Re: [Asterisk-Users] Reasons why I shouldn't use Asterisk?
On 6 Nov 2003, at 04:32, Tilghman Lesher wrote: OK, let me get this straight. Because the Asterisk voicemail menu is fault tolerant and lets you undo a delete, it's therefore unacceptable. I don't think the OP said it was unacceptable, just that it wasn't as configurable as they would like and they considered that a con. I can sympathise - the voicemail system is complicated. I can punch through all the messages in my mobile phone voicemail with three keys. 1 plays the message again, 2 saves it, 3 deletes it. If I save or delete a message it automatically advances to the next one. In comparison, the Asterisk voicemail program is a dog. Having complex functionality is fine as long as the basic functionality isn't made obscure. And before you accuse me of being unable to handle moderately complex systems as well. The point the OP was making is that it's not *configurable* not that it's too hard. If I choose to have a simpler system - or more importantly choose for all the users at an installation to have a simpler system - I can't do that. Jonathan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New here...
On 24/10/2003 6:28, TODD WALLACE - Mail Lists wrote: Can I get Asterisk up and going without voice cards using it with SoftPhones internally as a proof of concept. (just calling extensions and leaving voice mail) Is there a jump start config that would accomplish this? Yup. Others have already replied to this, but I'd add that I started out with an article from O'Reilly that explained pretty well how to get asterisk going: http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html?page=1 Jonathan -- Jonathan Hogg Director, Technology Seventh Wave Systems Ltd. 4-14 Tabernacle Street London EC2A 4LU Telephone: +44 20 7074 0423 http://www.seventh-wave-systems.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk behind NAT to SIP provider
Hi all, OK. I've tried trawling the archives, but I'm not getting very far. I've got an Asterisk box behind a NAT which I want to register with a SIP provider. In my sip.conf I have (edited to protect the innocent): - [general] port = 5060 bindaddr = 0.0.0.0 disallow = all allow = alaw allow = ulaw allow = gsm context = bogus-calls tos = lowdelay nat = yes register = 8703405315:[EMAIL PROTECTED] [8703405315] type = friend reinvite = no canreinvite = no nat = yes username = 8703405315 secret = context = from-sip-provider - With 'sip debug' on, I can see it sending the REGISTER requests and getting back a response with STUN headers like so (also edited): - SIP/2.0 407 Proxy Authorization Required X-Stun-Server: w.x.y.z:3478 X-Observed-Adr: a.b.c.d ... - However, when Asterisk sends the auth it doesn't sends the REGISTER again to the same address without seeming to take into account the STUN details, a la: - REGISTER sip:sip-provider.not SIP/2.0 Via: SIP/2.0/UDP 10.20.15.4:5060;branch=z9hG4bK43e3ead5 ... Contact: sip:[EMAIL PROTECTED] ... - This results in me getting a 406 Bad Contact (NAT) response. My questions: a) Does Asterisk support what I want to do (please don't tell me to use IAX instead - I am already talking to the provider about that, but they are in the early stages of playing with Asterisk)? b) What have I done wrong in my sip.conf? I've been hacking it around for a while this afternoon so it's a bit of a mess of mangled attempts to make it work. Any help gratefully appreciated. Jonathan -- Jonathan Hogg Director, Technology Seventh Wave Systems Ltd. 4-14 Tabernacle Street London EC2A 4LU Telephone: +44 20 7074 0423 http://www.seventh-wave-systems.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Festival on RH9?
On 24/10/2003 15:46, Rich Adamson wrote: and still get the same error msg in /etc/asterisk/festival_server.log: client(5) Fri Oct 24 08:50:21 2003 : rejected from phoenix.routers.com not in access list every time I dial the extension that festival is supposed to say Testing The netstat -an indicates festival is listening on port 1314 (which was started before asterisk). Anyone have any ideas, or, does anyone have a RH9 system working with festival that could take a look at the contents of /etc/hosts.allow for me? Isn't the access list something that is passed to 'festival --server' in a config by the 'festival_server' script? I'm not in front of festival at the moment, but you might want to try tracing through that script and see what it's doing. Jonathan -- Jonathan Hogg Director, Technology Seventh Wave Systems Ltd. 4-14 Tabernacle Street London EC2A 4LU Telephone: +44 20 7074 0423 http://www.seventh-wave-systems.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind NAT to SIP provider
On 24/10/2003 16:05, Olle E. Johansson wrote: If you've travelled around the archives, you should now that this is a FAQ. I'm sure it is, but in the absence of a FAQ on the Asterisk website, this is a little hard for new users to determine. At this moment, Asterisk behind a NAT can't connect to an outside SIP provider. If you put asterisk outside your NAT, your inside clients can connect to Asterisk and Asterisk will be able to connect to your providers. I suspected this would be the case. The problem is that I have no control over the NAT. I guess I'll just have to work on my provider a bit more to support IAX. There are bug reports, web pages and mail in the archive that document this. Start at http://www.voip-info.org - click on Asterisk. I could find plenty of emails from people asking how to do it, but I couldn't find any answers - odd that Google only seems to show the questions, but not the answers. Thanks for the link and the help. Jonathan -- Jonathan Hogg Director, Technology Seventh Wave Systems Ltd. 4-14 Tabernacle Street London EC2A 4LU Telephone: +44 20 7074 0423 http://www.seventh-wave-systems.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind NAT to SIP provider
On 24/10/2003 19:31, rnc Info Lists wrote: I have the same problem and have solved it by using iaxtel.com. Asterisk talks to IAXtel quite well on inbound and outbound from behind my NAT router. Yeah, I got that working as a test that Asterisk could successfully route calls in and out to my extensions, but I need a PSTN gateway service that can offer numbers in London and NY. I'm talking to a UK provider, but they only do SIP at the moment. I'm working with one of their tech guys to see if they can support IAX via an Asterisk installation at their end. Jonathan -- Jonathan Hogg Director, Technology Seventh Wave Systems Ltd. 4-14 Tabernacle Street London EC2A 4LU Telephone: +44 20 7074 0423 http://www.seventh-wave-systems.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with * and IAXTel/FWD
On 23/10/2003 12:38, David J Carter wrote: I have tried to call 1800 and the * says that a connection to IAXTEL is made but I get no ringing or anything from the remote end. Does anyone have a 1700XXX number I can call, or can somebody call mine, 17008188820. Hey likewise. I can only seem to ring myself (1 700 873 7731). I just tried you and got nothing. Jonathan -- Jonathan Hogg Director, Technology Seventh Wave Systems Ltd. 4-14 Tabernacle Street London EC2A 4LU Telephone: +44 20 7074 0423 http://www.seventh-wave-systems.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Festival on RH9?
On 23/10/2003 21:16, Rich Adamson wrote: I'm about to download Festival source, apply the astrisk diff's, and initiate basic testing. Thoughts are to download v1.4.3 (latest per the fesitval website. If anyone has an existing how-to, install notes, tips, or any suggestions I'd greatly appreciate it. Direct email is fine if you'd rather not post them. I just did exactly that and it was pretty straightforward. My method: * Download festival, speechtools, festlex_OALD, festlex_POSLEX, and festvox_don. * Unpack all of the above into a new directory. * cd into speech tools: ./configure make * cd into festival: patch -p1 .../festival-1.4.3.diff ./configure make * Add $PWD/bin to PATH * Run festival_server Seems to work fine. Jonathan -- Jonathan Hogg Director, Technology Seventh Wave Systems Ltd. 4-14 Tabernacle Street London EC2A 4LU Telephone: +44 20 7074 0423 http://www.seventh-wave-systems.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Defragmenting mailboxes
On 22/10/2003 1:42, Chris Albertson wrote: IMAP would work as would an NFS mounted maildir. But I still would prefer a DBMS based store to support some voicemail features such as. [...] For all of the above a simple SQL query would do the trick The user interface could be either menues built in extensin.conf or web based. or a Java ap using JDBC. The DBMS method would scale to population of a large city. Alayerd design the allows for various types of stores would be best as a DBMS for a home office is silly. Yes, requiring a RDBMS for voicemail would be a pain for a lot of us, but I like the idea of using an email storage backend that can be configured. Perhaps the University of Washington c-client library, which can have different backends plugged into it and provides mbox and maildir (via a 3rd party module/patch), or something similar. The idea of storing voicemails as audio/wav attachments to mails in a maildir appeals to me for running an IMAP server on top of it and providing easy remote access to voicemail. Jonathan -- Jonathan Hogg Director, Technology Seventh Wave Systems Ltd. 4-14 Tabernacle Street London EC2A 4LU Telephone: +44 20 7074 0423 http://www.seventh-wave-systems.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On 21/10/2003 11:14, Andrew Kohlsmith wrote: [...] 6 - POE (12V-48V input range) [...] 5 - integrated 100mbit switch ***capable of sustaining 100mbit*** [...] +1 on both of these points. The power brick is cheap and nasty. POE would be a huge plus. A 100mb bridge would make the phone a lot more attractive in an office full of cables. I'd also add my voice to the request for a better speakerphone. The dialtone comes out loud and clear but everything else is too muted. If I up the volume to hear calls, then the dialtone becomes deafening - as does the handset when used. I'm less concerned about the codecs as I'm happy to use ULAW/ALAW on the internal network and have Asterisk transcode to something else for external calls. There should be a way of locking the menu button, as it is too easy to muck with the settings. For central configuration, the cfg.txt file format would be nice, but is still a pain. Ideally I'd like to be able to configure the phone via DHCP extensions. That would be ideal as I can configure the lease time to manage how frequently the phones update and I can centralise the configuration with the IP details. Jonathan -- Jonathan Hogg Director, Technology Seventh Wave Systems Ltd. 4-14 Tabernacle Street London EC2A 4LU Telephone: +44 20 7074 0423 http://www.seventh-wave-systems.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
