[Asterisk-Users] problems detecting hangup events
Im having problems with asterisk detecting when a calling party through a PSTN line has hung up. It takes 10 sec for it to finally detect. Im revieving my service through telus residential line. i have a SPA-3000 and a wildcard fxo, both behave identical. Ive checked voltages, everething seems correct. is this a problem with my setup? is this a telus problem? any suggestions would be appreciated as this is causing havoc with voicemail.. all extensions are recieving blank voice mail messages because of this. thanks.. Vince ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] I want to load chan_h323.so
I use Fedora core 2, and openssl-0.9.7, expat-1.95.7 is installed by rpm packages. I downloaded pwlib-1.5.2 and openh323-1.12.2 at my home directory(/root/root_src), asterisk 1.0.4 at directory /usr/src/ and have installed successfully. Asterisk is executed normally, but module chan_h323.so cannot be loaded. The message is : # asterisk ?vvvgc . .some message . Asterisk Ready. *CLI load chan_h323.so /root/root_src/openh323/lib/libh323_linux_x86_r.so.1.12.2: undefined symbol: _Z13vpb_dial_synciPc Unable to load module chan_h323.so *CLI Please give me your solutions. Thank you for your reading. My install log is : # tar xvfz pwlib-1.5.2.tar.gz # tar xvfz openh323-1.12.2.tar.gz # cd /root/root_src/pwlib # ./configure # make # cd /root/root_src/openh323 # ./configure # make opt # cd /usr/src # export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot # cvs login # cvs co -r v1-0 asterisk # echo $PWLIBDIR /root/root_src/pwlib # echo $OPENH323DIR /root/root_src/openh323 # echo $LD_LIBRARY_PATH /root/root_src/pwlib/lib:/root/root_src/openh323/lib # cd /usr/src/asterisk/channels/h323 # make # cd /usr/src/asterisk # make install " , Daum" http://www.daum.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: Open source CRM systems with * integration
Has anyone any experience of the above. Key feature for me is tracking incoming and outgoing emails and linking them to the contact record. Thanks, sorry for the OT ;-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Caller IP-Addr from agi ?
Hi Anyone have a hint how to get callers IP-Address from a php-agi script ? /HH ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Codec Issue on IAX trunk? (Solved)
On Sat, Feb 12, 2005 at 10:44:11AM -0600, Rich Adamson wrote: I haven't tried to keep track of the code changes involving reloads (or cli restarts for that matter), but having been around * for a fair amount of time and having been caught with making changes that have had no affect, I'll usually play it very safe and simply stop / start asterisk for many different changes. Iax and sip def's in particular. Reloads are fine for lots of things, but experience is the only way to know what's acceptable at this point. I've noticed this myself. However, I have been able to acheive a similar effect by unloading and then reloading the module. In my case I was testing H323, it might be trickier if you're actually using what you're playing with... Hope this helps, -- Martijn van Oosterhout ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sangoma A102 cards testing
Does anyone have any experience ith configureing the sangoma A102 card for testing using a e1 cross cable i've configured and installed the cards properly even the lights on the card are green which proves that my cross cable is properly built too. my problem is with asterisk which gives me these errors PRI got event: HDLC Abort (6)on Primary D-channel of span 1 PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 No D-channels available! Using Primary on channel anyways 47! PRI: !! Not good - head of queue has not been transmitted yet I've tried everything i can think off with the wancfg configuration files here is my zaptel and zapata configs. span=1,0,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 span=2,1,0,ccs,hdb3 bchan=32-46 dchan=47 bchan=48-62 -- zapata.conf switchtype=euroisdn signalling=pri_net group=1 channel=1-15 channel=17-31 group=2 signalling=pri_cpe channel=32-46 channel=48-62 --- do i need to fool around with some jumpers on the card or something to activate internal clock on the card. zttol says INTERNALLY CLOCKED for both the ports. There are NO Alarms and no missed IRQ's I'm using asterisk 1.0.5 on debian with 2.4.29 kernel -- regards Vikram (http://www.vicramresearch.com) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Speech Recognition
Hi Dean, What relevance has that to what we were discussing? We were talking about free form speech to text. That is a world apart from a voice activated IVR. Besides that, I have never found a voice activated IVR in English that gets better than about 30% accuracy on a fairly limited decision. A slight divergence from the typical 98% they claim. In contrast, I have seen very good accuracy for Cantonese and Mandarin, which have been less intensively developed. Regards, Steve dean collins wrote: Disagree with you Matt. Check out www.angel.com If anyone wants some contacts over there email me. I'm sure they would be happy to set up on API for utilizing their services in conjunction with asterisk. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Klein Sent: Saturday, February 12, 2005 11:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Speech Recognition Agreed, Steve. Iq, Maybe it is for your voice, but speech to text is a long ways away from being as advanced as you think it is. Check out dragon speek, and see what it takes to train a voice... -m On Sun, 13 Feb 2005, Steve Underwood wrote: Iqbal wrote: Hi I dont know jack about speech recognition, however since this topic came up anyonw know how spinvox do speech ercognition, in fact its so good it converst the speech to text and sends the voicemail as a SMS, I think a awesome addone to the sms module in asterisk. If it works really well, there is probably a human operator involved. A number of systems that try to look automated actually rely on human operators. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma A102 cards testing
On Sun, February 13, 2005 23:01, Vikram Rangnekar said: span=1,0,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 span=2,1,0,ccs,hdb3 bchan=32-46 dchan=47 bchan=48-62 At a guess 16,32,48 and 64 are d channels, where as you are telling it to use b channels for d channels... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the long run the pessimist may be proved right, but the optimist has a better time on the trip. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 7912G: Takes the same firmware as 7940/60?
Here it is: http://www.cisco.com/cgi-bin/tablebuild.pl/ip-phone-7905 software is the same for 7905 / 7912 De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Marty Mastera Envoy: samedi 12 fvrier 2005 18:47 : Asterisk Users Mailing List - Non-Commercial Discussion Objet: [Asterisk-Users] 7912G: Takes the same firmware as 7940/60? Does anyone know if the 7912G (which the wiki says can do either sccp or sip) uses the 7940/60 sip firmware? I ask this because the only firmware I can seem to find on TAC for the 7912G is sccp, no sip...if it takes it's own firmware and doesn't use 7940/60 firmware, can someone point me to the right location for it? Thanks, Marty Mastera M3 Resources [EMAIL PROTECTED] Phone: 303.680.1283 x200 FAX: 206.666.1786 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Sangoma A102 cards testing
+++ Duane [13/02/05 22:56 +1100]: On Sun, February 13, 2005 23:01, Vikram Rangnekar said: span=1,0,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 span=2,1,0,ccs,hdb3 bchan=32-46 dchan=47 bchan=48-62 At a guess 16,32,48 and 64 are d channels, where as you are telling it to use b channels for d channels... -- Best regards, Duane I'm sorry i didnt quite understand what you meant why would i need 4 d-channels i've only used 16 and 47 as my dchannels and want span 1 to generate the clock for this e1 setup. -- regards Vikram (http://www.vicramresearch.com) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Sangoma A102 cards testing
On Sun, February 13, 2005 23:19, Vikram Rangnekar said: I'm sorry i didnt quite understand what you meant why would i need 4 d-channels i've only used 16 and 47 as my dchannels and want span 1 to generate the clock for this e1 setup. As far as I'm aware each E1 has 30 b channels, and 2 d channels... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the long run the pessimist may be proved right, but the optimist has a better time on the trip. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Sangoma A102 cards testing
Duane wrote: On Sun, February 13, 2005 23:19, Vikram Rangnekar said: I'm sorry i didnt quite understand what you meant why would i need 4 d-channels i've only used 16 and 47 as my dchannels and want span 1 to generate the clock for this e1 setup. As far as I'm aware each E1 has 30 b channels, and 2 d channels... Wrong. An E1 used for ISDN has 30 B channels, and 1 D channel. 1-16 and 17-31 are Bs, 16 is the D. Channel 0 is used by the framers for synchronisation, and is not accessible as a channel by the user. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: bad sound ISDN bristuff
Sjaak Nabuurs wrote: Hello * users I've problems with sound quality on zaphfc Asterisk works fine good sound quality. If I do make load in the bristuf.xx zaphfc dir then sound quality drops directly. Even if I don't load the chan_zap in the modules.conf I use this config on more (even old 400Mhz machines) and works correctly. Looks like an hardware problem but I can't find it. I don't see any conflics on IRQ or interupts Using : Asterisk 1.0.1-BRIstuffed-0.2.0-RC1 ISDN HFC cologne cards If you have any solutions I would like to hear it. Use Florz patch for the beginnig. Corvin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mobile Wireless IP Phone
On 14:46, Sat 12 Feb 05, eric m wrote: Hi! I would like to have feedback on wireless (wifi / 802.11b) IP phone to use with Asterisk PBX. Can you sugest model, The best and also the worst to use. Thanks, eric. Hi, I read on sf that the cisco wireless phone is almost 100% working with chan_sccp. When the sound is as good as the 7905, you will have a great device. -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Open source CRM systems with * integration
On 10:56, Sun 13 Feb 05, John Middleton wrote: Has anyone any experience of the above. Key feature for me is tracking incoming and outgoing emails and linking them to the contact record. Thanks, sorry for the OT ;-) Hi, Have a look at http://www2.covide.net Maybe that's what you want. The project page is at http://sourceforge.net/projects/covide Have fun. -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi or chan_mISDN vs bristuff
On 21:21, Sat 12 Feb 05, Robert Rozman wrote: Hi, could you give some more info about your setup. How do you get 2 fritz cards working (I thought it works only on 2.4 kernels ) ? What capi drivers do you use ? Thanks, regards, Rob. Hi, I followed the instructions in the wiki to alter the module source. I dont know if this works for 2.6, as we use 2.4 I know there is a check in the source for 2.5 It's worth a try right ? As for the versions. I use the Debian packages in Debian Sid It is asterisk-1.0.5-2 asterisk-chan-capi 0.3.5-9 -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Speech Recognition
Steve then you have had your head up your arse for a number of years. Nuance was delivering 90% in 1999 and I have a number of happy customers to prove it. You also obviously didn't look at either the Nuance or angel sites because both of them offer free form speech to text capabilities. One of the first customers I had in Australia for Nuance was ordering of stock for Revlon cosmetics using a speech to an automated ordering system using their antiquated stock database. Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: Sunday, February 13, 2005 6:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Speech Recognition Hi Dean, What relevance has that to what we were discussing? We were talking about free form speech to text. That is a world apart from a voice activated IVR. Besides that, I have never found a voice activated IVR in English that gets better than about 30% accuracy on a fairly limited decision. A slight divergence from the typical 98% they claim. In contrast, I have seen very good accuracy for Cantonese and Mandarin, which have been less intensively developed. Regards, Steve dean collins wrote: Disagree with you Matt. Check out www.angel.com If anyone wants some contacts over there email me. I'm sure they would be happy to set up on API for utilizing their services in conjunction with asterisk. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Klein Sent: Saturday, February 12, 2005 11:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Speech Recognition Agreed, Steve. Iq, Maybe it is for your voice, but speech to text is a long ways away from being as advanced as you think it is. Check out dragon speek, and see what it takes to train a voice... -m On Sun, 13 Feb 2005, Steve Underwood wrote: Iqbal wrote: Hi I dont know jack about speech recognition, however since this topic came up anyonw know how spinvox do speech ercognition, in fact its so good it converst the speech to text and sends the voicemail as a SMS, I think a awesome addone to the sms module in asterisk. If it works really well, there is probably a human operator involved. A number of systems that try to look automated actually rely on human operators. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] soho fax suggestions?
Steve, Need to replace our older soho fax machine with something more current. Would like to run the fax line through *, but haven't been able to make spandsp work correctly with digium TDM04b card. Our fax volume This seems to be a problem with the current wctdm driver. It seems to be broken for audio going out. I used to be able to send faxes reliably using spandsp and a TDM40P card, but I no longer can. I haven't had time to look in detail at what is wrong. I'd love to get this working for receive only. We probably get something in the neighborhood of 90% junk/spam faxes each week, and being able to view them online (and forward to the appropriate office) would be very helpful. If spandsp doesn't work now, spandsp won't work through a T.38 channel. I was afraid of that. It seems the T.38 in a number of units doesn't really work. I'm not clear how widespread that problem is, but since there are only a few suppliers of protocol stacks for these boxes I suspect it may be widespread. Okay, then it would appear my best choice for the short term is to try to get spandsp working again (for incoming fax). I tried somewhere around spandsp-pre4, but got totally lost with the unfamiliar debug messages and my inability to translate those messages into corrective action steps. (I can arrange other methods for transmitting a fax.) Is there any kind of reference document available from anywhere that would help me understand the spandsp debug messages? More then willing to give it another try with current cvs head (patching manually is not a problem). Rich ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Open source CRM systems with * integration
John Middleton wrote: Has anyone any experience of the above. Key feature for me is tracking incoming and outgoing emails and linking them to the contact record. Thanks, sorry for the OT ;-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users www.compiere.org www.bestpractical.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ast_data does not patch
Lonnie, If you look at: http://www.voip-info.org/wiki-Asterisk+RealTime it says that MySQL _is_ supported. I don't know whether RealTime PostgreSQL, but I can't upgrade to RealTime anyway ... I need a stable version of asterisk, and the current stable version does not include RealTime. :( I am hoping to be able to use MySQL as well ... I've got all the config files set up correctly, I believe, but I get nowhere because: a) ast_data seems to be currently broken b) the shipped version of asterisk app_voicemail.c seems to have a bug in its select statement ... MySQL's log shows that it's connecting correctly to the database, but then it issues a command SEL instead of the command SELECT * from users WHERE ... Very frustrating. I don't really want to debug this code, but it looks like I'm going to give it a shot anyway. (No, not the ast_data code, the basic app_voicemail.c code.) Lonnie, if you have any luck getting ast_data to work, or find a contact address for rgagnon (who seems to have done the original development and presumably is still maintaining it), please forward that to me. Cheers, Maya --- [EMAIL PROTECTED] wrote: Thanks I'll look into it, but from the little that I read on RealTime, I was under the impression that it did not use MySQL or PostgreSQL which is a database feature that I was hoping to use. --Lonnie Why not just use the built-in database features to do what you want? Its called RealTime. Lots of info on it on the wiki. -Matthew - Original Message - From: [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Saturday, February 12, 2005 4:56 PM Subject: [Asterisk-Users] ast_data does not patch Hello all, I have just been trying to install the latest ast_data from: http://svn.asteriskdocs.org/res_data/ast_data/ into my cvs version of Asterisk and have found that the install patching fails. - patching file contrib/scripts/sip-friends.sql patching file contrib/scripts/iax-friends.sql patching file apps/app_voicemail.c patching file apps/app_directory.c patching file channels/chan_sip.c Hunk #2 succeeded at 621 (offset 9 lines). Hunk #3 FAILED at 1480. Hunk #4 succeeded at 1549 (offset 11 lines). Hunk #5 succeeded at 1617 (offset 18 lines). Hunk #6 succeeded at 1972 (offset 11 lines). 1 out of 6 hunks FAILED -- saving rejects to file channels/chan_sip.c.rej patching file channels/chan_iax2.c Hunk #2 succeeded at 593 with fuzz 2 (offset 13 lines). Hunk #3 FAILED at 944. Hunk #4 succeeded at 4441 (offset 57 lines). Hunk #5 FAILED at 5234. 2 out of 5 hunks FAILED -- saving rejects to file channels/chan_iax2.c.rej patching file Makefile patching file pbx.c Hunk #6 succeeded at 1390 (offset 18 lines). Hunk #8 succeeded at 1439 (offset 18 lines). Hunk #10 succeeded at 1508 (offset 18 lines). patching file asterisk.c Hunk #2 succeeded at 1922 (offset 76 lines). -- Does anyone know how to get in touch with the developer or have another viable and working option that will allow me to dynamically place my users information in a MySQL database? Thanks, Lonnie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Flash Pane - Monitor Parked Calls?
Thank you for the response ... Nicolas (the author of Flash Panel) had responded with this too, but you have to be using 0.20-unstable, where as I was using 0.19-stable. I have 0.20-unstable running and the park button works for the most part - seems to stay lit even after parking times out, but I'll wait for 0.20-stable before I say anything :-). Thanks again!! Bruce -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thorben Jensen Sent: Sunday, February 13, 2005 2:11 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: SV: [Asterisk-Users] Flash Pane - Monitor Parked Calls? Need help with how to configure for parked calls in the Flash Operator Panel's op_buttons.cfg file ... I've looked on the wiki, google and asternic's site and can't seem to find how to setup op_buttons.cfg to monitor parked calls. For example, if someone parks in 701, I'd like to see that represented on the panel. I've tried a number of things ... this is what I have now and it does not work ... [701] Position=12 Label=Park 701 Extension=701 Context=parkedcalls Icon=1 Any help would be great! Thanks, Bruce [EMAIL PROTECTED] Hi Bruce, Try this; I took this from the sample configuration: [PARK701] Position=17 Icon=3 Extension=700 Label=Park 701 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Speech Recognition
Hi Dean, You seem to have had your head up the supplier's arse for a number of years. :-) I last tried a Nuance demo system in about 2002, and found it useless. Speechworks (now scansoft) was rather better, but still useless for English. I'm British. Trying the British system gave poor results. Trying the US system seldom gave the right answer. Speechwork's Chinese (Cantonese and Mandarin) was pretty good, though. I've never seen Nuance offer free form speech to text, and I can't see Angel or Nuance's sites claiming that. They offer free form IVR input within a limited domain, which is something quite different - the set of possible outcomes is so much smaller. The best free form speech to text systems still require considerable user specific training to achieve reasonable accuracy. Some people eventually get good results, while others never do. Maybe some people just talk in a much more consistent way. Regards, Steve dean collins wrote: Steve then you have had your head up your arse for a number of years. Nuance was delivering 90% in 1999 and I have a number of happy customers to prove it. You also obviously didn't look at either the Nuance or angel sites because both of them offer free form speech to text capabilities. One of the first customers I had in Australia for Nuance was ordering of stock for Revlon cosmetics using a speech to an automated ordering system using their antiquated stock database. Dean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] iax.conf config and iax based clients
Try using context (with a trailing T!!) in your config, and lose the spaces around the equal sign, just in case. -Original Message- From: Wesley Jay Deypalan [mailto:[EMAIL PROTECTED] Sent: Saturday, February 12, 2005 9:33 PM To: [EMAIL PROTECTED]; asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] iax.conf config and iax based clients Hi, I changed the dialplan and the same error. By the way the * server has public IP address and the firefly clients are behind firewall(iptables). here is the error and config chan_iax2.c:5718 socket_read: Rejected connect attempt from 33.44.5.55, request '[EMAIL PROTECTED]' does not exist chan_iax2.c:5718 socket_read: Rejected connect attempt from 33.44.5.55, request '[EMAIL PROTECTED]' does not exist iax.conf [general] bindport=4569 bindaddr=2.3.4.5 bandwidth=low jitterbuffer=no tos=lowdelay [QIax1] type = friend host = dynamic accountcode = iaxy secret = 12345678 contex = from-iax disallow = all allow = ilbc allow = gsm auth = md5 trunk = no qualify = no [QIax2] type = friend host = dynamic accountcode = iaxy secret = 12345678 contex = from-iax disallow = all allow = ilbc allow = gsm auth = md5 trunk = no qualify = no [QIax3] type = friend host = dynamic accountcode = iaxy secret = 12345678 contex = from-iax disallow = all allow = ilbc allow = gsm auth = md5 trunk = no qualify = no extension.conf [general] static = yes writeprotect = yes [bogon-calls] exten = _.,1,Congestion [from-iax] exten = 105,1,Dial(IAX2/QIax1,20) ;exten = 105,2,Voicemail(u2000) ;exten = 105,102,Voicemail(b2000) exten = 105,103,Hangup exten = 106,1,Dial(IAX2/QIax2,20) ;exten = 106,2,Voicemail(u2001) ;exten = 106,102,Voicemail(b2001) exten = 106,103,Hangup exten = 107,1,Dial(IAX2/QIax3,20) ;exten = 107,2,Voicemail(u2002) ;exten = 107,102,Voicemail(b2002) exten = 107,103,Hangup TIA WEsley correct your dialplan. something like this [from-iax] exten = 105,1,Dial(IAX2/QIax1,20) exten = 106,1,Dial(IAX2/QIax2,20) exten = 107,1,Dial(IAX2/QIax3,20) hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Speech Recognition
The limited domain reference is obsolete, Telstra have a 2 million record database (yeh I know it's a lot smaller when you dice it phonetically but it's still big enough). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: Sunday, February 13, 2005 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Speech Recognition Hi Dean, You seem to have had your head up the supplier's arse for a number of years. :-) I last tried a Nuance demo system in about 2002, and found it useless. Speechworks (now scansoft) was rather better, but still useless for English. I'm British. Trying the British system gave poor results. Trying the US system seldom gave the right answer. Speechwork's Chinese (Cantonese and Mandarin) was pretty good, though. I've never seen Nuance offer free form speech to text, and I can't see Angel or Nuance's sites claiming that. They offer free form IVR input within a limited domain, which is something quite different - the set of possible outcomes is so much smaller. The best free form speech to text systems still require considerable user specific training to achieve reasonable accuracy. Some people eventually get good results, while others never do. Maybe some people just talk in a much more consistent way. Regards, Steve dean collins wrote: Steve then you have had your head up your arse for a number of years. Nuance was delivering 90% in 1999 and I have a number of happy customers to prove it. You also obviously didn't look at either the Nuance or angel sites because both of them offer free form speech to text capabilities. One of the first customers I had in Australia for Nuance was ordering of stock for Revlon cosmetics using a speech to an automated ordering system using their antiquated stock database. Dean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PLEASE HELP Adit 600 went kaput?
--- Andrew Kohlsmith [EMAIL PROTECTED] wrote: On February 12, 2005 07:31 pm, Richard Reina wrote: On thing that is odd is that although the t1 cross over cable is plugged in to both * and the Adit. Both t1 and t1 leds on the Adit are red. How can they both have the same status if one is hooke up and on is not? Could my cross over cable have some loose wiring? Unplug it and plug in a loopback plug (pin 1-5, pin 2-6) -- if the T1 alarm doesn't go away, the T1 controller itself is kaput. If it goes green (or off), then your wire is suspect. You can certainly have both T1 controllers showing alarm if you never turned the second one off. Honestly it sounds as if you didn't do *any* basic diagnostics here. Tell us what you *have* tried, and we can suggest other possible tests. The problem is that I don't know how to do any diagnostics. I'm having the telco wiring vendor come out today hopefully he will have a loop-back and/or another crossover cable. __ Do you Yahoo!? Meet the all-new My Yahoo! - Try it today! http://my.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Speech Recognition
On Mon, February 14, 2005 2:18, dean collins said: The limited domain reference is obsolete, Telstra have a 2 million record database (yeh I know it's a lot smaller when you dice it phonetically but it's still big enough). Maybe it's just me, but I found their database very hit and miss, not to mentioned biased towards their own services, for things such as internet... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the long run the pessimist may be proved right, but the optimist has a better time on the trip. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax.conf config and iax based clients
Try using context (with a trailing T!!) in your config, and lose the spaces around the equal sign, just in case. Well, I was wondering why the error log showed that the phones where in default context. That just show that I should never answer before my first coffee ;-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Speech Recognition
Oh yeh, their database admins have been playing funny games with the rules. It's been demonstrated on more than a few 'key words' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Duane Sent: Sunday, February 13, 2005 10:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Speech Recognition On Mon, February 14, 2005 2:18, dean collins said: The limited domain reference is obsolete, Telstra have a 2 million record database (yeh I know it's a lot smaller when you dice it phonetically but it's still big enough). Maybe it's just me, but I found their database very hit and miss, not to mentioned biased towards their own services, for things such as internet... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the long run the pessimist may be proved right, but the optimist has a better time on the trip. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk+GNOMEMeeting=No Sound.
On Sun, 2005-02-13 at 04:46 +0100, Andres Gmez Garca wrote: I've tried GNOMEMeeting also. It works fine with a P2P client connections (ALSA works fine) but, even when I success connecting to an asterisk server, I haven't hear anything. I mean, I don't hear the demo successfull messages. I've looking the GNOMEMeeting logs and it says that it closes the sound channel as soon as it connects to the asterisk server. This is my h323.conf file: Had the same issue with Debian Sarge. I didn't actually investigate it, but I strongly suspect the openh323/pwlib packages don't work with the asterisk-h323 package. The H323 README specifically says btw to don't use the packages of the distribution but rather the versions recommended there. I finally decided to compile * 1.0.5 from scratch, as well as use chan_oh323 instead of chan_h323, and all works well now. As to the linphone problems, don't know, it should work. If not, it'd be rather a linphone issue. As to an IAX phone, the only choice on linux currently seems to be iaxcomm/iaxclient. For me, it's not really usable because of latency issues, but to test the * installation it'll suffice anyway. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk@home .05 release questions on setup.
Roger Hanson wrote: I've downloaded 2x and burned 2 cds and get an error invalid compressed format (err=2) system halted message both times. It'd be nice to have a MD5 to verify my download is OK. It'd narrow down the problem to either the download or the burn, wouldn't it? Here is an _un-official_ md5sum from my burnable and installable image. 9d5657b7c833830b8a1fd1f024215d46 asteriskathome-0.5.iso I got it to install yesterday but ran into a couple of errors after the final reboot ( asterisk did compiled , etc. ) that I have to sort out today. One was I only had 128 MBytes of RAM and got an error that some program wouldn't run with less than 256. I'm doing this between LUG meetings and week-end jobs so if there is an error and I don't have time to work on it then I come back to it later. Bought more RAM yesterday. The other error was about FXS and FXO not being configured correctly and I suspect it was because I didn't have a phone line plugged in to the TDM400P. This for today also. What did you use to download the iso? If you're stuck with a Windows system and using MS's FTP program remember it defaults to ASCII mode. ( Sorry if this is a lame suggestion. :-) Rod -- --- [This E-mail scanned for viruses by Declude Virus] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk+GNOMEMeeting=No Sound.
Addendum: I did a little investigation and found this http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=272259 Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Open source CRM systems with * integration
Michiel van Baak wrote: On 10:56, Sun 13 Feb 05, John Middleton wrote: Has anyone any experience of the above. Key feature for me is tracking incoming and outgoing emails and linking them to the contact record. Thanks, sorry for the OT ;-) Hi, Have a look at http://www2.covide.net Maybe that's what you want. The project page is at http://sourceforge.net/projects/covide Have fun. Is there an online demo site? Thanks, Mike Clark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ast_data does not patch
beonice wrote: I don't know whether RealTime PostgreSQL, but I can't upgrade to RealTime anyway ... I need a stable version of asterisk, and the current stable version does not include RealTime. :( You need a stable version of Asterisk, but you're willing to patch with an unsupported change like ast_data? Seems a little contradictory to me :-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Open source CRM systems with * integration
On 11:52, Sun 13 Feb 05, Mike Clark wrote: Michiel van Baak wrote: On 10:56, Sun 13 Feb 05, John Middleton wrote: Has anyone any experience of the above. Key feature for me is tracking incoming and outgoing emails and linking them to the contact record. Thanks, sorry for the OT ;-) Hi, Have a look at http://www2.covide.net Maybe that's what you want. The project page is at http://sourceforge.net/projects/covide Have fun. Is there an online demo site? Thanks, Mike Clark Yes. But it will take till next week before the english demo forms will be online. If you think you can handle the dutch form look at: http://create.demo.covide.net -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MusicOnHold Native Mode, Please Clarify
Hi Guys, Ive attempted to get this moh-native thing to work with no success. Ive reviewed wiki, mantis and e-mail postings and Im confused. The latest Ive read is native moh should be in asterisk-addons in format_mp3, but what version will it work with? Ive tried asterisk 1.0.1, 1.0.5, addons 1.0.1, 1.0.4 and also r stable CVS. I followed the wiki example with no luck, all I get is unable to start music on hold at the console. Musiconhold.conf: [classes] [moh_files] default = /var/lib/asterisk/moh-native I have an mp3 that came with asterisk in this file: fpm-calm-river.mp3 extensions.conf: [moh] exten = 5551,1,Answer exten = 5551,2,WaitMusicOnHold(60) modules.conf: [modules] autoload=yes ; ; If you want, load the GTK console right away. ; Don't load the KDE console since ; it's not as sophisticated right now. ; noload = pbx_gtkconsole.so ;load = pbx_gtkconsole.so noload = pbx_kdeconsole.so ; ; Intercom application is obsoleted by ; chan_oss. Don't load it. ; noload = app_intercom.so ; ; Explicitly load the chan_modem.so early on to be sure ; it loads before any of the chan_modem_* 's afte rit ; noload = chan_modem.so noload = chan_modem_bestdata.so noload = chan_modem_i4l.so noload = chan_modem_aopen.so load = format_mp3.so load = res_musiconhold.so Console: Asterisk Dynamic Loader Starting: [format_mp3.so] = (MP3 format [Any rate but 8000hz mono optimal]) == Registered file format mp3, extension(s) mp3 [res_musiconhold.so] = (Music On Hold Resource) == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' *CLI dial 5551 -- Executing Answer(OSS/dsp, ) in new stack Console call has been answered -- Executing WaitMusicOnHold(OSS/dsp, 60) in new stack Feb 13 04:47:53 WARNING[11685]: res_musiconhold.c:370 moh1_exec: Unable to start music on hold for 60 seconds on channel OSS/dsp Hangup on console Any guidance will be appreciated. Thanks. JR ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cannot reset an IAXy box!!!
[EMAIL PROTECTED] wrote: asterisk-users-bounces at lists.digium.com wrote: Hi everyone, I was working yesterday and after I provide my IAXy box it loose any network comunication, the link light (green) is on and the activity light (orange) when the power is turned on it does nothing, but when I pickup the phone connected to the box, this light start blinking once per second. I've use ethereal to sniff a bit and I found that the box keeps asking to broadcast the MAC address for the IP of the asterisk server, the server answer but the IAXy miss it and keeps asking forever. A detail of the capture file of ethereal is attached to this message in plain text. The reset button does nothing (I've read that this button is just a cosmetic button here: http://lists.digium.com/pipermail/asterisk-users/2004-November /074909.html ). Any body has an idea to solve this issue??? Try re-programming the IAXy. That often fixes these types of problems. Well, I try to re-provisioning my IAXy box but it has no IP at all as I can see in the Ethereal capture: Sender MAC address: 00:0f:d3:00:0a:f0 (Digium_00:0a:f0) Sender IP address: 0.0.0.0 (0.0.0.0) Hmm. Any chance it's trying to get another IP address from DHCP? (from your description it doesn't sound like it, but who knows - that 0.0.0.0 address is kinda suspicious). And if I try to use the 0.0.0.0 IP, the loopback interface answer the request and no provisioning is made. Yeah, I can't see that working. 0.0.0.0 isn't really an address; more like a lack of one. That's what's got me wondering about DHCP. I'm still stuck, if I'm missing something obious please tell me what it is. Those IAXys are pretty quirky when it comes to configuration. You might need to run this one by Digium support. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.7 - Release Date: 10/02/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7912G: Takes the same firmware as 7940/60?
On Sun, 13 Feb 2005 12:58:47 +0100, B. Vallet - www.acropolistelecom.net [EMAIL PROTECTED] wrote: Here it is : http://www.cisco.com/cgi-bin/tablebuild.pl/ip-phone-7905 software is the same for 7905 / 7912 It's not actually. Firmware for both versions is available from that page, but each phone has its own firmware. -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk+GNOMEMeeting=No Sound.
El dom, 13-02-2005 a las 16:57 +0100, Bruno Hertz escribió: Had the same issue with Debian Sarge. I didn't actually investigate it, but I strongly suspect the openh323/pwlib packages don't work with the asterisk-h323 package. The H323 README specifically says btw to don't use the packages of the distribution but rather the versions recommended there. I finally decided to compile * 1.0.5 from scratch, as well as use chan_oh323 instead of chan_h323, and all works well now. Thanks Bruno, I'll try it. Greetings. -- Andrés Gómez García Computer Science Engineer Telf: +34 981 91 39 91 Fax: +34 981 91 39 49 mailto:[EMAIL PROTECTED] http://personales.igalia.com/agomez IGALIA, S.L. http://www.igalia.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mobile Wireless IP Phone
On 12 Feb 2005, at 19:46, eric m wrote: Hi! I would like to have feedback on wireless (wifi / 802.11b) IP phone to use with Asterisk PBX. Can you sugest model, The best and also the worst to use. I've been using the Zyxel P2000 for a month or so now. I was going to deploy several of them around the office, but after living with it for a while I'm not so sure. Good points: 1) it is light, feels fine and sounds fine (*) 2) it looks unthreatening and it works. 3) it is relatively cheap. 4) it gets on ok with * Bad points: 1) The UI is a disaster. I (often) press digits too fast for it (This is unforgivable in a phone) 2) It (almost) always swallows the first digit 'unlocking' itself. 3) Configuring WEP keys is a royal pain but you only do this a few times 4) the web gui is odd, but usable in IE (only) 5) picking it up off the charger doesn't auto-answer, you learn this quite quickly when you put it to your ear between rings and then it rings again. (talk about loud!) (*) it took a fair amount of work to get to the point where the sound is acceptable, you need either: ulaw and no WEP or G729 and wep Basically the chip doing WEP can't cope with 8k/s ulaw, but using g729 slows the data rate down enough for it to keep up. in a WLAN where asterisk is 'near' you seem to get better results by turning the packet sizes down, it is shipped tuned for internet use. In essence I think that the phone would be ok with a 50% faster CPU, but I guess the weight and power figures would suffer. So _definitely_ don't buy a box load until you have tried one for a while. Tim Thanks, eric. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://www.westhawk.co.uk/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Strategy for a stable IAXy
How do you make SIP work behind NAT without having to change anything on the firewall for example, those cable modems So far, Ive tested this using softphones and only iaxphone has been able to work using IAX, eye lite or something for FWD that uses SIP says it cant connect to the provider... So, which way to go? IAX or SIP? IAXy or Sipura? All ip phones use SIP right? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Nixon Sent: Jueves, 10 de Febrero de 2005 01:30 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Re: Strategy for a stable IAXy On Thursday 10 February 2005 20:35, Colin Anderson wrote: Why would someone choose these over other boxes, such as the Sipura 2000 and 3000? Because I want NAT traversal and a low bandwidth codec. That's the whole point of IAX2 as opposed to SIP. There is no low bandwidth codec available with the IAXy that I know of... Minimum is 32k + IP overhead SIP does work through NAT btw.. -- Peter Nixon http://www.peternixon.net/ PGP Key: http://www.peternixon.net/public.asc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sangoma A102 cards testing
We got also these problems and where searching like fools for solutions ... until the time we changed the main board of the server! (Interrupt sharing or Hyper threading stuff, I don't remember) we replaced the Supermicro board with an intel. Try the same config on another machine (maybe an older P3 or P4 or AMD) Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vikram Rangnekar Sent: zondag 13 februari 2005 13:02 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Sangoma A102 cards testing Does anyone have any experience ith configureing the sangoma A102 card for testing using a e1 cross cable i've configured and installed the cards properly even the lights on the card are green which proves that my cross cable is properly built too. my problem is with asterisk which gives me these errors PRI got event: HDLC Abort (6)on Primary D-channel of span 1 PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 No D-channels available! Using Primary on channel anyways 47! PRI: !! Not good - head of queue has not been transmitted yet I've tried everything i can think off with the wancfg configuration files here is my zaptel and zapata configs. span=1,0,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 span=2,1,0,ccs,hdb3 bchan=32-46 dchan=47 bchan=48-62 -- zapata.conf switchtype=euroisdn signalling=pri_net group=1 channel=1-15 channel=17-31 group=2 signalling=pri_cpe channel=32-46 channel=48-62 --- do i need to fool around with some jumpers on the card or something to activate internal clock on the card. zttol says INTERNALLY CLOCKED for both the ports. There are NO Alarms and no missed IRQ's I'm using asterisk 1.0.5 on debian with 2.4.29 kernel -- regards Vikram (http://www.vicramresearch.com) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7912G: Takes the same firmware as 7940/60?
On 04:06, Mon 14 Feb 05, Shaun Ewing wrote: On Sun, 13 Feb 2005 12:58:47 +0100, B. Vallet - www.acropolistelecom.net [EMAIL PROTECTED] wrote: Here it is : http://www.cisco.com/cgi-bin/tablebuild.pl/ip-phone-7905 software is the same for 7905 / 7912 When I go to that url i get: There are currently no files for this type. Do I need more access? I just registered a normal account. -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. signature.asc Description: Digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - SER Configuration
Yes, but I have to configure a route for each host in every host! A the moment i have about 120 Asterisk hosts and every astersk have about 50-100 users! Is for that I want a single sip proxy that route dial. I read more about ser, and the suggestion is to use ser for accounting and route, and asterisk only for PBX gateway and for voicemail. In my situation this isn't perfect because I have to use asterisk for sip login... Bye, Alberto On Fri, 2005-02-11 at 00:36, Steve Blair wrote: In my opinion this would be overkill. Just use Asterisk to forward calls to other Asterisk boxes. $0.02 Alberto Zuin wrote: Hello all! I'm new in this ML and I write you for a suggestion about integrate Asterisk and SER. My idea is to use Asterisk as a local PBX server where users can authenticate and make local calls, but when a user dial a non local number, an asterisk extension call SER Server who redirct to right remote asterisk. Originally I make this only with asterisk where in everyone I setted iax.conf to connect to every remote server. The size of my net in increasing, and then I want to modify it in a star center network and I want use ser in center to be sure to avoid rtp traffic. Now, you can point me to a working configuration example for asterisk and ser? Thanks, Alberto Zuin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATA's
Guys.. which ATA is better for connecting analog phones (features, stability, experiences, etc)? Sipura 2000 or Handy Tone 286, etc? What are you experiences? __ Anton Krall ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] soho fax suggestions?
Rich Adamson wrote: Steve, Need to replace our older soho fax machine with something more current. Would like to run the fax line through *, but haven't been able to make spandsp work correctly with digium TDM04b card. Our fax volume This seems to be a problem with the current wctdm driver. It seems to be broken for audio going out. I used to be able to send faxes reliably using spandsp and a TDM40P card, but I no longer can. I haven't had time to look in detail at what is wrong. I'd love to get this working for receive only. We probably get something in the neighborhood of 90% junk/spam faxes each week, and being able to view them online (and forward to the appropriate office) would be very helpful. Why not go to a fax to E-mail service then? For low volume Fax in, this works well. I receive no junk faxes to speak of, ( junk fax reception in the US is covered under some earlier telecom law, and senders can be subject to fines if they can be found ) John Novack ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Strategy for a stable IAXy
So, which way to go? IAX or SIP? IAXy or Sipura? I prefer by far IAX All ip phones use SIP right? Nope, now there's IAX hardphone, like there : http://www.iaxtalk.com/ hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-bsd] Asterisk not accepting multiple SIP phone logins
On Fri, 11 Feb 2005, Brian Buhrow wrote: Hello. You can't have two phones login with the same extension. You need to assign one phone to 101, and the other to 102. Set the user to 101 on one and 102 on the other. Actually, that isn't quite 100% accurate. The more accurate statement is that you can't have two phones log in as the same username/etc in sip.conf. You can, however have extensons.conf ring numerous phones all at the same time for a given extension. What you can do is set up two separate phone configurations in sip.conf, one per phone. I.E: [101-phone1] ...sip config... [101-phone2] ...sip config... and then modify your dial command in extensions.conf to look something like: exten = 101,1,Dial(SIP/101-phone1SIP/101-phone2,20,tr) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: [Asterisk-bsd] Asterisk not accepting multiple SIP phone logins
Forrest W. Christian wrote: Actually, that isn't quite 100% accurate. And even yours wasn't 100% accurate. Instead of messy extension lines you could setup a Queue as well. Flexibility, this is why Asterisk rules! Jeremy McNamara ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intermediary jitter buffering
On Sat, 12 Feb 2005, Michael Giagnocavo wrote: Hello, I understand that only the destination of a call should do jitter buffering. So, if IAX2/PhoneA calls IAX2/PhoneB through my server (no transfers), PhoneA and PhoneB need to perform their own jitter buffering, and Asterisk will just forward the frames, correct? What happens if the peer does not support jitter buffering, but is close by so there's no need for jitter buffering? My situation is that I have an Asterisk machine right in front of our provider's systems (same switch, 1ms latency). If they don't have jitter buffering, how can I force my Asterisk machine to jitter buffer calls from my users to them? Interesting question. Its an issue when you set up a IAX link between two close by peers. You tend to think that you don't need jitter buffer, or can set the settings down low. But if you take a call from that close-by peer that actually comes from somewhere else, and you suddenly might need a big buffer. So it would be nice if JB setting could somehow by negotiated. Anyway - a trick that comes to mind to get your packets dejittered before sending to the other box is to interpose a Local channel. So - when you handle the incoming call, on your intermediary machine, rather than Dial() the third box, rather dial a Local/ channel that then dials to 3rd machine in turn. Then, chan_iax2 will by bridged to the local/ channel, and will dejitter. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk@home .05 release questions on setup.
I found out the CD's I made were OK - I used one on a different computer and it worked fine. [EMAIL PROTECTED] doesn't like the current Asterisk box I'm using now. It's an IBM Netfinity 3500 - dual 233MHz processor, SCSI, 512MB, DVD-ROM, blah, blah. That's the only computer I get the error message with. I'll buy a new computer and switch others around and put it on a different one. Thanks for the offers for CD's, but for some reason [EMAIL PROTECTED] doesn't like the IBM computer. I know it's only a dual P-II 233 system and [EMAIL PROTECTED] states it wants 300MHz, but my current AMP/Asterisk installation works great on the box now. - Original Message - From: Roderick A. Anderson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, February 13, 2005 10:20 AM Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] .05 release questions on setup. Roger Hanson wrote: I've downloaded 2x and burned 2 cds and get an error invalid compressed format (err=2) system halted message both times. It'd be nice to have a MD5 to verify my download is OK. It'd narrow down the problem to either the download or the burn, wouldn't it? Here is an _un-official_ md5sum from my burnable and installable image. 9d5657b7c833830b8a1fd1f024215d46 asteriskathome-0.5.iso I got it to install yesterday but ran into a couple of errors after the final reboot ( asterisk did compiled , etc. ) that I have to sort out today. One was I only had 128 MBytes of RAM and got an error that some program wouldn't run with less than 256. I'm doing this between LUG meetings and week-end jobs so if there is an error and I don't have time to work on it then I come back to it later. Bought more RAM yesterday. The other error was about FXS and FXO not being configured correctly and I suspect it was because I didn't have a phone line plugged in to the TDM400P. This for today also. What did you use to download the iso? If you're stuck with a Windows system and using MS's FTP program remember it defaults to ASCII mode. ( Sorry if this is a lame suggestion. :-) Rod -- --- [This E-mail scanned for viruses by Declude Virus] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA's
Anton Krall wrote: Guys.. which ATA is better for connecting analog phones (features, stability, experiences, etc)? Sipura 2000 or Handy Tone 286, etc? What are you experiences? In my experience the Sipura 2000 has three hardware advantages: * 2 independent phone ports * Mounting holes * The price for a single Sipura 2000 is less than the price for two Grandstreams. As far as software and compatibility with * goes, I only have experience in a LAN environment, where both worked (with the right firmware) without a problem. The Sipuras seem a little bit louder (or so the users tell me). -- Best regards Peer Oliver Schmidt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dlink VPNs??
Hi, I am thinking of purchasing a cheap Dlink VPN for testing purposes for use with my Asterisk box and would like to ask the list for advice on how to pick a VPN that will work with my box. I am a newbie to both VPN's and Asterisk so any advice will be appreciated. Thanks, Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA's
The Sipuras have a ton of configurable parameters. If you understand them (and there is no good manual, unfortunately) then you can be of great benefit. Otherwise they'll be worthless. I particularly miss the dial-plan, distinctive ring and audio gain options on the Grandstreams. Remote syslog can also be useful for debugging. It all depends what you need, I guess. Further, the Sipuras have a more detailed status, that is accessible WHILE you are engaged in a conversation. I think you're paying a bit more for the 1000 (1 line version) as compared to the Grandstream 286, but if you need/want two independent lines, then the Spa 2000 is more economical (as Peter said). --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ast_data does not patch
Heh. Good point, Kevin. I didn't realise that ast_data was also a third party add-on. :) So I submitted a bug report to digium with my gdb trace (http://bugs.digium.com/bug_view_page.php?bug_id=0003580), and markster there suggested that I should update to the latest stable asterisk from CVS. I did. And now the core asterisk can see my voicemail configuration in MySQL just fine. I must have originally retrieved a buggy version of the stable asterisk. :) Thanks, everyone, for all your help! Cheers, Maya --- Kevin P. Fleming [EMAIL PROTECTED] wrote: beonice wrote: I don't know whether RealTime PostgreSQL, but I can't upgrade to RealTime anyway ... I need a stable version of asterisk, and the current stable version does not include RealTime. :( You need a stable version of Asterisk, but you're willing to patch with an unsupported change like ast_data? Seems a little contradictory to me :-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? All your favorites on one personal page Try My Yahoo! http://my.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7912G: Takes the same firmware as 7940/60?
On 19:36, Sun 13 Feb 05, Stefan Gofferje wrote: Michiel van Baak schrieb: On 04:06, Mon 14 Feb 05, Shaun Ewing wrote: On Sun, 13 Feb 2005 12:58:47 +0100, B. Vallet - www.acropolistelecom.net [EMAIL PROTECTED] wrote: Here it is : http://www.cisco.com/cgi-bin/tablebuild.pl/ip-phone-7905 software is the same for 7905 / 7912 When I go to that url i get: There are currently no files for this type. Do I need more access? I just registered a normal account. Ye need to be Cisco [Voice|Silver|Gold] Partner, CCIE (not sure about this) or have a service contract for this phone reg'd to your CCO account. Thnx. No luck for me I guess. chan_sccp it will be. -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. signature.asc Description: Digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA's
Good evening, allow me to join in right here. Which ATA/TA would you suggest for connecting analogue fax machines to Asterisk? One of the ones named before or e.g. a ATA-186 made by Cisco? Cheers Sascha The Sipuras have a ton of configurable parameters. If you understand them (and there is no good manual, unfortunately) then you can be of great benefit. Otherwise they'll be worthless. I particularly miss the dial-plan, distinctive ring and audio gain options on the Grandstreams. Remote syslog can also be useful for debugging. It all depends what you need, I guess. Further, the Sipuras have a more detailed status, that is accessible WHILE you are engaged in a conversation. I think you're paying a bit more for the 1000 (1 line version) as compared to the Grandstream 286, but if you need/want two independent lines, then the Spa 2000 is more economical (as Peter said). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Still stuck trying to make Asterisk read MySQL (SOLVED)
Thanks to everyone who responded. I submitted a bug report to digium (http://bugs.digium.com/bug_view_page.php?bug_id=0003580), and markster responded, suggesting that I get an updated version of stable asterisk from CVS. I did, and now it's all working fine. I must have initially downloaded a not-quite-stable stable asterisk. :) The only glitch I seem to notice is that it demands that the VM context be 'default'. :) I set it to something else, and the generated SQL code is still looking for context=default. Oh well, at least that's easy to work around. Thanks again, Maya --- Joe Dennick [EMAIL PROTECTED] wrote: I've been working with RealTime configuration from MySQL Server, and have had good results. You might check it out. You can do a search for 'realtime' on the Wiki and get some good documentation on how to set it up. I think in the extconfig.conf file, not only do you need to identify the engine (ODBC in your case), but you also need to identify the actual table you used for your Voicemail configuration. If I recall correctly, the default is a table named 'voicemail' and since you are using a different name, you need to specify the name in the extconfig.conf file so it can find it. beonice ([EMAIL PROTECTED]) wrote: I've been continuing to experiment with MySQL. I'm having absolutely no luck getting asterisk to read voicemail configuration data and mailbox configuration data from mysql tables instead of from voicemail.conf. The default Asterisk setup that reads from voicemail.conf and extensions.conf works fine. I'm using Asterisk CVS-v1-0-12/12/04-15:58:29 on a Whitebox Enterprise Linux box. I'm not using any telephony hardware or SIP phones. I've just got a voicepulse DID talking to asterisk via IAX. I've got mysql downloaded and installed and have successfully got the contributed script reading from my asterisk_vm database to set up the extensions.conf, as per the instructions at: http://www.voip-info.org/wiki-Asterisk+extensions+from+mysql Now I'm trying to get Asterisk to look up voicemail configs from the asterisk_vm database. In order to do this, I've been following the instructions at: http://www.voip-info.org/wiki-Asterisk+voicemail+database So, I've: 1) Updated the /usr/src/asterisk/apps/Makefile to have USE_MYSQL_VM_INTERFACE=1 and recompiled asterisk, with make clean; make; make install 2) Updated voicemail.conf to have the appropriate entries: dbuser=username ;; Yes I changed this to my username dbpass=password ;; Yes I changed this to my password dbhost=localhost dbname=asterisk_vm 3) Created the users table in the asterisk_vm database. +-++--+--+---+---+++ | context | mailbox| password | fullname | email | pager | options| stamp | +-++--+--+---+---+++ | default | | 1234 | Moron Tester | [EMAIL PROTECTED] | | attach=yes | 20050211131641 | +-++--+--+---+---+++ 4) Updated extensions.conf to have the following line: exten = ,1,VoiceMail(u) I tried restarting asterisk at this point, called in and tried to leave voicemail for extension (and mailbox) . Here's the message I get: *CLI Feb 11 13:21:36 WARNING[18393]: app_voicemail.c:1539 leave_voicemail: No entry in voicemail config file for '' So I dug around some more and found http://www.voip-info.org/wiki-Asterisk+res_config Decided to try these instructions as well. So: 5) I created the ast_config table as directed: Here is the data: ++++---++--+--+-+ | id | cat_metric | var_metric | commented | filename | category | var_name | var_val | ++++---++--+--+-+ | 1 | 0 | 0 | 0 | voicemail.conf | default | | | ++++---++--+--+-+ 6) I edited /etc/asterisk/configs/res_odbc.conf to contain: [mysql1] dsn = MySQL-asterisk username = myuser password = mypass pre-connect = yes [mysql1] dsn = asterisk_vm username = myuser ;; changed to my userid on mysql password = mypass ;; changed to my password on mysql pre-connect = yes [mysql2] dsn = MySQL2-asterisk username = myuser2 password = mypass2 enabled = no [ENV] VAR=VALUE 7) Inserted glue to tell asterisk where to look: ; /etc/asterisk/res_config_odbc.conf [settings] table = ast_config connection = mysql1 8) Rerouted Asterisk's config engine: ;
Re: [Asterisk-Users] ATA's
Sascha E. Pollok wrote: Good evening, allow me to join in right here. Which ATA/TA would you suggest for connecting analogue fax machines to Asterisk? One of the ones named before or e.g. a ATA-186 made by Cisco? At the moment I am deploying Grandstream ATAs for faxing machines with out a problem so far. -- Best regards Peer Oliver Schmidt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] connect asterisk to ISDN in China
Hi, I plan to install asterisk and connect it to telco through ISDN in China. I'd love to know if the ISDN standard in China has any difference than in America before I buy the digium card. anybody has experience in it? or anybody who installed asterisk with ISDN in asia can share their expierience? Or, can anybody give me some links to educate me ISDN knowledge about the difference in China? (My heard there is something different there, but i dont know the details.) Thanks __ Do you Yahoo!? The all-new My Yahoo! - What will yours do? http://my.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ATA's
-Original Message- From: Luki [mailto:[EMAIL PROTECTED] The Sipuras have a ton of configurable parameters. If you understand them (and there is no good manual, unfortunately) Really? 87 pages aren't enough for you? http://www.sipura.com/Documents/SipuraSPAUserGuidev2.0.9.pdf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No CallerID on TDM11B?
Hi list! I'm not getting incoming CallerID in The Netherlands on my TDM11B. Everything was configures according to the docs at digium.com. The error on the console is this: Feb 13 16:49:40 ERROR[16123]: callerid.c:260 callerid_feed: fsk_serie made mylen 0 (-84) Feb 13 16:49:40 WARNING[16123]: chan_zap.c:5396 ss_thread: CallerID feed failed: Success Feb 13 16:49:40 WARNING[16123]: chan_zap.c:5438 ss_thread: CallerID returned with error on channel 'Zap/4-1' Any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cannot reset an IAXy box!!!
Jim Van Meggelen wrote: Yeah, I can't see that working. 0.0.0.0 isn't really an address; more like a lack of one. That's what's got me wondering about DHCP. The IAXy does not use DHCP, it uses the older BOOTP protocol. Most DHCP servers support BOOTP (but it may have to be enabled) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Strategy for a stable IAXy
Anton Krall wrote: How do you make SIP work behind NAT without having to change anything on the firewall for example, those cable modems So far, Ive tested this using softphones and only iaxphone has been able to work using IAX, eye lite or something for FWD that uses SIP says it cant connect to the provider... So, which way to go? IAX or SIP? IAXy or Sipura? All ip phones use SIP right? Generally you just set nat=yes, canreinvite=no, and qualify=yes in sip.conf and that's it. No need for STUN or any of that other crap. This assumes that Asterisk is on a public IP, of course. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7912G: Takes the same firmware as 7940/60?
Michiel van Baak wrote: On 04:06, Mon 14 Feb 05, Shaun Ewing wrote: On Sun, 13 Feb 2005 12:58:47 +0100, B. Vallet - www.acropolistelecom.net [EMAIL PROTECTED] wrote: Here it is : http://www.cisco.com/cgi-bin/tablebuild.pl/ip-phone-7905 software is the same for 7905 / 7912 When I go to that url i get: There are currently no files for this type. Do I need more access? I just registered a normal account. You need a CCO account with the correct permissions. Unless you have a support contract you can't get the firmware. It's not free. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: MusicOnHold Native Mode, Please Clarify
Hey guys, I got moh-native working with todays CVS of asterisk and asterisk-addons so Im guessing there were some code problems with versions 1.0.1, 1.0.4 and current CVS stable. Following the wiki instructions worked fine. Also the mp3s that come with Asterisk sound perfect, whereas my own mp3s have some sound pops, I suspect due to compression not at 8K mono but 128k stereo. I did have some compilation errors in todays CVS head with the dundi app, so I commented it out in the Makefile in /usr/src/asterisk/pbx/ then asterisk compiled fine. I do prefer to use the stable code, not CVS head. Does anyone have an idea what could be causing native-moh to work in CVS head and not CVS stable? Thanks. JR ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Broadvoice international dialling question
Id be grateful if someone could point me in the right
direction.
I have a Broadvoice trunk attached to Asterisk which I use
for frequent calls to the UK
using the following in extensions.conf
exten =
_0[1-68].,1,Ringing
exten =
_0[1-68].,2,Dial(SIP/BV/01144${EXTEN:1})
exten =
_0[1-68].,3,Hangup
The caller hears immediate ringing, though it seems that
Broadvoice takes a long time to make the international connection and sometimes
fails altogether
-- Executing
Ringing(SIP/100-4ad1, ) in new stack
-- Executing Dial(SIP/100-4ad1,
SIP/BV/011441234654321) in new stack
-- Called BV/011441234654321
-- SIP/BV-9dfd is ringing
-- SIP/BV-9dfd answered
SIP/100-4ad1
or
-- Executing
Ringing(SIP/100-3894, ) in new stack
-- Executing
Dial(SIP/100-3894, SIP/BV/011441234654321) in new
stack
-- Called BV/011441234654321
-- Got SIP response 408
Request Timeout back from 147.135.0.128
== No one is available to
answer at this time
-- Executing
Hangup(SIP/100-3894, ) in new stack
147.135.0.128 is Broadvoices server and I understand
that I need to take the request timeout issue up with them, but can anyone
suggest how I might configure Asterisk to perform an unattended transfer rather
than giving misleading ring-tones even when the destination phone is not
ringing? I feel that it gives a clearer indication of call progress to
have a long silence after dialing, followed by the ring (or congestion tone),
rather than the current immediate ringing.
I have searched for extensions.conf examples of this, but
havent come across any which work for me.
Many thanks,
Malcolm
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Re: [Asterisk-Users] 7912G: Takes the same firmware as 7940/60?
On 20:08, Sun 13 Feb 05, Stefan Gofferje wrote: Michiel van Baak schrieb: Thnx. No luck for me I guess. chan_sccp it will be. Not for the 79[05|12]... At least my 7905 does not like chan_sccp too much and they crashed my * (1.0.5)... unless you bounty the chan_sccp developers for 79[05|12] support OR ask your local Cisco dealer for a 79[05|12] SIP-license (which comes on CD). If you are (as your name suggests) from .nl, I would recommend www.zendus.de. They turned out to have really good prices... Thnx, My 7905g is working ok with chan_sccp. But only basic features work. What does work: Voicemail led. Directories for missed/received/placed calls getting/placing calls transfering using the # key That's about it. And from what I read the SIP image can really use the rest of the phone like speed dial, call forward etc. who knows my tax refunds allow me to buy the sip image ;) (if there's money left after the new laptop and new fileserver case) The prices @ zendus are WAY better than anything I seen here in .nl (the phone alone was 250 euro or something here in nl) -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. signature.asc Description: Digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] soho fax suggestions?
This seems to be a problem with the current wctdm driver. It seems to be broken for audio going out. I used to be able to send faxes reliably using spandsp and a TDM40P card, but I no longer can. I haven't had time to look in detail at what is wrong. I'd love to get this working for receive only. We probably get something in the neighborhood of 90% junk/spam faxes each week, and being able to view them online (and forward to the appropriate office) would be very helpful. Why not go to a fax to E-mail service then? For low volume Fax in, this works well. I receive no junk faxes to speak of, ( junk fax reception in the US is covered under some earlier telecom law, and senders can be subject to fines if they can be found ) Any suggestions on a reputable service? (personal email is fine) Rich ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Sangoma A102 cards testing
+++ Michael Devenijn [13/02/05 18:23 +0100]: We got also these problems and where searching like fools for solutions ... until the time we changed the main board of the server! (Interrupt sharing or Hyper threading stuff, I don't remember) we replaced the Supermicro board with an intel. Try the same config on another machine (maybe an older P3 or P4 or AMD) Michael Actually I am using a supermicro board the P4SCI wonder if I can turn off hyperthreading i dont think there is a bio option i'm running kernel 2.4.29 does it use hyperthreading and can i turn it off ? i dont think its a interrupt problem since the wanpipe hardware seems to be getting interrupts CPU0 0: 98454IO-APIC-edge timer 1: 3974IO-APIC-edge keyboard 8: 3IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 14: 53337IO-APIC-edge ide0 18:482 IO-APIC-level eth0 24:3525044 IO-APIC-level wanpipe1, wanpipe2 NMI: 0 LOC: 99917 ERR: 0 MIS: 0 did all the problems disappear after you changed over to intel. I still think maybe its somthing to do with some jumpr on the A102 card which i need to set to make the card use an internal clock. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vikram Rangnekar Sent: zondag 13 februari 2005 13:02 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Sangoma A102 cards testing Does anyone have any experience ith configureing the sangoma A102 card for testing using a e1 cross cable i've configured and installed the cards properly even the lights on the card are green which proves that my cross cable is properly built too. my problem is with asterisk which gives me these errors PRI got event: HDLC Abort (6)on Primary D-channel of span 1 PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 No D-channels available! Using Primary on channel anyways 47! PRI: !! Not good - head of queue has not been transmitted yet I've tried everything i can think off with the wancfg configuration files here is my zaptel and zapata configs. span=1,0,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 span=2,1,0,ccs,hdb3 bchan=32-46 dchan=47 bchan=48-62 -- zapata.conf switchtype=euroisdn signalling=pri_net group=1 channel=1-15 channel=17-31 group=2 signalling=pri_cpe channel=32-46 channel=48-62 --- do i need to fool around with some jumpers on the card or something to activate internal clock on the card. zttol says INTERNALLY CLOCKED for both the ports. There are NO Alarms and no missed IRQ's I'm using asterisk 1.0.5 on debian with 2.4.29 kernel -- regards Vikram (http://www.vicramresearch.com) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- regards Vikram (http://www.vicramresearch.com) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP 3000 configuration
I have set Asterisk as a gateway on the Polycom and set gatekeeper to
"No"
So to dial on the Polycom I would then dial (0+the number). No way to
just dial directly without the 0?
The other side of this is how do I dial "to" the Polycom, I have tried
everything that I can think of for the "exten" definition and nothing
seems to work.
I did this setup via the web interface so I can't test until Monday.
Thanks
Tim Courcy wrote:
You need to
set the asterisk as a gateway
in the polycom.. then to dial out. Lets say you set the * as GW 0 on
the
polycom you would dial 0*{exten} in order to dial through a gw on the
ip3000
you have to use the prefix for the gateway. So 0* for GW 0 and 1* for
GW 1
Hope this
helps if you need more info mail
me off list.
Thanks
Tim
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Scott Henderson
Sent: Saturday,
February 12, 2005
7:12 PM
To: Asterisk Users
Mailing List -
Non-Commercial Discussion
Subject: Re:
[Asterisk-Users]
Polycom IP 3000 configuration
I see that typo I made for
this suggestion, but the
real problem is that the system doesn't seem to register with Asterisk.
I can't dial out or even if I fix the error in my config will I be able
to dial
the extension.
This phone just doesn't seem to want to work with Asterisk. I have
found
some old posts where people got this phone to work but they never post
the
solution so i am hopeful someone has the answer.
Scott Henderson
Finite Technologies Incorporated
3763 Image Drive, Anchorage, Alaska 99504
Phone: 907.339.8085 ext 6101, Fax: 907.333.4482
http://www.finite-tech.com
http://www.chillywall.com
http://www.virtuale.cc
http://www.mphage.com
Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK
harry gaillac wrote:
hello
try: exten = 8908,1,Dial(h323/8908,20,Ttr) !
harry
--- Scott Henderson [EMAIL PROTECTED] a crit :
I am trying to add a Polycom IP 3000 to our Asterisk
system and am not
getting anywhere.
h323.conf
[8908]
type=friend
host=192.168.104.25
secret=polycom
context=crv-default
callerid="Conference Room Polycom"
extensions.conf
exten = 8908,1,Dial(h323/polycom,20,Ttr)
; Polycom
exten = 8908,2,Hangup
I have tried setting the Asterisk system as both
gatekeeper and gateway
in the polycom config.
To date nothing seems to work and Polycom is now on
a week return a
support call to the reseller that sold us the unit.
--
Scott Henderson
Finite Technologies Incorporated
3763 Image Drive, Anchorage, Alaska 99504
Phone: 907.339.8085 ext 6101, Fax: 907.333.4482
http://www.finite-tech.com
http://www.chillywall.com
http://www.virtuale.cc
http://www.mphage.com
Current Local Time:
http://www.worldtimeserver.com/time.asp?locationid=US-AK
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[Asterisk-Users] TDM-400P Sound Quality issues
I'm running a TDM-400P with 2 x FXS and 2 x FXO. I'm finding that there seems to be an odd relationship to sound quality on the card to my local when connecting via a SIP client. When I'm on my local network, if I connect to Asterisk via a SIP client (such as x-pro), and dial an outside line through the card, sound quality seems quite good. However, when I'm at a remote location and connect via the same SIP client and dial an outside line, the audio quality is fuzzy, sometimes quiet, and generally more difficult to understand. I spent a bunch of time troubleshooting the SIP end of things, thinking that's where the problem was, until I realized that every other SIP connection I make (from remote) yields a high quality call. ie. I can dial another SIP client and maintain high quality audio. Additionally, I can dial an extension that not only SIP connects to my server, but from there goes out an IAX2 connection to another remote Asterisk server, from there to another SIP client, and the audio quality is excellent. Therefore, I don't think the audio issue I'm experiencing is on the SIP end. Are there some wierd SIP - ZAP timing / conversion / other issues that could be causing this? thoughts? regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM-400P alternatives?
Are there any other relatively low cost analog cards available? I'm interested in finding something that might work a bit more reliably than the TDM-400P regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] soho fax suggestions?
On Feb 12, 2005, at 4:38 PM, Rich Adamson wrote: For planning purposes, is it appropriate to think in terms of purchasing a t38 capable box even if its not supported by * today? (I'm well aware of the bounty and Steve's work.) That's what I would do. In fact, I already have T.38 capable VOIP adapter (an Azatel 200) for my current fax machine but plan to upgrade that box to a Sipura 2100. If now is the time to purchase a t38 capable fax machine, anyone have any suggestions on a low-volume soho-sized box? I don't think there is such a thing as a T38 capable fax machine. T38 is for faxing over VOIP and I have yet to see a fax machine with a built-in network port so it can connect directly to the Internet...if you know what I mean. FWIW, I have had absolutely zero problems receiving faxes over VOIP, via Asterisk, using Voicepulse Connect, IAX trunking, and g.711. My problems for faxing are all related to outbound faxing (using the same Voicepulse setup or Sixtel iax.cc). Not sure why outbound is giving me problems when inbound isn't giving me any. shrug Of course I need to fax outbound more often than I need to receive inbound! -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What quad/octo BRI cards are best/stable for EuroISDN and Asterisk ?
I'm currently deciding on what card to pruchase for octo/quad BRI card to use with Asterisk on EuroISDN lines. I'm aware of at least two options (Junghanns or Beronet), but don't know how stable and well supported they are. Which ones are better supported ? Any experiences? Any advice ? How tos ? Using Junghanns you can have support. Maciej Kietlinski ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaphfc NOTICE[6799]: chan_zap.c:7685 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1
Hi, my success story with the zaphfc incl. florz patch has been to early. Allthough sound drop outs no longer happen, the following happens after a longer period (2 days) of inactivity on the asterisk box. Feb 13 22:30:15 NOTICE[6799]: chan_zap.c:7685 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Maybe this is helpful to find where the problem is. I will go and unload the drivers (and hope not to crash the box). -- Best regards Peer Oliver Schmidt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk+GNOMEMeeting=No Sound.
On Sun, 2005-02-13 at 18:10 +0100, Andres Gmez Garca wrote: Thanks Bruno, I'll try it. Also, you might take a look again at http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=272259 Following your mail, I wrote to that list (cf the last mails there), and it looks like a working oh323 package will turn up soon. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Sangoma A102 cards testing
On 21:47, Sun 13 Feb 05, Vikram Rangnekar wrote: +++ Michael Devenijn [13/02/05 18:23 +0100]: We got also these problems and where searching like fools for solutions ... until the time we changed the main board of the server! (Interrupt sharing or Hyper threading stuff, I don't remember) we replaced the Supermicro board with an intel. Try the same config on another machine (maybe an older P3 or P4 or AMD) Michael Actually I am using a supermicro board the P4SCI wonder if I can turn off hyperthreading i dont think there is a bio option i'm running kernel 2.4.29 does it use hyperthreading and can i turn it off ? Hi, Kernel 2.4 does not have HT support. You can check by running: cat /proc/cpuinfo It will list info for CPU 0 only. -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] soho fax suggestions?
Rich Adamson wrote: <> This seems to be a problem with the current wctdm driver. It seems to be broken for audio going out. I used to be able to send faxes reliably using spandsp and a TDM40P card, but I no longer can. I haven't had time to look in detail at what is wrong. I'd love to get this working for receive only. We probably get something in the neighborhood of 90% junk/spam faxes each week, and being able to view them online (and forward to the appropriate office) would be very helpful. Why not go to a fax to E-mail service then? For low volume Fax in, this works well. I receive no junk faxes to speak of, ( junk fax reception in the US is covered under some earlier telecom law, and senders can be subject to fines if they can be found ) Any suggestions on a reputable service? (personal email is fine) I use JFAX which I think is also known as Efax. If you are open to a new fax number anywhere else in the US from your home Zip code, then it is free. Otherwise there is a quarterly fee. AFAIK, you can't port an existing number to them, but I could be off on that. http://www.j2.com/jconnect/twa/page/servicesOverview Premier is what they try to direct you to, as they charge something for that one. John Novack ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom 190's vs Softphone
I have been playing with asterisk for a couple of weeks now and I have been very happy with its performance. However, I have run into a problem with how I want to deploy this solution. I have a mix of softphones (SJ and Xlite), ATA's, and a couple of IP phones (Snom 190). The asterisk box is on the public network. For my primary users they will reside behind a watchguard 4500 firewall. For the others there will be a mix of Ethernet routers (Linksys/Netgear/DLink). I have been testing the different deployments and have found that the softphones and ATA's work like a champ in getting around the firewall and the NAT of the work at home users. However the Snom phones don't perform as well. I have played with qualify statement in sip.conf and at first I thought that I might have been OK with not using that argument, but it didn't work. I continue to get the following: *CLI Feb 13 15:50:59 NOTICE[14043]: chan_sip.c:7971 sip_poke_noanswer: Peer '9093' is now UNREACHABLE! When this happens, I can't receive any calls (of course) but I can place outbound calls. Is this normal? Can anyone help me with this? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Sangoma A102 cards testing
At 03:36 PM 2/13/2005, you wrote: On 21:47, Sun 13 Feb 05, Vikram Rangnekar wrote: +++ Michael Devenijn [13/02/05 18:23 +0100]: We got also these problems and where searching like fools for solutions ... until the time we changed the main board of the server! (Interrupt sharing or Hyper threading stuff, I don't remember) we replaced the Supermicro board with an intel. Try the same config on another machine (maybe an older P3 or P4 or AMD) Michael Actually I am using a supermicro board the P4SCI wonder if I can turn off hyperthreading i dont think there is a bio option i'm running kernel 2.4.29 does it use hyperthreading and can i turn it off ? Hi, Kernel 2.4 does not have HT support. You can check by running: cat /proc/cpuinfo It will list info for CPU 0 only. -- You mean like this? # uname -a Linux 2.4.20-31.9smp #1 SMP Tue Apr 13 17:40:10 EDT 2004 i686 i686 i386 GNU/Linux # uptime 16:01:24 up 218 days, 11:29, 1 user, load average: 0.00, 0.00, 0.00 # cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 15 model : 2 model name : Intel(R) Pentium(R) 4 CPU 2.40GHz stepping: 9 cpu MHz : 2395.944 cache size : 512 KB physical id : 0 siblings: 2 fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 2 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm bogomips: 4784.12 processor : 1 vendor_id : GenuineIntel cpu family : 15 model : 2 model name : Intel(R) Pentium(R) 4 CPU 2.40GHz stepping: 9 cpu MHz : 2395.944 cache size : 512 KB physical id : 0 siblings: 2 fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 2 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm bogomips: 4784.12 Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP 3000 configuration
It is 0* + number not 0 + number only
other way is to use a gatekeeper and register the asterisk and the polycom to
it..
In my h323.conf
[4500]
type=user
host=10.10.10.59
context=default
in my extensions.conf
[h323]
exten = 4200,1,Dial,H323/10.10.10.49
exten = 4300,1,Dial,H323/10.10.10.47
exten = 4500,1,Dial,H323/10.10.10.59
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott Henderson
Sent: Sunday, February 13, 2005
3:54 PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Polycom IP 3000 configuration
I have set Asterisk as a gateway on the Polycom and
set gatekeeper to No
So to dial on the Polycom I would then dial (0+the number). No way to
just dial directly without the 0?
The other side of this is how do I dial to the Polycom, I have
tried everything that I can think of for the exten definition and
nothing seems to work.
I did this setup via the web interface so I can't test until Monday.
Thanks
Tim Courcy wrote:
You
need to set the asterisk as a gateway in the polycom.. then to dial out. Lets
say you set the * as GW 0 on the polycom you would dial 0*{exten} in
order to dial through a gw on the ip3000 you have to use the prefix for the
gateway. So 0* for GW 0 and 1* for GW 1
Hope this helps if you need more info mail
me off list.
Thanks
Tim
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]
On Behalf Of Scott Henderson
Sent: Saturday, February 12, 2005
7:12 PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Polycom IP 3000 configuration
I see that typo I made for this suggestion, but the
real problem is that the system doesn't seem to register with Asterisk.
I can't dial out or even if I fix the error in my config will I be able to dial
the extension.
This phone just doesn't seem to want to work with Asterisk. I have found
some old posts where people got this phone to work but they never post the
solution so i am hopeful someone has the answer.
Scott HendersonFinite Technologies Incorporated3763 Image Drive, Anchorage, Alaska 99504Phone: 907.339.8085 ext 6101, Fax: 907.333.4482http://www.finite-tech.comhttp://www.chillywall.comhttp://www.virtuale.cchttp://www.mphage.comCurrent Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK
harry gaillac wrote:
hellotry: exten = 8908,1,Dial(h323/8908,20,Ttr) !harry --- Scott Henderson [EMAIL PROTECTED] a écrit :
I am trying to add a Polycom IP 3000 to our Asterisksystem and am not getting anywhere.h323.conf[8908]type=friendhost=192.168.104.25secret=polycomcontext=crv-defaultcallerid=Conference Room Polycomextensions.confexten = 8908,1,Dial(h323/polycom,20,Ttr) ; Polycom exten = 8908,2,HangupI have tried setting the Asterisk system as bothgatekeeper and gateway in the polycom config.To date nothing seems to work and Polycom is now ona week return a support call to the reseller that sold us the unit.-- Scott Henderson
Finite Technologies Incorporated3763 Image Drive, Anchorage, Alaska 99504Phone: 907.339.8085 ext 6101, Fax: 907.333.4482http://www.finite-tech.comhttp://www.chillywall.comhttp://www.virtuale.cchttp://www.mphage.comCurrent Local Time:
http://www.worldtimeserver.com/time.asp?locationid=US-AK
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[Asterisk-Users] Who makes these phones?
http://www.broadbandphone.com.au/global/pnp.htm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Who makes these phones?
On Mon, 14 Feb 2005 09:53:36 +1100, PHP Mechanic wrote: http://www.broadbandphone.com.au/global/pnp.htm They look like they are all PA1688 based. Gary . ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 190's vs Softphone
Additionally when I do receive the unreachable message as soon as I place an outbound call the peer becomes reachable.. dabigshiznizzle wrote: I have been playing with asterisk for a couple of weeks now and I have been very happy with its performance. However, I have run into a problem with how I want to deploy this solution. I have a mix of softphones (SJ and Xlite), ATA's, and a couple of IP phones (Snom 190). The asterisk box is on the public network. For my primary users they will reside behind a watchguard 4500 firewall. For the others there will be a mix of Ethernet routers (Linksys/Netgear/DLink). I have been testing the different deployments and have found that the softphones and ATA's work like a champ in getting around the firewall and the NAT of the work at home users. However the Snom phones don't perform as well. I have played with qualify statement in sip.conf and at first I thought that I might have been OK with not using that argument, but it didn't work. I continue to get the following: *CLI Feb 13 15:50:59 NOTICE[14043]: chan_sip.c:7971 sip_poke_noanswer: Peer '9093' is now UNREACHABLE! When this happens, I can't receive any calls (of course) but I can place outbound calls. Is this normal? Can anyone help me with this? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Who makes these phones?
On Mon, 2005-02-14 at 10:10, Gary wrote: On Mon, 14 Feb 2005 09:53:36 +1100, PHP Mechanic wrote: http://www.broadbandphone.com.au/global/pnp.htm They look like they are all PA1688 based. The black one is a dead copy of the one sitting on my desk, made by Hirakawa Electronics according to the label underneath. The middle white one looks similar - dunno out the other white one. ...and yes, they are PA1688 based. Gary . ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Melbourne Asterisk Users meet next Thursday
Should be a good night - looking forward to seeing some unfamiliar faces! Regards, PaulH -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jurgen Sent: Thursday, 10 February 2005 12:55 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Melbourne Asterisk Users meet next Thursday Hi all, If you're in Melbourne Australia and interested in Asterisk, you're invited to join us for a casual evening to talk about Asterisk, VOIP, networks, and just generally get geeky about IP phone stuff. Ultimately, I think it would be interesting and useful to turn this into a monthly get-together, so I'd like to talk about that too. Anyone with an interest is welcome; from Asterisk Gods to newbies who have recently downloaded it, from people administering several hundred seats to people playing with it at home and annoying their families. When: Next Thursday evening, the 17th, at 7pm. Where: Niagara Hotel, 383 Lonsdale Street (between Queen and Elizabeth) in the city. The Niagara's a relaxed, comfortable place. I'm going to try and get us a table, and put an old analogue phone on it, so you'll know how to find us. Any questions, you can reach me on 0415 276 127, or email [EMAIL PROTECTED] Hope to see you there! ...jurgen -- [EMAIL PROTECTED] is jurgen's gmail address. Visit http://jurgen.ca/ for more yummy goodness. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] connect asterisk to ISDN in China
Hi, Telecoms in China is not based on American standards. It is based on European standards. IDN in China is exactly the same as ISDN in Europe, and European configurations on Asterisk will work in China. Regards, Steve Xu, Duo wrote: Hi, I plan to install asterisk and connect it to telco through ISDN in China. I'd love to know if the ISDN standard in China has any difference than in America before I buy the digium card. anybody has experience in it? or anybody who installed asterisk with ISDN in asia can share their expierience? Or, can anybody give me some links to educate me ISDN knowledge about the difference in China? (My heard there is something different there, but i dont know the details.) Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM-400P alternatives?
You didn't say what your fxs/fxo requirements are but: A T1 card ($500) and a used channel bank ($300) might be a good alternative. You also might check out the voicetronix cards. Cheers, Jon. On Sunday 13 February 2005 02:55 pm, Paul Fielding wrote: Are there any other relatively low cost analog cards available? I'm interested in finding something that might work a bit more reliably than the TDM-400P regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM-400P Sound Quality issues
I spent a bunch of time troubleshooting the SIP end of things, thinking that's where the problem was, until I realized that every other SIP connection I make (from remote) yields a high quality call. ie. I can dial another SIP client and maintain high quality audio. Additionally, I can dial an extension that not only SIP connects to my server, but from there goes out an IAX2 connection to another remote Asterisk server, from there to another SIP client, and the audio quality is excellent. Therefore, I don't think the audio issue I'm experiencing is on the SIP end. Are there some wierd SIP - ZAP timing / conversion / other issues that could be causing this? thoughts? Pure guess... you're probably bumping into some of the same issues that many of us TDM users are hitting. Seems like either an interrupt handling (latency) or pci bus issue. You'll find hundreds of postings relative to this over the last six months or so. Not everyone has problems with the TDM, but some have found that swapping motherboards does clear up the issue. Processor speed and ram have nothing to do with it, nor does single vs dual processors, etc. Several people have opened trouble tickets with digium, but seems all have gone into a black hole (thus far). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] soho fax suggestions?
For planning purposes, is it appropriate to think in terms of purchasing a t38 capable box even if its not supported by * today? (I'm well aware of the bounty and Steve's work.) That's what I would do. In fact, I already have T.38 capable VOIP adapter (an Azatel 200) for my current fax machine but plan to upgrade that box to a Sipura 2100. If now is the time to purchase a t38 capable fax machine, anyone have any suggestions on a low-volume soho-sized box? I don't think there is such a thing as a T38 capable fax machine. T38 is for faxing over VOIP and I have yet to see a fax machine with a built-in network port so it can connect directly to the Internet...if you know what I mean. Well, I did find Okifax, Ricoh and Konica say they have it based on the web stuff. Don't have a clue whether they have delivered or even if it works. FWIW, I have had absolutely zero problems receiving faxes over VOIP, via Asterisk, using Voicepulse Connect, IAX trunking, and g.711. My problems for faxing are all related to outbound faxing (using the same Voicepulse setup or Sixtel iax.cc). Not sure why outbound is giving me problems when inbound isn't giving me any. shrug Of course I need to fax outbound more often than I need to receive inbound! Can't offer any clue on the above either. Based on Steve Underwood's comments earlier (relative to outbound fax now fails on the TDM when it was working earlier), it would almost sound like a timing issue of some sort that is associated with calls initiated within *. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PLEASE HELP Adit 600 went kaput?
For whatever it's worth, it was the crossover cable. --- Andrew Kohlsmith [EMAIL PROTECTED] wrote: On February 12, 2005 09:21 pm, David Coulson wrote: If he gets a green light with a loopback plug wired like that, his controller is definatly screwed up :-) 1-4 2-5 That was how I always learned to wire a loop plug anyway. You're absolutely right, I made a pretty big (and public) thinko... hahaha -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Meet the all-new My Yahoo! - Try it today! http://my.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA's
We have had a big success with the Linksys PAP2-NA. 2 FX ports and 1 WAN port. Only downside is that only 1 call can be using 729 at a time. This has been confirmed with Linksys. They will be releasing PAP2-NAv2 in March to overcome this. In the meantime, get a Sipura 2100, supports 2 729 calls and has both WAN/LAN ports. Personally, I dislike the lack of LEDs on the 2100. -Matthew - Original Message - From: Luki [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, February 13, 2005 12:39 PM Subject: Re: [Asterisk-Users] ATA's The Sipuras have a ton of configurable parameters. If you understand them (and there is no good manual, unfortunately) then you can be of great benefit. Otherwise they'll be worthless. I particularly miss the dial-plan, distinctive ring and audio gain options on the Grandstreams. Remote syslog can also be useful for debugging. It all depends what you need, I guess. Further, the Sipuras have a more detailed status, that is accessible WHILE you are engaged in a conversation. I think you're paying a bit more for the 1000 (1 line version) as compared to the Grandstream 286, but if you need/want two independent lines, then the Spa 2000 is more economical (as Peter said). --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Sangoma A102 cards testing
I had the same problems with Tormenta2 card from Digium. Same behaviour, both cards were receiving irq`s, but when spans got up, lots of messages (Bad FCS) came up too on my asterisk console... everything died with kernel panic in the end... The motherboard was an Asus with dual Pentium3 933MHz, distro was Debian SID with 2.6.10-1 kernel, asterisk 1.0.5. I also changed the motherboard with a classical intel P4 Board (an older one, GERG2) and it worked fine. Try booting up your kernel with noapic and nolapic parameters. That should disable the crappy IRQ routing through ACPI. Please let me know if it works for you, because i really miss my old Asus dual P3 configuration. :) Too bad i had to loose a whole night to figure it out it was the motherboard... i even reinstalled everything from the scratch too with no results... :( Calin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM-400P alternatives?
- Original Message - From: Jon Gabrielson [EMAIL PROTECTED] You didn't say what your fxs/fxo requirements are but: A T1 card ($500) and a used channel bank ($300) might be a good alternative. Basically my fxs/fxo requirements are the same as my existing TDM-400P ( 2 in 2 out). Just trying to find something that works more reliably than this card has turned out to be. Paul Cheers, Jon. On Sunday 13 February 2005 02:55 pm, Paul Fielding wrote: Are there any other relatively low cost analog cards available? I'm interested in finding something that might work a bit more reliably than the TDM-400P regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM-400P Sound Quality issues
Pure guess... you're probably bumping into some of the same issues that many of us TDM users are hitting. Seems like either an interrupt handling (latency) or pci bus issue. You'll find hundreds of postings relative to this over the last six months or so. Not everyone has problems with the TDM, but some have found that swapping motherboards does clear up the issue. I did a bunch of searching through the list, found lots of messages regarding misc. TDM400p problems, but none that sounded like the issue I'm seeing. Can anyone point me to any discussions regarding this? The thing that I find so odd about it is that the sound quality only degrades on the zap channel when I'm connecting from a *remote* SIP client, but on local network the zap channel sounds fine (see description below). I'm willing to get a different MB if that's really the fix, but I'd hate to go through the work and $$ to make that happen only to find that the problem doesn't go away... Paul - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, February 13, 2005 9:12 PM Subject: Re: [Asterisk-Users] TDM-400P Sound Quality issues I spent a bunch of time troubleshooting the SIP end of things, thinking that's where the problem was, until I realized that every other SIP connection I make (from remote) yields a high quality call. ie. I can dial another SIP client and maintain high quality audio. Additionally, I can dial an extension that not only SIP connects to my server, but from there goes out an IAX2 connection to another remote Asterisk server, from there to another SIP client, and the audio quality is excellent. Therefore, I don't think the audio issue I'm experiencing is on the SIP end. Are there some wierd SIP - ZAP timing / conversion / other issues that could be causing this? thoughts? Pure guess... you're probably bumping into some of the same issues that many of us TDM users are hitting. Seems like either an interrupt handling (latency) or pci bus issue. You'll find hundreds of postings relative to this over the last six months or so. Not everyone has problems with the TDM, but some have found that swapping motherboards does clear up the issue. Processor speed and ram have nothing to do with it, nor does single vs dual processors, etc. Several people have opened trouble tickets with digium, but seems all have gone into a black hole (thus far). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Debian way of compiling zaptel kernel modules
Tzafrir Cohen wrote: BTW: did I mention that we have binary packages for standard Debian Sarge kernels in our apt source? zaptel is the only package that never worked for me from apt-get. I need to download, compile and install the kernel (specially because the original debian install is pre 2.4.20), then download all the CVS (or whatever) files for asterisk and zaptel, compile-but-not-install the asterisk and then compile the zaptel. Not terrible, but not quite easy for a beginner. Or did I miss something? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intermediary jitter buffering
On Feb 12, 2005, at 9:10 PM, Michael Giagnocavo wrote: Hello, I understand that only the destination of a call should do jitter buffering. So, if IAX2/PhoneA calls IAX2/PhoneB through my server (no transfers), PhoneA and PhoneB need to perform their own jitter buffering, and Asterisk will just forward the frames, correct? What happens if the peer does not support jitter buffering, but is close by so there's no need for jitter buffering? My situation is that I have an Asterisk machine right in front of our provider's systems (same switch, 1ms latency). If they don't have jitter buffering, how can I force my Asterisk machine to jitter buffer calls from my users to them? Assuming this is all IAX, presently, the jitterbuffer is either on, or off, as you configure; it doesn't go off automatically if it's in the middle of a bridge (although native bridging does bypass it). So, in your situation, with the current code, disable native bridging, and enable the jitterbuffer, and you should get it. But, we're working on improving this area a lot; this is an uncommon situation, though: Why doesn't the peer support jitterbuffering? -SteveK ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Q: Does anyone have a WE multi-line card dialer phone working with *?
Folks, I recently obtained a Western Electric multi-line phone and am seeking help with getting this beast working with *. The interesting stuff in my * implementation consists of a T100P card, a TDM400P card, and an Adtran TA750 channel bank with three quad-port FXS modules and a quad-port FXO. The TA750 is wired to a 24-port Cat 5 patch panel via a 25-pair Amp cable. The phone is a model 2662A1M; it has five lines, a hold button (I presume), card dialer capability, and a 25-pair Amp cable for connecting to The Phone System. (The card dialer feature, IMHO, scores major geek points. If you're not familiar with it, you take a special plastic card about the size of a credit card and punch out two tiny discs for each digit in a phone number. When it's time to call that number, you insert the card in the phone, take the handset off hook, push the START button, and--voila!--the phone speed dials your party.) Each line in the phone uses three pairs in the Amp cable; the first pair is for ring and tip, the second pair is a mystery (I'm eagerly awaiting a copy of one of the phone's BSPs so I can find out), and the third pair illuminates the lamp in the button. Most of the remaining pairs in the Amp cable connect to one of the terminal boards inside the phone, and one pair connects to the phone's network (presumably for common ringing, since the leads connect to L1 and L2). If I were to connect the first pair of each line to the patch panel, I would have a perfectly serviceable five-line phone (I haven't yet tried the hold button). I would not have, however, illuminated buttons to indicate if channels were in use; nor would the phone ring on an incoming call. If I connect the first and third pairs of a line and plug that mess into a patch panel port, the lamp illuminates and the channel (according to the TA750 and *) goes off-hook--but I do not get a dial tone. I have not, BTW, performed any experiments with a port on the TDM400P. So . . . does anyone have any experience with such a project, or have any ideas on how to trick this up? Cheers, Rob ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7912G: Takes the same firmware as 7940/60?
On Sun, Feb 13, 2005 at 08:35:33PM +0100, Michiel van Baak arranged a set of bits into the following: On 20:08, Sun 13 Feb 05, Stefan Gofferje wrote: Michiel van Baak schrieb: Thnx. No luck for me I guess. chan_sccp it will be. Not for the 79[05|12]... At least my 7905 does not like chan_sccp too much and they crashed my * (1.0.5)... unless you bounty the chan_sccp developers for 79[05|12] support OR ask your local Cisco dealer for a Which while nice wouldn't necesserily help. For myself at least I can't afford one of each of the phones, and what I have comes from judicious eBaying. Loan of phones is much more usefull then anything else. That's about it. And from what I read the SIP image can really use the rest of the phone like speed dial, call forward etc. Speed dials should work, just configure them in sccp.conf, and most of the rest is under (slow) development. I actually have implementations of some of the features ready, they just need testing with soem more phones. Thanks, Julien chan_sccp developer pgpicLcbzz6pR.pgp Description: PGP signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Who makes these phones?
Message: 1 Date: Mon, 14 Feb 2005 09:53:36 +1100 From: PHP Mechanic [EMAIL PROTECTED] Subject: [Asterisk-Users] Who makes these phones? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; format=flowed; charset=iso-8859-1; reply-type=original http://www.broadbandphone.com.au/global/pnp.htm they are called a Kitty Ethernet Phone, seem to be available in 3 or 4 models but with identical Guts. The only info I have found on them is Gateway Technologies, supposedly the Chinese manufacturer website... http://www.ipgw.net/EN/index.htm I bought one off a guy who is flogging them in Au for about $90 each. Nice looking, cheap ip phone. But information manual are next to useless. The only technical info I have been able to find is the 8 page manual that comes with it (copy on website) which tells you nothing. I haven't yet tried it live, still working out how to set it up. Seems to have features like talking speed dial etc, but haven't yet worked out how to drive the functions and manual is less than helpful. Would appreciate if anybody has already managed to get one of these working and would like to share the setup and how to use the functions on them. Regards, Craig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
