[Asterisk-Users] Re: Google newsgroup or Forum setup.
Ryan Tucker <[EMAIL PROTECTED]> said: >I will counter with problems that web-based forums have: > And I agree. >I could perhaps go for the Usenet method of doing things. In fact, an >ideal solution would be an e-mail to Usenet gateway. > http://www.gmane.org/ - I have subscribed to the list with 'nomail' switched on and I'm reading this with my favorite newsreader, trn. Like most of the mailing lists I'm on :-) news://news.gmane.org/gmane.comp.telephony.pbx.asterisk.user -- Cees de Groot http://www.cdegroot.com <[EMAIL PROTECTED]> GnuPG 1024D/E0989E8B 0016 F679 F38D 5946 4ECD 1986 F303 937F E098 9E8B Cogito ergo evigilo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP security (was: New ATA clone out)
was breezing over http://voxilla.com/ Looks like a new ATA from the founder of Komodo Technology (aka the Cisco 186) Sipura SPA 2000 http://www.sipura.com/products/spa2000.htm to join the others Cisco ATA 186/188 http://www.cisco.com/warp/public/cc/pd/as/180/186/ 8x8 DTA-310 http://www.8x8.com/products/home_office/dta-310/index.asp.html Grandstream HandyTone 286 http://www.grandstream.com/y-product.htm Note that the Sipura supports TLS and SSL/HTTPS for what I assume must be security for SIP messages. They say "media encryption", but I don't see any mention of SRTP. Anybody want to start churning on getting some support for TLS into Asterisk? Even TCP doesn't exist yet (see the bugtracker for comments on an outstanding request for TCP support.) I really want to have more security in SIP, especially now that I have an 802.11 phone. I understand very clearly that POTS lines are just as easy to tap as my home phone, but I have a real desire to have MORE security on my SIP lines than on my POTs lines. Since this is a jumbled post to start with, let's add some more junk: I have an 802.11 phone, from pulver.com. It doesn't work with Asterisk, period. I've seen them work with FWD and the Jasomi box that they run in front of that service, but I haven't been able to convince it to work with * in any way other than getting signalling out from the phone (the phone seems to be ignoring the SIP reply packets under some circumstances, though it REGISTERs just fine.) JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with PHPconfig setup??
Dave Packham wrote: in the phpconfig_init.php you need to make sure that the files paths are correct p0lar Thanks, That worked.. I had to put the full path to the asterisk.reload file.. I thought it would be somthing simple.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] configuration
costas wrote: Is everything with Asterisk setup using configuration files or is there a GUI type of an interface that can be used? Thanks Costas Yes its all in the .conf files.. There is phpconfig which will allow you to edit your config files in a web browser, but you are still editing the config files.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] tdm400p + x100p config problem
Hi, I have configured tdm400p alone and it works fine for me. But now i tried to add 2 x100p in the same machine with the following configs, but asterisk refuses to load, can someone guide me what i have done wrong. Following r the configs and output. TIA, Azher >> ./asterisk -vc [chan_zap.so] => (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, FXS Kewlstart signalling -- Registered channel 2, FXS Kewlstart signallingWARNING[16384]: File chan_zap.c, Line 600 (zt_open): Unable to specify channel 3: No such deviceERROR[16384]: File chan_zap.c, Line 4909 (mkintf): Unable to open channel 3: No such devicehere = 0, tmp->channel = 0, channel = 3ERROR[16384]: File chan_zap.c, Line 6663 (load_module): Unable to register channel '3'WARNING[16384]: File loader.c, Line 301 (ast_load_resource): chan_zap.so: load_module failed, returning -1WARNING[16384]: File loader.c, Line 396 (load_modules): Loading module chan_zap.so failed! >> Zaptel.conf loadzone = usdefaultzone=usfxsks=1,2fxoks=3,4 >> zapata.conf group=1context=twowaycallgroup=1pickupgroup=1immediate=no busydetect=yes;callprogress=yes;musiconhold=default; for dialin services signalling = fxs_kschannel => 1,2 group=2context=cso; to forward calls to staff connected with tdm400p signalling = fxo_kschannel => 3,4 >>> ztcfg -vvv Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01)Channel 02: FXS Kewlstart (Default) (Slaves: 02)Channel 03: FXO Kewlstart (Default) (Slaves: 03)Channel 04: FXO Kewlstart (Default) (Slaves: 04) 4 channels configured. Do you Yahoo!? The New Yahoo! Shopping - with improved product search
Re: [Asterisk-Users] configuring TE410P for four E1 PRI lines
Thanks everyone. I changed the jumper settings and it worked fine. Leo Ann Boon wrote: Rahul Arvind Jadhav wrote: hi, I'm trying to configure my newly acquired TE410P card to work as 4 E1 spans. This is supposed to be a drop-in replacement to the earlier E100P card. However, on loading the zaptel module it gets configured as T1 spans basically doing a 'cat' on /proc/zaptel/1 thru 4, it shows 24 channels per span. After this ztcfg fails saying 'ZT_CHANCONFIG failed for channel 97'. This is obviously because Quad T1 has only 96 channels. The question is how do i get the zaptel module to recognize the card as Quad E1 and not Quad T1. My current /etc/zaptel.conf goes as follows: --- span=1,0,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 span=3,0,0,ccs,hdb3,crc4 span=4,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 bchan=63-77,79-93 dchan=78 bchan=94-108,110-124 dchan=109 loadzone = us defaultzone = us --EOF--- Thanks, Rahul You need to set the jumpers on the card. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Rahul Arvind Jadhav Sr. Software Engineer NetYantra India Pvt. Ltd -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] number detection problem.
Thanks for all of the suggestions. There is no noticable static on the channels and I'm using the tor card as primary timing source... I'm now trying relaxdtmf and turning the gain way down. Since it's an intermittant problem I won't know for a while. Thanks again. Jon Steven Critchfield <[EMAIL PROTECTED]> wrote .. > On Fri, 2003-09-26 at 15:09, Jon Hopper wrote: > > Hello, > > > > We're using asterisk with a z-plex 10 and a Zap tormentia card at a non-profit > home for the aged. > > > > Sometimes when we dial out, our numbers are misintepreted with one number > being detected twice. For instance 337-4411 becomes 337-7441 7's 4's and > 0's seem to be particularly prone to this issue. > > > > Any ideas? I've tried turning down rxgain and txgain. > > > > A little debugging info: > > > > # uname -a > > Linux pbx 2.4.18 #1 Mon Aug 25 14:27:34 EDT 2003 i686 unknown > > Debian system > > 1.2Ghz Athalon Thunderbird > > Asterisk CVS-08/25/03-14:35:09, Copyright (C) 1999-2001 Linux Support > Services, Inc. > > Do you hear a little static during the dialtone? Do you sometimes hear > extra noise as you add on more active channels? If so it has to do with > timing problems and slips. Asterisk will detect a DTMF then a static pop > will signal the end of the DTMF, then asterisk hears the DTMF again. > -- > Steven Critchfield <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outgoing call spool
Andrew Joakimsen wrote: Post the line with the Dial() from your extensions.conf Do you mean post it here? exten => _[1-9]XX,206,dial,sip/BYEXTENSION|30| Bill Im not sure that it will work, but its worth a try. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FYI-New ATA clone out
DTA310 does not count because I cannot get it to function properly (as well as another member on this list) > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of TC > Sent: Sunday, September 28, 2003 11:06 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] FYI-New ATA clone out > > was breezing over http://voxilla.com/ > Looks like a new ATA from the founder of Komodo Technology > (aka the Cisco 186) > Sipura SPA 2000 http://www.sipura.com/products/spa2000.htm > > to join the others > Cisco ATA 186/188 http://www.cisco.com/warp/public/cc/pd/as/180/186/ > 8x8 DTA-310 http://www.8x8.com/products/home_office/dta-310/index.asp.html > Grandstream HandyTone 286 http://www.grandstream.com/y-product.htm > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FYI-New ATA clone out
was breezing over http://voxilla.com/ Looks like a new ATA from the founder of Komodo Technology (aka the Cisco 186) Sipura SPA 2000 http://www.sipura.com/products/spa2000.htm to join the others Cisco ATA 186/188 http://www.cisco.com/warp/public/cc/pd/as/180/186/ 8x8 DTA-310 http://www.8x8.com/products/home_office/dta-310/index.asp.html Grandstream HandyTone 286 http://www.grandstream.com/y-product.htm ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR Web Search Frontend
Java has the best solution to this As long as the schema stays the same (and you don't use vendor-specific sql stuff) you can change databases by simply using a different database URL. It's literally that easy... We've been moving some java reports from our AS/400 to an oracle database server, and you change one configuration line and bingo! -G - Original Message - From: "Andrew Kohlsmith" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, September 27, 2003 8:13 AM Subject: Re: [Asterisk-Users] CDR Web Search Frontend > > Since * and MySQL have had a licensing scuffle, is there a way to set it > > up so that we can specify wether or not it's in the mysql database, or > > use the plaintext file that * generates with cdr_csv.so? > > Or do something really smart like the Perl guys and have a > backend-mostly-independent DB infrastructure. Hell I think that PHP > finally smartened up and went this way, too. > > Regards, > Andrew > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Google newsgroup or Forum setup.
On Sun, 2003-09-28 at 16:12, costas wrote: > I am sure this has been asked before, but why not use Google newsgroup > or at least some forum BBS software instead of this cumbersome mailing > list process? Because bbs software requires net connection at the time of browsing. Because newsgroups are basically email with central IMAP server, and newsgroups don't have access control to handle stupid spammers. Mailing lists also allow any individual that is subscribed create their own archive. Mailing lists also are able to be read offline at the users convenience. Mailing lists can be easily read on any number of alternative devices. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Outgoing call spool
Post the line with the Dial() from your extensions.conf Im not sure that it will work, but its worth a try. > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Bill Leckey > Sent: Sunday, September 28, 2003 8:45 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Outgoing call spool > > Andrew Joakimsen wrote: > > No, because asterisk cannot deal with the G723 codec, it can only act as > > a "middle man" of sorts between devices that support it. > > Ok, that makes sense. Could I get the ringing somehow if I changed to > (say) the G711 codec? > > Or, is it possible that this could be done by (say) the SIP RINGING > message? I believe that while the remote phone is being rung then the > originating call is currently in a "call up" state, which means a SIP > RINGING isn't allowed, but I guess I'm wondering if something like this > might work? > > Thanks, > Bill > > > > > > > > > > >>-Original Message- > >>From: [EMAIL PROTECTED] [mailto:asterisk-users- > >>[EMAIL PROTECTED] On Behalf Of Bill Leckey > >>Sent: Sunday, September 28, 2003 7:03 PM > >>To: [EMAIL PROTECTED] > >>Subject: [Asterisk-Users] Outgoing call spool > >> > >>I've been playing with the outgoing call spooling feature a bit lately > >>and it all works as it should with the exception of one irritation. > >> > >>I'm mostly using SIP to talk to the phones and using G.723.1 > >> > >>I copy the call file into the spool/outgoing directory and the > >>originating phone rings. I pick it up and the remote phone rings. > >>However there is dead silence from the originating earpiece. Is it > >>possible to somehow generate a ring in the earpiece until the remote > >>phone is picked up? > >> > >>Bill > >> > >>-- > >> > >>___ > >>Asterisk-Users mailing list > >>[EMAIL PROTECTED] > >>http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > . > > > > > -- > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outgoing call spool
Andrew Joakimsen wrote: No, because asterisk cannot deal with the G723 codec, it can only act as a "middle man" of sorts between devices that support it. Ok, that makes sense. Could I get the ringing somehow if I changed to (say) the G711 codec? Or, is it possible that this could be done by (say) the SIP RINGING message? I believe that while the remote phone is being rung then the originating call is currently in a "call up" state, which means a SIP RINGING isn't allowed, but I guess I'm wondering if something like this might work? Thanks, Bill -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Bill Leckey Sent: Sunday, September 28, 2003 7:03 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Outgoing call spool I've been playing with the outgoing call spooling feature a bit lately and it all works as it should with the exception of one irritation. I'm mostly using SIP to talk to the phones and using G.723.1 I copy the call file into the spool/outgoing directory and the originating phone rings. I pick it up and the remote phone rings. However there is dead silence from the originating earpiece. Is it possible to somehow generate a ring in the earpiece until the remote phone is picked up? Bill -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users . -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Forwarding SIP over IAX problem: No One Available
I'm hoping someone can help me out with this. I am basically just
trying to separate my outbound and inbound calls into separate contexts
instead of having everything in a single context. Any help would be
appreciated. Perhaps I've missed something really obvious
Here is the network layout:
<--TDM400P--> <--IAX--> <--SIP-->
This is the error I am getting when someone trys to call be via FWD
(all outbound calling works fine, and inbound has worked before as long
as I have everything in the same context, but I am trying to seperate
the contexts like in my previous dialplan):
Executing Dial("SIP/fwd.pulver.com2-312e",
"IAX2/madsen:[EMAIL PROTECTED]/[EMAIL PROTECTED]") in new stack
-- Called madsen:[EMAIL PROTECTED]/[EMAIL PROTECTED]
-- Hungup 'IAX2[207.61.247.201:4569]/1'
== No one is available to answer at this time
And here are the config files:
*** NAT'd Box ***
extensions.conf
---
[globals]
PHONE1=Zap/1
PHONE1VM=18924
CALLFILENAME=foo
FOO=foo
[intern]
include => outbound-fwd
include => from-sip
[outbound-fwd]
exten => _7.,1,Dial(IAX2/madsen:[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten => _7.,2,DISA,no-password|intern
[from-sip]
exten => 18924,1,Dial(${PHONE1},30,t)
exten => 18924,2,Voicemail(u${PHONE1VM})
exten => 18924,3,Hangup
exten => 18924,102,Voicemail(b${PHONE1VM})
exten => 18924,103,Hangup
exten => 5,1,Dial(${PHONE1},15,t)
exten => 5,2,Hangup
iax.conf
[general]
port=5036
bindaddr=0.0.0.0
register => madsen:[EMAIL PROTECTED]
[madsen]
type=friend
username=madsen
secret=password
auth=md5
context=intern
host=dynamic
In zapata.conf, the context=intern
*** GW Box ***
extensions.conf
---
[globals]
FWDUSERID=18924
FWDUSERNAME=Leif Madsen
CALLFILENAME=foo
FOO=foo
[intern]
include => outbound-fwd
include => from-sip
[outbound-fwd]
exten => _7.,1,SetCallerID(${FWDUSERID})
exten => _7.,2,SetCIDName(${FWDUSERNAME})
exten => _7.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten => _7.,4,Playback(invalid)
[from-sip]
exten => 18924,1,Dial(IAX2/madsen:[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten => 5,1,Dial(IAX2/madsen:[EMAIL PROTECTED]/[EMAIL PROTECTED])
iax.conf
[general]
port=5036
bindaddr=0.0.0.0
register => madsen:[EMAIL PROTECTED]
[madsen]
type=friend
username=madsen
secret=password
auth=md5
context=intern
host=dynamic
sip.conf
[general]
disallow=all
allow=gsm
allow=ulaw
allow=alaw
context=intern
maxexpirey=180
defaultexpirey=160
tos=reliability
register=18924:[EMAIL PROTECTED]/18924
register=5:[EMAIL PROTECTED]/5
[fwd.pulver.com]
type=friend
secret=password
username=18924
host=fwd.pulver.com
[fwd.pulver.com2]
type=friend
secret=password
username=5
host=fwd.pulver.com
In zapata.conf the context=intern
Thanks,
Leif Madsen.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Google newsgroup or Forum setup.
I prefer the list as it is. Text based, not GUI'd to death Faster is a relative term. A horse can get you there faster than a car... On Sun, Sep 28, 2003 at 06:03:43PM -0400, costas wrote: > I think back in the startup days mailing lists did the job but, given the advances > that have been made to bbs software it would beehove using them. > > Here is some issues with emails that bbs software solve. > > 0) You need email software. Why clutter everyones email boxes with thousands of > messages. > 1) Not everyone uses outlook or email clients. (I use web based email exclusively) > 2) Since I don't have all of the messages from day one, I have to download all the > files and keep them somewhere where I can remember. Any of my associates have to do > the same. > 3) Its hard to search for something unless I download the archives and search a > humoungs files and then open them. > 4) I prefer forums like the phpBB. They have so many cool things that can be setup > for users. Easy to organize, easy to find things, give incentives and recognition to > users. Add links to all the websites. > > I guess in the end its the advantage of an automobile over a horse buggy. They both > do the same thing. Just one is faster and has more features than the other. > > Costas > > -- Original Message -- > From: "Paul Crick" <[EMAIL PROTECTED]> > Reply-To: [EMAIL PROTECTED] > Date: Sun, 28 Sep 2003 14:32:03 -0700 > > >> I am sure this has been asked before, but why not use > >> Google newsgroup or at least some forum BBS software > >> instead of this cumbersome mailing list process? > >Someone's wanting to get flamed today eh? ;-) > > > >I'm a big fan of bulletin board type systems and have used phpBB quite a bit > >(www.phpbb.com) - it's great because it has a "show me all new messages > >since I last logged in" feature and allows for multiple forums/topics etc. > >You can get it to email you if someone replies to a thread that you've > >marked/are "watching". > > > >That said, I like the Asterisk mailing lists too - they end up in different > >folders in my Outlook, I can follow threads etc, it works well too. > > > >Six of one, half a dozen of the other I guess - everyone has their personal > >preferences. As for "cumbersome" - hmm, that's the word that'll maybe rock > >the boat on the list. What's the problem with a mailing list? So long as > >people can handle the basics of reply quoting etiquette and keep away from > >HTML formatting, there should be few problems? I guess the only thing is the > >search capabilities but then you can do that through Google or another > >archive indexer - it's just maybe not that obvious/clear how to do it.. > > > >For list reference/archiving, go to www.google.com, in the search keywords > >box type "site:lists.digium.com" followed by the keywords you want to > >search. > > > >Anyone else got thoughts/evidence on email versus BBS type working? > > > >___ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > Costas Menico > Meezon Software Corp > 201-224-8111 > [EMAIL PROTECTED] > > -- > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outgoing call spool
On Mon, 29 Sep 2003, Bill Leckey wrote: > I've been playing with the outgoing call spooling feature a bit lately > and it all works as it should with the exception of one irritation. > > I'm mostly using SIP to talk to the phones and using G.723.1 > > I copy the call file into the spool/outgoing directory and the > originating phone rings. I pick it up and the remote phone rings. > However there is dead silence from the originating earpiece. Is it > possible to somehow generate a ring in the earpiece until the remote > phone is picked up? Asterisk can neither originate or terminate calls using G.723. You'll need to use another codec. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Outgoing call spool
No, because asterisk cannot deal with the G723 codec, it can only act as a "middle man" of sorts between devices that support it. > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Bill Leckey > Sent: Sunday, September 28, 2003 7:03 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Outgoing call spool > > I've been playing with the outgoing call spooling feature a bit lately > and it all works as it should with the exception of one irritation. > > I'm mostly using SIP to talk to the phones and using G.723.1 > > I copy the call file into the spool/outgoing directory and the > originating phone rings. I pick it up and the remote phone rings. > However there is dead silence from the originating earpiece. Is it > possible to somehow generate a ring in the earpiece until the remote > phone is picked up? > > Bill > > -- > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NAT/SIP solution?
I had been warned about British sense of humour, but this even a South American like myself find funny. Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of WipeOut Sent: Sunday, September 28, 2003 3:21 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] NAT/SIP solution? Stig Hess wrote: > I meant where Asterisk is behing a NAT... sorry for the confusion. > > Regards, > > Stig H. Oh.. :) Well thats a bigger problem.. and i doubt the "Gods of SIP" are going to fix it any time soon.. :( Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outgoing call spool
I've been playing with the outgoing call spooling feature a bit lately and it all works as it should with the exception of one irritation. I'm mostly using SIP to talk to the phones and using G.723.1 I copy the call file into the spool/outgoing directory and the originating phone rings. I pick it up and the remote phone rings. However there is dead silence from the originating earpiece. Is it possible to somehow generate a ring in the earpiece until the remote phone is picked up? Bill -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with PHPconfig setup??
in the phpconfig_init.php you need to make sure that the files paths are correct p0lar >>> [EMAIL PROTECTED] 9/28/2003 1:17:28 PM >>> Hi, Just giving phpconfig a try but can't find and setup instructions.. What I have done so far.. 1. Copied the phpconfig files to the web dir on the server. 2. Edited the manager.conf and enabled manager access and setup an accound (really just copied the sample config but set my own username and secret) 3. Made /etc/asterisk/* world writable. (I guess I should have made the owned by apache and then user writable but its a dev server so I am not to worried about that) 4. Edited phpconfig_init.php with the username from manager.conf. 5. Edited asterisk.reload and changed the values for Username: and Secret: 6. Made asterisk.reload executable. So far all seems to be working except for "Re-Read Configs".. I get "Reset failed." when I try.. So what have I left out??.. If I execute asterisk.reload in a shell ir works fine so it must be somthing to do with calling it from phpconfig.. (I am using a Linux desktop and Mozilla 1.4 but that shouldnt make any difference.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: asterisk-u] Google newsgroup or Forum setup.
On Sun, 28 Sep 2003, costas wrote: > I think back in the startup days mailing lists did the job but, given the advances > that have been made to bbs software it would beehove using them. ... of course your web based email client has chosed to disregard linewrapping ... > Here is some issues with emails that bbs software solve. > > 0) You need email software. Why clutter everyones email > boxes with thousands of messages. What you are describing is push, vs pull access to content. I would rather have potentially interesting new content to appear in my unified message interface, duly stripped of malicious content (think i-frame exploit in a post forever trapped in a web archive) with full management capability, rather than losing a 'bookmark'. Also, incrementally compiling an archive of messages for review and search is one; offline mail reading (when travelling or on a bech); not having to keep a metered internet connection live. -- Russ Herrold ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Google newsgroup or Forum setup.
On Sun, 28 Sep 2003, costas wrote: > I think back in the startup days mailing lists did the job but, given the advances > that have been made to bbs software it would beehove using them. > > Here is some issues with emails that bbs software solve. [...] I will counter with problems that web-based forums have: 1) A connection to the Internet is required. Oftentimes, especially with dialup situations and while travelling, it's difficult to be online enough to read web-based forums. A good, high-traffic list like asterisk-users is great for long flights or immigration queues. 2) You need web browser software. While not that big of a problem these days, it still means firing up Opera and using that. One could use Lynx, I suppose. 3) Those f***ing smileys. 4) Limited search/archival options. With this list, there's a couple different sets of archives out there, which get indexed regularly by Google. As long as people use accurate Subject: lines and proper threading protocol, all goes quite well. 5) Sloow. When you're dealing with hundreds of messages per day, having to wait a couple of seconds for the next message to come up is unacceptable. I could perhaps go for the Usenet method of doing things. In fact, an ideal solution would be an e-mail to Usenet gateway. To keep spam down, it could be one way -- to post, you'd e-mail in the response instead of posting it to the newsgroup. In addition to sending the message to all who want the e-mail, it'd go out NNTP-style and end up on Google Groups, for the web interfaceness. DISCLAIMER: I've tried following web-based forums, but in general, I've found myself unable to do so. As such, I lean very heavily against moving mailing lists to web-based forums. I also usually use e-mail software which is quite good at handling busy mailing lists, so I've grown lazy. :-) -rt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NAT/SIP solution?
On Mon, 2003-09-29 at 00:00, Eric Wieling wrote: > There are several SIP aware NAT routers. Any Cisco router with a > firewall load has SIP aware NAT. There is at least one other brand of > SIP aware NAT router out there, but I don't recall the brand. > > On Sun, 2003-09-28 at 17:26, Uriel Carrasquilla wrote: > > Nikotel has a solution and one participant in thi list is doing a trial on a > > SIP/NAT router (claiming to be the first one in this realm). http://www.intertex.se -- Dave Cotton <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Google newsgroup or Forum setup.
I would say that the advantage of an e-mail list is that the reader can decide which mail reader they want to use rather than being forced into what ever interface the web site wants them to use. I, for one, would not be on the mailing list if I did not have the great filtering and sorting features of my e-mail client (Evolution). It's just SO much faster than trying to access some web based discussion board. --Eric On Sun, 2003-09-28 at 17:03, costas wrote: > I think back in the startup days mailing lists did the job but, given the advances > that have been made to bbs software it would beehove using them. > > Here is some issues with emails that bbs software solve. > > 0) You need email software. Why clutter everyones email boxes with thousands of > messages. > 1) Not everyone uses outlook or email clients. (I use web based email exclusively) > 2) Since I don't have all of the messages from day one, I have to download all the > files and keep them somewhere where I can remember. Any of my associates have to do > the same. > 3) Its hard to search for something unless I download the archives and search a > humoungs files and then open them. > 4) I prefer forums like the phpBB. They have so many cool things that can be setup > for users. Easy to organize, easy to find things, give incentives and recognition to > users. Add links to all the websites. > > I guess in the end its the advantage of an automobile over a horse buggy. They both > do the same thing. Just one is faster and has more features than the other. > > Costas > > -- Original Message -- > From: "Paul Crick" <[EMAIL PROTECTED]> > Reply-To: [EMAIL PROTECTED] > Date: Sun, 28 Sep 2003 14:32:03 -0700 > > >> I am sure this has been asked before, but why not use > >> Google newsgroup or at least some forum BBS software > >> instead of this cumbersome mailing list process? > >Someone's wanting to get flamed today eh? ;-) > > > >I'm a big fan of bulletin board type systems and have used phpBB quite a bit > >(www.phpbb.com) - it's great because it has a "show me all new messages > >since I last logged in" feature and allows for multiple forums/topics etc. > >You can get it to email you if someone replies to a thread that you've > >marked/are "watching". > > > >That said, I like the Asterisk mailing lists too - they end up in different > >folders in my Outlook, I can follow threads etc, it works well too. > > > >Six of one, half a dozen of the other I guess - everyone has their personal > >preferences. As for "cumbersome" - hmm, that's the word that'll maybe rock > >the boat on the list. What's the problem with a mailing list? So long as > >people can handle the basics of reply quoting etiquette and keep away from > >HTML formatting, there should be few problems? I guess the only thing is the > >search capabilities but then you can do that through Google or another > >archive indexer - it's just maybe not that obvious/clear how to do it.. > > > >For list reference/archiving, go to www.google.com, in the search keywords > >box type "site:lists.digium.com" followed by the keywords you want to > >search. > > > >Anyone else got thoughts/evidence on email versus BBS type working? > > > >___ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > Costas Menico > Meezon Software Corp > 201-224-8111 > [EMAIL PROTECTED] > > -- > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Google newsgroup or Forum setup.
I think back in the startup days mailing lists did the job but, given the advances that have been made to bbs software it would beehove using them. Here is some issues with emails that bbs software solve. 0) You need email software. Why clutter everyones email boxes with thousands of messages. 1) Not everyone uses outlook or email clients. (I use web based email exclusively) 2) Since I don't have all of the messages from day one, I have to download all the files and keep them somewhere where I can remember. Any of my associates have to do the same. 3) Its hard to search for something unless I download the archives and search a humoungs files and then open them. 4) I prefer forums like the phpBB. They have so many cool things that can be setup for users. Easy to organize, easy to find things, give incentives and recognition to users. Add links to all the websites. I guess in the end its the advantage of an automobile over a horse buggy. They both do the same thing. Just one is faster and has more features than the other. Costas -- Original Message -- From: "Paul Crick" <[EMAIL PROTECTED]> Reply-To: [EMAIL PROTECTED] Date: Sun, 28 Sep 2003 14:32:03 -0700 >> I am sure this has been asked before, but why not use >> Google newsgroup or at least some forum BBS software >> instead of this cumbersome mailing list process? >Someone's wanting to get flamed today eh? ;-) > >I'm a big fan of bulletin board type systems and have used phpBB quite a bit >(www.phpbb.com) - it's great because it has a "show me all new messages >since I last logged in" feature and allows for multiple forums/topics etc. >You can get it to email you if someone replies to a thread that you've >marked/are "watching". > >That said, I like the Asterisk mailing lists too - they end up in different >folders in my Outlook, I can follow threads etc, it works well too. > >Six of one, half a dozen of the other I guess - everyone has their personal >preferences. As for "cumbersome" - hmm, that's the word that'll maybe rock >the boat on the list. What's the problem with a mailing list? So long as >people can handle the basics of reply quoting etiquette and keep away from >HTML formatting, there should be few problems? I guess the only thing is the >search capabilities but then you can do that through Google or another >archive indexer - it's just maybe not that obvious/clear how to do it.. > >For list reference/archiving, go to www.google.com, in the search keywords >box type "site:lists.digium.com" followed by the keywords you want to >search. > >Anyone else got thoughts/evidence on email versus BBS type working? > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Costas Menico Meezon Software Corp 201-224-8111 [EMAIL PROTECTED] -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NAT/SIP solution?
There are several SIP aware NAT routers. Any Cisco router with a firewall load has SIP aware NAT. There is at least one other brand of SIP aware NAT router out there, but I don't recall the brand. On Sun, 2003-09-28 at 17:26, Uriel Carrasquilla wrote: > Nikotel has a solution and one participant in thi list is doing a trial on a > SIP/NAT router (claiming to be the first one in this realm). -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Flash and Call Transfer
I am new to asterisk and have been playing with it for just a few days. The GrandStream BudgeTone-102 equipment I have allows me to do a blind transfer, but I am interested to know how to do something which isn't blind. I have managed to enable the call parking which allows me to send a call to extension 700 for somebody to collect, but it still seems rather awkward. I also have problems with the 'flash' button, nothing seems to happen when I press it. Strangely though the flash button did start working at one point and I was able to switch between two different calls using it, but rebooting the phones seemed to result in it not working anymore? I would be delighted to read any documentation, but I am struggling to find anything that refers to these issues. I have seen handbook version 2, is there anything else I am missing? Many thanks, Michael -- Michael T Farnworth Maxima Systems Ltd (http://www.maximasystems.com) 16 Woodbourne Sq Douglas Isle of Man IM1 4DB Tel: +44 (0)1624 665826 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT/SIP solution?
Greetings, I was wondering if somebody is working on a solution to the NAT/SIP-issues? It seems to me that the problem has been identified, is that correct? Just hoping that someone with more skills will provide us with a solution sooner or later... Regards, Stig http://bugs.digium.com/bug_view_page.php?bug_id=104 This has been identified and is waiting for a solution. Feel free to contribute a patch! JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NAT/SIP solution?
I might be putting words into Stig's message but I think what he means to ask was the following scenario that causes problems: SIP --- NAT --- Internet --- NAT --- Asterisk Nikotel has a solution and one participant in thi list is doing a trial on a SIP/NAT router (claiming to be the first one in this realm). To answer Stig's question as I understand it: I don't think anybody is working on a solution in this list since the by-pass is to put Asterisk directly on the Internet with its own public-IP address. Regards, Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of WipeOut Sent: Sunday, September 28, 2003 2:21 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] NAT/SIP solution? Stig Hess wrote: > Greetings, > > I was wondering if somebody is working on a solution to the > NAT/SIP-issues? It seems to me that the problem has been identified, > is that correct? > > Just hoping that someone with more skills will provide us with a > solution sooner or later... > > Regards, > > Stig The solution is to use "nat=yes" in your sip.conf.. so far this has worked great for me.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Google newsgroup or Forum setup.
> I am sure this has been asked before, but why not use > Google newsgroup or at least some forum BBS software > instead of this cumbersome mailing list process? Someone's wanting to get flamed today eh? ;-) I'm a big fan of bulletin board type systems and have used phpBB quite a bit (www.phpbb.com) - it's great because it has a "show me all new messages since I last logged in" feature and allows for multiple forums/topics etc. You can get it to email you if someone replies to a thread that you've marked/are "watching". That said, I like the Asterisk mailing lists too - they end up in different folders in my Outlook, I can follow threads etc, it works well too. Six of one, half a dozen of the other I guess - everyone has their personal preferences. As for "cumbersome" - hmm, that's the word that'll maybe rock the boat on the list. What's the problem with a mailing list? So long as people can handle the basics of reply quoting etiquette and keep away from HTML formatting, there should be few problems? I guess the only thing is the search capabilities but then you can do that through Google or another archive indexer - it's just maybe not that obvious/clear how to do it.. For list reference/archiving, go to www.google.com, in the search keywords box type "site:lists.digium.com" followed by the keywords you want to search. Anyone else got thoughts/evidence on email versus BBS type working? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Google newsgroup or Forum setup.
I am sure this has been asked before, but why not use Google newsgroup or at least some forum BBS software instead of this cumbersome mailing list process? -- Costas Menico Meezon Software Corp 201-224-8111 [EMAIL PROTECTED] -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] configuration
Is everything with Asterisk setup using configuration files or is there a GUI type of an interface that can be used? Thanks Costas -- Costas Menico Meezon Software Corp 201-224-8111 [EMAIL PROTECTED] -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE410P timing and multiple, different spans
Greetings... I have a TE410P with four T1's going into it. Things look roughly like this: #1 Goes to PBX -- we're responsible for timing #2 E&M span to telco 1 #3 PRI span to telco 1 #4 PRI span to telco 2 If I set primary sync source to span 2, users report strange echo, distortion, and crosstalk problems, which sound remarkably like frame slippage on spans 3 and 4. If I set it to use #3 as primary, the other spans report problems, etc. Generally, it seems like the clocks just aren't quite close enough to make things work right all of the time. I don't forsee either involved telco doing anything at all about this. Is there any way to see if there is some sort of weird timing problem? Even better, is there a way to get things to work right on the TE410P? Thanks :-) -rt -- Ryan Tucker Network Engineer NetAccess, Inc. 1159 Pittsford-Victor Road Bldg. 5, Suite 140 Pittsford, New York 14534 585-419-8200 www.netacc.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
Hi, Checked out latest CVS and no sound from Playback, Background, MOH or bridged channels. mpg123 is active but no sound. Master ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Latest CVS breaks sound
Title: Message Hi, Checked out latest CVS and no sound from Playback, Background, MOH or bridged channels. mpg123 is active but no sound. Master
Re: [Asterisk-Users] NAT/SIP solution?
Stig Hess wrote: I meant where Asterisk is behing a NAT... sorry for the confusion. Hi Stig, If you are able to run * on your NAT'd box, then I have come up with a work around (thanks wasim!!!) that will allow you to run an * box behind your NAT, and still recieve and make SIP calls. I haven't got the whole thing figured out yet in terms of extensions.conf (but I am working on that today, will post on my website later) but this is basically my configuration: <--TDM400P--> <*> <--IAX--> <*> <--SIP--> |-NAT-||FW| So basically the * on the GW machine which is also the NAT / FW box recieves the connection from the SIP remote end, then forwards all the traffic over IAX to the NAT'd * box. I just tested it, and it works fine! Once I get some more complex extensions.conf files setup, I will post them. Thanks, Leif Madsen. BTW: As for just passing SIP through SIP, I believe it's a limitation of the SIP protocol as the RTP ports are different than the connection port, whereas IAX is all the same port for everything (from what I gather) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT/SIP solution?
I've heard of this: http://sarp.sourceforge.net I have no idea if it will be of any help, but it's an interesting programme for handleing SIP over NAT. Brian. - Original Message - From: Stig Hess To: [EMAIL PROTECTED] Sent: Sunday, September 28, 2003 2:17 PM Subject: [Asterisk-Users] NAT/SIP solution? Greetings, I was wondering if somebody is working on a solution to the NAT/SIP-issues? It seems to me that the problem has been identified, is that correct? Just hoping that someone with more skills will provide us with a solution sooner or later... Regards, Stig
Re: [Asterisk-Users] NAT/SIP solution?
Stig Hess wrote: I meant where Asterisk is behing a NAT... sorry for the confusion. Regards, Stig H. Oh.. :) Well thats a bigger problem.. and i doubt the "Gods of SIP" are going to fix it any time soon.. :( Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with PHPconfig setup??
Hi, Just giving phpconfig a try but can't find and setup instructions.. What I have done so far.. 1. Copied the phpconfig files to the web dir on the server. 2. Edited the manager.conf and enabled manager access and setup an accound (really just copied the sample config but set my own username and secret) 3. Made /etc/asterisk/* world writable. (I guess I should have made the owned by apache and then user writable but its a dev server so I am not to worried about that) 4. Edited phpconfig_init.php with the username from manager.conf. 5. Edited asterisk.reload and changed the values for Username: and Secret: 6. Made asterisk.reload executable. So far all seems to be working except for "Re-Read Configs".. I get "Reset failed." when I try.. So what have I left out??.. If I execute asterisk.reload in a shell ir works fine so it must be somthing to do with calling it from phpconfig.. (I am using a Linux desktop and Mozilla 1.4 but that shouldnt make any difference.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT/SIP solution?
I meant where Asterisk is behing a NAT... sorry for the confusion. Regards, Stig H. - Original Message - From: WipeOut To: [EMAIL PROTECTED] Sent: Sunday, September 28, 2003 8:21 PM Subject: Re: [Asterisk-Users] NAT/SIP solution? Stig Hess wrote:> Greetings,> > I was wondering if somebody is working on a solution to the > NAT/SIP-issues? It seems to me that the problem has been identified, > is that correct?> > Just hoping that someone with more skills will provide us with a > solution sooner or later...> > Regards,> > StigThe solution is to use "nat=yes" in your sip.conf.. so far this has worked great for me..Later..___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT/SIP solution?
Stig Hess wrote: Greetings, I was wondering if somebody is working on a solution to the NAT/SIP-issues? It seems to me that the problem has been identified, is that correct? Just hoping that someone with more skills will provide us with a solution sooner or later... Regards, Stig The solution is to use "nat=yes" in your sip.conf.. so far this has worked great for me.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NAT/SIP solution?
Greetings, I was wondering if somebody is working on a solution to the NAT/SIP-issues? It seems to me that the problem has been identified, is that correct? Just hoping that someone with more skills will provide us with a solution sooner or later... Regards, Stig
[Asterisk-Users] "i8253 counter to high. Resetting..." message
Hi, I have one * box with this message repeating endless in the system console. It dissapear after a system reset, but it is back after a while. The only PCI card installed is a X100P. It can be related to Asterisk and/or X100P card driver or it is a system failure? The system is a [EMAIL PROTECTED], original Intel Motherboard with RH9. Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk CVS viewer on line
Hi to all. I've put on line a cvs viewer for asterisk source code. Is based onto the suite horde+chora. The website is http://asterisk.espia-net.net The cvs modules shown are * asterisk * asterisk-addons * zaptel * zapata * libpri * libr2 * libiax * libiax2 * gnophone * phpconfig * gastman all revisions, branch , comments & whatever cvs is has been preserved. this could be a sort of mirror. I've installed such system, since I prefer to have a quick view of the cvs changes via web. and chora is pretty good for that. Note that no cvs anon access isn't possible for now (although could be enabled). feel free to use it ;) P.S. the mirror is updated daily, at 05.00 Rome Time (italy). Digium staff: hope that isn't wrong for you ;) Matteo. -- Matteo Brancaleoni Espia System Administrator http://www.espia.it This message was sent using IMP, the Internet Messaging Program. Service is provided by Espia - Emmegi Srl - http://www.espia.it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] how stable is dynextendb
Jeremy: where can I find "retrieve_extensions_from_mysql.pl"? Regards, Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jeremy McNamara Sent: Saturday, September 27, 2003 9:17 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] how stable is dynextendb > > >> I'm looking for a way to manage large dial plans. >> >> Blitz on IRC mentioned DynExtenDB >> >> I'm wondering how stable it is since its not been >> updated since 2002-12-15 >> >> >> Any other ideas ?? >> I want to have my dial plan in a SQL database > > > I actually just stumbled upon this today, and looks very interesting > and useful. Is anyone actually using this to do what John is looking > to do? IMHO, DynExtenDB is the absolute wrong way to deal with Extensions from a database.Have you actually looked in the asterisk source tree? See retrieve_extensions_from_mysql.pl. Granted it is not perfect or quite how I would do it, but it does get the job done without any craziness. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CDR Web Search Frontend
Jamie: thank you. I will install later today. Regards, Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jamie Carl Sent: Saturday, September 27, 2003 11:17 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] CDR Web Search Frontend *This message was transferred with a trial version of CommuniGate(tm) Pro* It's in the archives. People on this list usually don't take too well to repeating stuff. :) (i'm not fussed tho) http://asterisk.jazz-inc.net Yes, the source is of course available for download. :) Enjoy! J > -Original Message- > From: Uriel Carrasquilla [mailto:[EMAIL PROTECTED] > Sent: Sunday, 28 September 2003 1:05 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] CDR Web Search Frontend > > > *This message was transferred with a trial version of > CommuniGate(tm) Pro* > does it include the source in PhP? > what was the link again please? > Uriel > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Jamie Carl > Sent: Saturday, September 27, 2003 3:42 AM > To: Asterisk Users (E-mail); Asterisk Dev (E-mail) > Subject: RE: [Asterisk-Users] CDR Web Search Frontend > > > *This message was transferred with a trial version of > CommuniGate(tm) Pro* > > Hey all, > > New versions available. Now written in PHP with totals for Billing > Seconds and Duration. > > Help yourselves and please send me more suggestions!!! > Thanx! > > J > > > -Original Message- > > From: Dimitri Bellini [mailto:[EMAIL PROTECTED] > > Sent: Friday, 26 September 2003 10:40 PM > > To: [EMAIL PROTECTED] > > Subject: Re: [Asterisk-Users] CDR Web Search Frontend > > > > > > *This message was transferred with a trial version of > > CommuniGate(tm) Pro* > > Hi Carl > > i see web frontend i action is very good!! The total > > time at end is good > > thing. > > Thanks for great work. Can you put the script in some place > > to download. > > > > Dimitri > > > > > *This message was transferred with a trial version of > > CommuniGate(tm) Pro* > > > > > > Hey all, > > > > > > I've just done a quick (but functional) web front end for > > searching the > > > CDRs in a MySQL database. Anyone interested in trying it > out? I'm > > > wondering what to add to it next. > > > > > > So far you can seach using source, destination, CLI, > > channel and date > > > ranges. It also displays ALL fields in the database table. > > > > > > If interested, email me on [EMAIL PROTECTED] Do not reply > > directly to > > > this email, it will bounce. Depending on the level of > > interest, I may > > > post this somewhere for your free downloading pleasure. > > > > > > Regards, > > > > > > Jamie Carl > > > Jazz Inc. > > > http://www.jazz-inc.net > > > Email: [EMAIL PROTECTED] > > > JID: [EMAIL PROTECTED] > > > Phone: +61-414-365466 > > > > > > > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Config TE410P + TDM400
At 18:28 26-9-2003 +0200, you wrote: When configuring a TE410P which is only attached to a single E1 together with a TDM400, how should one count the channels for the next Zap interface? Must I put 4 span lines in zapata.conf and define all channels up to 124? thus having the TDM400's start at 125? Or can I comment out the 3 spans I don't use and start at channel 32 for the TDM400? (this would get nasty when adding extra lines, but would stop asterisk from trying to look at E1's which are not connected) I have an E400P with just one active span. It will work just fine with the span-lines commented out :-) Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Installation counter
On September 27, 2003 10:23 pm, Paul Crick wrote: > I'm wondering about putting something similar to the Linux Counter > (http://counter.li.org/) together that would allow people to register > their real world installations, documenting hardware specs, setup, phones > used etc? Neat idea, you could have people either use SIP or IAX(2) to register. :-) Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Budgetone + NAT: Firmware Version?
On Sun, 2003-09-28 at 05:10, Uriel Carrasquilla wrote: > Is there anyway to prevent the BudgetTone from just doing a BIOS upgrade > without consulting? It can only do it if you have a valid address in the TFTP Server field, mine points to my own tftp server as does the NTP Server field. -- Dave Cotton <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Continuing Budgetone woes: asterisk was the culprit!!
On Sun, 2003-09-28 at 07:24, Brian Capouch wrote: > Brian Capouch wrote: > > I have spent the morning on this project, still without success. > > > > When I saw the mail on the list tonight from "[EMAIL PROTECTED]" it finally > dawned on me to try a CVS-revert and see what happens. > > It turns out that solved the problem--I can't say when exactly the bug > showed up, but I did fresh CVS builds a couple of times in the past > couple of days and they *have* the bug. I had a similar situation a few days back reverted 24 hours, problem solved then re-CVSed with no problem, now running 09/27/03. But in between I'd done a complete reboot of the server. -- Dave Cotton <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Installation counter
Something like Linux Counter, or I also found a site here in France asking for people who'd installed Openoffice with a few details of how they used it, they even had the possibility to link to your own site with a counter on that. I was surprised how many had looked at my site from that link. When I get time I'll find it again. -- Dave Cotton <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
