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
Re: [Asterisk-Users] G.729 . . . I SMELL SMOKE!
- Original Message - From: Steven Critchfield [EMAIL PROTECTED] You seem to not realize that those who are knowlegable are only so due to the vast amount of time we put into learning. I'm sure there are many people who are like me and are trying to spend a lot of time learning several projects that have no overlap. While we seek all this knowlege, I hope the others like me actually try and do things outside of the computer world as well. I agree with you. I also spend signifigant time in learning things about Asterisk and other projects. Now I want you to realize that many of the really newbie or lazy (these are NOT equal in the level I detest) questions that are answerable with a quick browse of the wiki or a simple google search end up being equivalent to SPAM in my mailbox as I try and search for information that furthers my knowlege. Understand that I learn from looking at what others are doing, and answers to others questions. SPAM is in a way not just uneccesary solicitation but debate as well. The kind of debate that nobody wins. the kind where the participants agree to disagree. These postings take a toll on a list such as this. (and now i'm guilty of it too) But my point is that this list generates a huge amount of mail (and noise) I would prefer the former over the latter. So when you try and run off those who know a fair amount but don't meet your manners requirement, I want you to think about why you feel newbie or lazy users should be of higher value than those with the knowlege? Why do you wish to preserve their participation at the detriment to those who have more answers than questions? I think that everyone who has participated in or used Asterisk can help someone else or learn from someone else. It doesn't make the helper smarter than the one asking the question. It allows someone who has been down a particular path to help guide someone down that same path. It's really all about community. Hoping that the person I talked to this weekend is actually reading. I don't consider myself any more important than anyone else in this list, but rather I like others, wish to defend this channel of information from descending below useful signal to noise ratio. Again I agree. We are a community and a community is made up of differing views, opinions, and beliefs. It's through a common interest we are all together and therefore we must insure that the very things that bring us together, that is the quest for knowledge about Asterisk and helping others with our experiences with Asterisk are preserved. We all understand that it is generally best to avoid feeding the trolls, but every now and then the townsfolk must grab shovels and pitchforks, and drive these beasts back into the caves from whence they came; where, one hopes, they will contemplate the value of a few simple manners, and perhaps even one day to attempt to give to the community without the needless rancour. I wish to apologize to Kevin Walsh. While I did not write the above, or agree with Kevin, It was my sentiment. at the time. Kevin I know you are an interested member of this community and you have contributed your knowledge in a meaningful way in the past. I hope you can forgive my rudeness. Best Regards, Mike Boger Jr. -- Steven Critchfield [EMAIL PROTECTED] ___ 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 ___ 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